
From magnus.westerlund@ericsson.com  Tue Oct  2 08:08:28 2012
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Date: Tue, 2 Oct 2012 17:08:23 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Agenda requests for Atlanta meeting
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WG,

We chairs like to have input into what you see should be on the Agenda
for our two meeting slots in Atlanta. Both document editors/authors and
participant are welcome to request or inform the WG or the chairs only
about issues that needs to be discussed in the meeting.

Two items we chairs believe should be on the agenda are:

- Video codec selection
- JSEP



Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From ted.ietf@gmail.com  Tue Oct  2 09:25:33 2012
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From: Ted Hardie <ted.ietf@gmail.com>
To: rtcweb@ietf.org
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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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Just a reminder that this deadline is approaching quickly; please get
your drafts in!

thanks,

Ted Hardie


On Thu, Aug 16, 2012 at 10:14 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
> As agreed at the recent meeting, the chairs solicit internet-drafts
> naming proposed mandatory-to-implement video codecs or codec sets by
> October 15th, 2012.  Please include any technical data you believe
> supports the choice you are proposing.  A non-exhaustive list of data
> you may wish to include is found in slides 3-6 of
> http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-4.pptx .
>
> Discussion based on these will commence immediately upon receipt,
> rather than waiting until the October 15th deadline, so early
> submission is encouraged.
>
> regards,
>
> Ted Hardie

From christer.holmberg@ericsson.com  Tue Oct  2 11:00:18 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 2 Oct 2012 20:00:15 +0200
Thread-Topic: JSEP: Relaxing SDP O/A rules?
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--_000_7F2072F1E0DE894DA4B517B93C6A0585340A1E3AB1ESESSCMS0356e_
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Hi,

One of the arguments for the Microsoft API proposal was regarding the JSEP =
usage of SDP Offer/Answer.

There has also been ideas about JSEP "relaxing" the SDP O/A rules.

What is the current status of that? Is the intention to "relax" something?

Regards,

Christer


--_000_7F2072F1E0DE894DA4B517B93C6A0585340A1E3AB1ESESSCMS0356e_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<font face=3D"Arial, sans-serif" size=3D"3">
<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">One of the arguments for the Microsoft API proposal w=
as regarding the JSEP usage of SDP Offer/Answer.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">There has also been ideas about JSEP &quot;relaxing&q=
uot; the SDP O/A rules.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">What is the current status of that? Is the intention =
to &quot;relax&quot; something? </font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
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From Markus.Isomaki@nokia.com  Tue Oct  2 11:30:31 2012
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From: <Markus.Isomaki@nokia.com>
To: <ted.ietf@gmail.com>, <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Video codec proposals due October 15th, 2012
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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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Hi,

Why do you want to receive these proposals as drafts instead of just on the=
 mailing list? I have a proposal/opinion and a few simple arguments for it,=
 but writing a draft about it seems like a lot of overhead.=20

Markus=20


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Ted Hardie
>Sent: 02 October, 2012 19:25
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
>
>Just a reminder that this deadline is approaching quickly; please get your
>drafts in!
>
>thanks,
>
>Ted Hardie
>
>
>On Thu, Aug 16, 2012 at 10:14 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>> As agreed at the recent meeting, the chairs solicit internet-drafts
>> naming proposed mandatory-to-implement video codecs or codec sets by
>> October 15th, 2012.  Please include any technical data you believe
>> supports the choice you are proposing.  A non-exhaustive list of data
>> you may wish to include is found in slides 3-6 of
>> http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-4.pptx .
>>
>> Discussion based on these will commence immediately upon receipt,
>> rather than waiting until the October 15th deadline, so early
>> submission is encouraged.
>>
>> regards,
>>
>> Ted Hardie
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From kaiduanx@gmail.com  Tue Oct  2 11:56:52 2012
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From: Kaiduan Xie <kaiduanx@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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I also have this concern particularly on how to inter-operate with the
in-service SIP world which follows the SDP O/A rules?

/Kaiduan

On Tue, Oct 2, 2012 at 2:00 PM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
> One of the arguments for the Microsoft API proposal was regarding the JSEP
> usage of SDP Offer/Answer.
>
> There has also been ideas about JSEP "relaxing" the SDP O/A rules.
>
> What is the current status of that? Is the intention to "relax" something?
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From martin.thomson@gmail.com  Tue Oct  2 11:59:56 2012
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On 2 October 2012 11:00, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> There has also been ideas about JSEP "relaxing" the SDP O/A rules.

JSEP already does that.

PRANSWER is probably the most obvious diversion, but it goes deeper.

There's a simple view of what JSEP is and it goes a little like this:

 - SDP provides a description of the session that you are going to create [1]

 - You need two SDP blobs so that you can get all the non-negotiated
parameters like crypto and candidate attributes

 - setLocalDescription and setRemoteDescription provide a way of
matching up two SDP blobs into a complete session description

 - negotiation is performed outside this machine, and the browser can
help you with that in two ways: providing createOffer and createAnswer
to build the SDP; and by nudging you with a negotiatedneeded event
when it learns of things that necessitate changes.

Obviously, it's a little more nuanced than that, but if you take this
view, you are more likely to get something to work.

--Martin

[1] Caveat: PRANSWER adds a layer of complexity to this by shifting
part of the session description to the offer rather than the answer

From ted.ietf@gmail.com  Tue Oct  2 12:02:36 2012
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From: Ted Hardie <ted.ietf@gmail.com>
To: Markus.Isomaki@nokia.com
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Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
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On Tue, Oct 2, 2012 at 11:30 AM,  <Markus.Isomaki@nokia.com> wrote:
> Hi,
>
> Why do you want to receive these proposals as drafts instead of just on the mailing list? I have a proposal/opinion and a few simple arguments for it, but writing a draft about it seems like a lot of overhead.
>
> Markus
>

Sorry that the draft system creates extra overhead, but we think it is
useful to make all the proposals parallel, and we believe at least
some of them will have technical justifications that would be
complicated to include in an email.  In particular, why you'd pick a
specific profile of a codec with multiple profiles may require
contextualizing the different profiles in ways that would be difficult
to put into an email.  When we have drafts, we are also more likely to
get reviews from those who are picking up on our work from grabbing
the related drafts.  Lastly, if folks need to declare IPR in a
proposal, the system is better set up to do so with a draft than an
email.

Thanks for taking it on despite the overhead,

regards,

Ted


>
>>-----Original Message-----
>>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>>Of ext Ted Hardie
>>Sent: 02 October, 2012 19:25
>>To: rtcweb@ietf.org
>>Subject: Re: [rtcweb] Video codec proposals due October 15th, 2012
>>
>>Just a reminder that this deadline is approaching quickly; please get your
>>drafts in!
>>
>>thanks,
>>
>>Ted Hardie
>>
>>
>>On Thu, Aug 16, 2012 at 10:14 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>>> As agreed at the recent meeting, the chairs solicit internet-drafts
>>> naming proposed mandatory-to-implement video codecs or codec sets by
>>> October 15th, 2012.  Please include any technical data you believe
>>> supports the choice you are proposing.  A non-exhaustive list of data
>>> you may wish to include is found in slides 3-6 of
>>> http://www.ietf.org/proceedings/84/slides/slides-84-rtcweb-4.pptx .
>>>
>>> Discussion based on these will commence immediately upon receipt,
>>> rather than waiting until the October 15th deadline, so early
>>> submission is encouraged.
>>>
>>> regards,
>>>
>>> Ted Hardie
>>_______________________________________________
>>rtcweb mailing list
>>rtcweb@ietf.org
>>https://www.ietf.org/mailman/listinfo/rtcweb

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To: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 3 Oct 2012 15:03:10 +0200
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From kaiduanx@gmail.com  Wed Oct  3 06:42:52 2012
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Date: Wed, 3 Oct 2012 09:42:51 -0400
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From: Kaiduan Xie <kaiduanx@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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There was a discussion on this,

http://www.ietf.org/mail-archive/web/rtcweb/current/msg05141.html

However I have not seen how the IETF webrtc work group address this issue.

/Kaiduan

On Wed, Oct 3, 2012 at 9:03 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> Hi,
>
> I've been looking at the O/A state machine, and the associated procedure
> text, in section 4.2 of draft-jsep, and a couple of issues.
>
> First, the text says:
>
>         "As in [RFC3264], an offerer can send an offer, and update it as
> long as it has not been answered."
>
> That is not true. In order to update an offer (before it has been answered),
> one needs to cancel the previous offer.
>
>
> Second, the text says:
>
>         "A description used as a "pranswer" may be applied as a response to
> an "offer", or an update to a previously sent "answer"."
>
> I don't understand the update-to-a-previously-sent-answer thing. I don't
> understand why one would update an "answer" with a "preanswer".
>
> And, the state machine doesn't seem to support it either. So, if done,  what
> state will I be in after such action? Can I then also send a new "answer",
> to update the "preanswer" that updated the "answer"?
>
>
> Third, is there a reason why a new "offer" can't be sent once in the
> "Preanswer" state?
>
>
> Regards,
>
> Christer
>
>
>
>
>
> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: 2. lokakuuta 2012 22:00
> To: Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
>
> On 2 October 2012 11:00, Christer Holmberg <christer.holmberg@ericsson.com>
> wrote:
>> There has also been ideas about JSEP "relaxing" the SDP O/A rules.
>
> JSEP already does that.
>
> PRANSWER is probably the most obvious diversion, but it goes deeper.
>
> There's a simple view of what JSEP is and it goes a little like this:
>
> - SDP provides a description of the session that you are going to create [1]
>
> - You need two SDP blobs so that you can get all the non-negotiated
> parameters like crypto and candidate attributes
>
> - setLocalDescription and setRemoteDescription provide a way of matching up
> two SDP blobs into a complete session description
>
> - negotiation is performed outside this machine, and the browser can help
> you with that in two ways: providing createOffer and createAnswer to build
> the SDP; and by nudging you with a negotiatedneeded event when it learns of
> things that necessitate changes.
>
> Obviously, it's a little more nuanced than that, but if you take this view,
> you are more likely to get something to work.
>
> --Martin
>
> [1] Caveat: PRANSWER adds a layer of complexity to this by shifting part of
> the session description to the offer rather than the answer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From martin.thomson@gmail.com  Wed Oct  3 08:26:39 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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These are just a few of the problems with the provisional answer.

On 3 October 2012 06:03, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>         "As in [RFC3264], an offerer can send an offer, and update it as
> long as it has not been answered."
>
> That is not true. In order to update an offer (before it has been answered),
> one needs to cancel the previous offer.

This might be OK if it were to include the fact that any outstanding
offer is implicitly cancelled.

Note that it is not clear how the current API could be used to cancel
an offer.  Various suggestions include passing a null offer or by
setting the local (or remote?) description to a previous offer.

>         "A description used as a "pranswer" may be applied as a response to
> an "offer", or an update to a previously sent "answer"."
>
> I don't understand the update-to-a-previously-sent-answer thing. I don't
> understand why one would update an "answer" with a "preanswer".

This doesn't make sense to me either.  I could understand if it was an
update to a previously sent *pranswer*, but since an answer is final,
any further answers are illegal.

> And, the state machine doesn't seem to support it either. So, if done,  what
> state will I be in after such action? Can I then also send a new "answer",
> to update the "preanswer" that updated the "answer"?

That's why I think that this is in error.

> Third, is there a reason why a new "offer" can't be sent once in the
> "Preanswer" state?

Yes.  If you send an offer out to multiple people and one gives a
provisional answer, the new offer you create would have to be cleverly
crafted to permit all sessions described by the previous offer as well
as any new stuff, such that an answer to the old offer could be
applied to the new offer.  That really complicates things.

In the assumed model, all answers and pranswers are made in respect to
a specific offer.  answers and pranswers only be applied to a session
that has applied their specific offer.  pranswers and answers can
replace pranswers that reference the same offer.

This is an expansion on 3264.  This is what draft-*-jsep needs to
describe clearly, if that is appropriate.  I am slowly reaching the
conclusion that pranswer is a complication that we don't need,
especially since - for those who care about the functionality - it can
be replicated by editing offers and answers.

--Martin

From harald@alvestrand.no  Thu Oct  4 01:57:35 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 10/03/2012 05:26 PM, Martin Thomson wrote:
> This is an expansion on 3264. This is what draft-*-jsep needs to 
> describe clearly, if that is appropriate. I am slowly reaching the 
> conclusion that pranswer is a complication that we don't need, 
> especially since - for those who care about the functionality - it can 
> be replicated by editing offers and answers.

I would be very happy to lose PRANSWER if the SIP guys tell us they can 
successfully gateway to systems using provisional SIP answers (which I 
think is the same thing as the one called "serial forking") without them.

For the Web use case, I think PRANSWER is an unnecessary distraction; if 
we're writing both initiator and responder, multiple O/A rounds, or use 
of multiple PeerConnections, is a much cleaner solution for the use 
cases I can wrap my head around.




From christer.holmberg@ericsson.com  Thu Oct  4 02:54:59 2012
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

>> This is an expansion on 3264. This is what draft-*-jsep needs to
>> describe clearly, if that is appropriate. I am slowly reaching the
>> conclusion that pranswer is a complication that we don't need,
>> especially since - for those who care about the functionality - it can
>> be replicated by editing offers and answers.
>
> I would be very happy to lose PRANSWER if the SIP guys tell us they can
> successfully gateway to systems using provisional SIP answers (which I
> think is the same thing as the one called "serial forking") without them.

I guess serial forking could be handled using cloning, in the same way as p=
arallel forking.

(Assuming we are going to support cloning, that is - currently it IS descri=
bed in JSEP)

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Oct  4 02:56:25 2012
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Hi,

>>         "As in [RFC3264], an offerer can send an offer, and update it=20
>> as long as it has not been answered."
>>
>> That is not true. In order to update an offer (before it has been=20
>> answered), one needs to cancel the previous offer.
>
> This might be OK if it were to include the fact that any outstanding offe=
r is implicitly cancelled.

Sure. My issue was about the RFC 3264 claim. It's one of the "relaxes" we w=
ould have to specify ourselves.

> Note that it is not clear how the current API could be used to cancel an =
offer.  Various suggestions include passing a null offer or by setting the =
local (or remote?) description to a previous offer.

I assume that setting the description to a previous offer, from an API pers=
pective, is the same as updating the offer with something new.

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Oct  4 04:26:02 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 4 Oct 2012 13:25:56 +0200
Thread-Topic: JSEP-02: Clone comments
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Subject: [rtcweb] JSEP-02: Clone comments
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Hi,

A couple of comments on the cloning stuff in JSEP:

Q1. The document doesn't say how the cloning is performed. If that is outsi=
de the scope of JSEP, and belongs to the W3C API spec, I think it should be=
 mentioned.

Q2. The text in section 4.7.2 says:

                "As a result of this cloning, the application will end up w=
ith N
                parallel sessions, each with a local and remote description=
 and their
                own local and remote addresses."

I think the "own local addresses" wording is a little misleading, as each c=
lone will share the same local address.

Regards,

Christer



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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span lang=3DFI>=
Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal>A couple of comments on the cloning stu=
ff in JSEP:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p clas=
s=3DMsoNormal>Q1. The document doesn&#8217;t say how the cloning is perform=
ed. If that is outside the scope of JSEP, and belongs to the W3C API spec, =
I think it should be mentioned.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nb=
sp;</o:p></p><p class=3DMsoNormal>Q2. The text in section 4.7.2 says:<o:p><=
/o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp; &#8220;As a result of this cloning, the application will end u=
p with N<o:p></o:p></p><p class=3DMsoNormal>&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; parallel sessions, e=
ach with a local and remote description and their<o:p></o:p></p><p class=3D=
MsoNormal>&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; own local and remote addresses.&#8221;<o:p></o:p></p><=
p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I think the &=
#8220;own local addresses&#8221; wording is a little misleading, as each cl=
one will share the same local address.<o:p></o:p></p><p class=3DMsoNormal><=
o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p></p><p class=3D=
MsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Christer<o:p></o:p></p>=
<p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><o:p>&nbsp;<=
/o:p></p></div></body></html>=

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Subject: Re: [rtcweb] JSEP-02: Clone comments
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On 10/04/2012 01:25 PM, Christer Holmberg wrote:
>
> Hi,
>
> A couple of comments on the cloning stuff in JSEP:
>
> Q1. The document doesn't say how the cloning is performed. If that is 
> outside the scope of JSEP, and belongs to the W3C API spec, I think it 
> should be mentioned.
>
> Q2. The text in section 4.7.2 says:
>
>                 "As a result of this cloning, the application will end 
> up with N
>
>                 parallel sessions, each with a local and remote 
> description and their
>
>                 own local and remote addresses."
>
> I think the "own local addresses" wording is a little misleading, as 
> each clone will share the same local address.
>

Not sure what to say here.

I can see multiple ways to implement this - some will end up with 
different local ports, some won't.

If the non-local (reflexive?) candidates are allocated using STUN on a 
per-port basis, addresses could be different too.


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    <div class="moz-cite-prefix">On 10/04/2012 01:25 PM, Christer
      Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A0585340A7BCD49@ESESSCMS0356.eemea.ericsson.se"
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        <p class="MsoNormal"><span lang="FI">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span lang="FI"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal">A couple of comments on the cloning stuff
          in JSEP:<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Q1. The document doesn&#8217;t say how the
          cloning is performed. If that is outside the scope of JSEP,
          and belongs to the W3C API spec, I think it should be
          mentioned.<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Q2. The text in section 4.7.2 says:<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &#8220;As a result of this
          cloning, the application will end up with N<o:p></o:p></p>
        <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; parallel sessions, each
          with a local and remote description and their<o:p></o:p></p>
        <p class="MsoNormal">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; own local and remote
          addresses.&#8221;<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">I think the &#8220;own local addresses&#8221; wording
          is a little misleading, as each clone will share the same
          local address.</p>
      </div>
    </blockquote>
    <br>
    Not sure what to say here.<br>
    <br>
    I can see multiple ways to implement this - some will end up with
    different local ports, some won't.<br>
    <br>
    If the non-local (reflexive?) candidates are allocated using STUN on
    a per-port basis, addresses could be different too.<br>
    <br>
  </body>
</html>

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From christer.holmberg@ericsson.com  Thu Oct  4 04:57:29 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 4 Oct 2012 13:57:26 +0200
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Subject: Re: [rtcweb] JSEP-02: Clone comments
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Hi,

>> A couple of comments on the cloning stuff in JSEP:
>>
>> Q1. The document doesn=1B$B!G=1B(Bt say how the cloning is performed. If=
 that is outside the scope of JSEP, and belongs to the W3C API spec, I thin=
k it should be mentioned.
>>
>> Q2. The text in section 4.7.2 says:
>>
>>                =1B$B!H=1B(BAs a result of this cloning, the application =
will end up with N
>>                parallel sessions, each with a local and remote descripti=
on and their
>>                own local and remote addresses.=1B$B!I=1B(B
=1B$B"d=1B(B
>> I think the =1B$B!H=1B(Bown local addresses=1B$B!I=1B(B wording is a lit=
tle misleading, as each clone will share the same local address.
>
> Not sure what to say here.
>
> I can see multiple ways to implement this - some will end up with differe=
nt local ports, some won't.

Earlier in the same section, the text says:

        =1B$B!H=1B(BSince the clone uses the same local description as its
        parent, creating a clone will fail if it is not possible to reserve
        the same resources for the clone as have already been reserved by t=
he
        parent.=1B$B!I=1B(B

> If the non-local (reflexive?) candidates are allocated using STUN on a pe=
r-port basis, addresses could be different too.

My assumption, and understanding of the text, is that a clone is a=1B$B!D=
=1B(B eeeeh=1B$B!D=1B(B clone - meaning that the local address is the same =
:)

Regards,

Christer




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<div><font color=3D"#FF0000">Hi,</font></div>
<div><font color=3D"#FF0000">&nbsp;</font></div>
<div>&gt;&gt; A couple of comments on the cloning stuff in JSEP:</div>
<div>&gt;&gt;</div>
<div>&gt;&gt; Q1. The document doesn=1B$B!G=1B(Bt say how the cloning is pe=
rformed. If that is outside the scope of JSEP, and belongs to the W3C API s=
pec, I think it should be mentioned.</div>
<div>&gt;&gt;&nbsp;</div>
<div>&gt;&gt; Q2. The text in section 4.7.2 says:</div>
<div>&gt;&gt;&nbsp;</div>
<div>&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; =1B$B!H=1B(BAs a result of this cloning, the a=
pplication will end up with N</div>
<div>&gt;&gt;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; parallel sessions, each with a local and remote des=
cription and their</div>
<div>&gt;&gt;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; own local and remote addresses.=1B$B!I=1B(B</div>
<div><font face=3D"Cambria Math, serif">=1B$B"d=1B(B</font></div>
<div>&gt;&gt; I think the =1B$B!H=1B(Bown local addresses=1B$B!I=1B(B wordi=
ng is a little misleading, as each clone will share the same local address.=
</div>
<div>&gt;</div>
<div>&gt; Not sure what to say here.</div>
<div>&gt;</div>
<div>&gt; I can see multiple ways to implement this - some will end up with=
 different local ports, some won't.</div>
<div>&nbsp;</div>
<div><font color=3D"#FF0000">Earlier in the same section, the text says:</f=
ont></div>
<div><font color=3D"#FF0000">&nbsp;</font></div>
<div><font color=3D"#FF0000">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =1B=
$B!H=1B(BSince the <b>clone uses the same local description as its</b></fon=
t></div>
<div><font color=3D"#FF0000"><b>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; </b><b>parent</b>, creating a clone will fail if it is not possible t=
o reserve</font></div>
<div><font color=3D"#FF0000">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; the same resources for the clone as have already been reserved by the</f=
ont></div>
<div><font color=3D"#FF0000">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; parent.=1B$B!I=1B(B</font></div>
<div>&nbsp;</div>
<div>&gt; If the non-local (reflexive?) candidates are allocated using STUN=
 on a per-port basis, addresses could be different too.</div>
<div>&nbsp;</div>
<div><font color=3D"#FF0000">My assumption, and understanding of the text, =
is that a clone is a=1B$B!D=1B(B eeeeh=1B$B!D=1B(B clone &#8211; meaning th=
at the local address is the same :)</font></div>
<div><font color=3D"#FF0000">&nbsp;</font></div>
<div><font color=3D"#FF0000">Regards,</font></div>
<div><font color=3D"#FF0000">&nbsp;</font></div>
<div><font color=3D"#FF0000">Christer</font></div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
</font>
</body>
</html>

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From Jim.Barnett@genesyslab.com  Thu Oct  4 05:47:57 2012
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Subject: Re: [rtcweb] JSEP-02: Clone comments
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If the local address isn$B!G(Jt the same, we shouldn$B!G(Jt call it a $B!F(Jclone$B!G(J.  I think that people will expect everything to be the same on a $B!F(Jclone$B!G(J.  

 

-          Jim

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmberg
Sent: Thursday, October 04, 2012 7:57 AM
To: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-02: Clone comments

 

Hi,

 

>> A couple of comments on the cloning stuff in JSEP:

>> 

>> Q1. The document doesn$B!G(Jt say how the cloning is performed. If that is outside the scope of JSEP, and belongs to the W3C API spec, I think it should be mentioned.

>> 

>> Q2. The text in section 4.7.2 says:

>> 

>>                $B!H(JAs a result of this cloning, the application will end up with N

>>                parallel sessions, each with a local and remote description and their

>>                own local and remote addresses.$B!I(J

$B"d(J

>> I think the $B!H(Jown local addresses$B!I(J wording is a little misleading, as each clone will share the same local address.

> 

> Not sure what to say here.

> 

> I can see multiple ways to implement this - some will end up with different local ports, some won't.

 

Earlier in the same section, the text says:

 

        $B!H(JSince the clone uses the same local description as its

         parent, creating a clone will fail if it is not possible to reserve

         the same resources for the clone as have already been reserved by the

         parent.$B!I(J

 

> If the non-local (reflexive?) candidates are allocated using STUN on a per-port basis, addresses could be different too.

 

My assumption, and understanding of the text, is that a clone is a$B!D(J eeeeh$B!D(J clone (J-(J meaning that the local address is the same :)

 

Regards,

 

Christer

 

 

 


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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If the local address isn$B!G(Bt the same, we shouldn$B!G(Bt call it a $B!F(Bclone$B!G(B.&nbsp; I think that people will expect everything to be the same on a $B!F(Bclone$B!G(B.&nbsp; <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span style='mso-list:Ignore'>-<span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Jim<o:p></o:p></span></!
 p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Christer Holmberg<br><b>Sent:</b> Thursday, October 04, 2012 7:57 AM<br><b>To:</b> Harald Alvestrand; rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] JSEP-02: Clone comments<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>Hi,</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri"!
 ,"sans-serif";color:red'>&nbsp;</span><span style='font-size:10.0pt;fo
nt-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt; A couple of comments on the cloning stuff in JSEP:<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;<o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt; Q1. The document doesn$B!G(Bt say how the cloning is performed. If that is outside the scope of JSEP, and belongs to the W3C API spec, I think it should be mentioned.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt; Q2. The text in section 4.7.2 says:<o:p></o:p></span></p></div><div><p class=MsoNormal><span!
  style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $B!H(BAs a result of this cloning, the application will end up with N<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; parallel sessions, each with a local and remote description and their<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; own local and remote addresses.$B!I(B<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family!
 :"Cambria Math","serif"'>$B"d(B</span><span style='font-size:10.0pt;
font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;&gt; I think the $B!H(Bown local addresses$B!I(B wording is a little misleading, as each clone will share the same local address.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;<o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt; Not sure what to say here.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt;<o:p>&nbsp;</o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&gt; I can see multiple ways to implement this - some will end up with different local ports, some won't.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-!
 size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>Earlier in the same section, the text says:</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; $B!H(BSince the <b>clone uses the same local description as its</b></span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb!
 sp;&nbsp; parent</span></b><span style='font-size:10.0pt;font-family:"
Calibri","sans-serif";color:red'>, creating a clone will fail if it is not possible to reserve</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the same resources for the clone as have already been reserved by the</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; parent.$B!I(B</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-seri!
 f"'>&gt; If the non-local (reflexive?) candidates are allocated using STUN on a per-port basis, addresses could be different too.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>My assumption, and understanding of the text, is that a clone is a$B!D(B eeeeh$B!D(B clone &#8211; meaning that the local address is the same :)</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>Regards,</span><span style='font-size:10.0pt;font-f!
 amily:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p clas
s=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>&nbsp;</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";color:red'>Christer</span><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'>&nbsp;<o:p></o:p></span></p></div></div></body></html>
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From harald@alvestrand.no  Thu Oct  4 06:26:27 2012
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Date: Thu, 04 Oct 2012 15:26:20 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Chrome M23 beta released with the PeerConnection API
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Normally product releases aren't the stuff of IETF mailing lists, but 
this one is a milestone:

Chrome M23 is now in beta, and includes the current PeerConnection API.
This is an important milestone: First shipping implementation!

http://blog.chromium.org/2012/10/supporting-new-media-experiences-on-web.html 
is the announcement.

https://sites.google.com/site/webrtc/faq-recent-topics gives some more 
details on exactly what's implemented in this version.

                    Harald



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<html>
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    Normally product releases aren't the stuff of IETF mailing lists,
    but this one is a milestone:<br>
    <br>
    Chrome M23 is now in beta, and includes the current PeerConnection
    API.<br>
    This is an important milestone: First shipping implementation!<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="http://blog.chromium.org/2012/10/supporting-new-media-experiences-on-web.html">http://blog.chromium.org/2012/10/supporting-new-media-experiences-on-web.html</a>
    is the announcement.<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a href="https://sites.google.com/site/webrtc/faq-recent-topics">https://sites.google.com/site/webrtc/faq-recent-topics</a>
    gives some more details on exactly what's implemented in this
    version.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
  </body>
</html>

--------------050904050408080702070202--

From christer.holmberg@ericsson.com  Thu Oct  4 07:33:54 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 4 Oct 2012 16:33:50 +0200
Thread-Topic: JSEP-02: SDP line
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Subject: [rtcweb] JSEP-02: SDP line
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Hi,

What is a "SDP line"? The same thing as a "SDP blob"?

Regards,

Christer

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Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal>What is a &#8221;SDP line&#8221;? The s=
ame thing as a &#8220;SDP blob&#8221;?<o:p></o:p></p><p class=3DMsoNormal><=
o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p></p><p class=3D=
MsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Christer<o:p></o:p></p>=
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The Chairs would like to foster resolution of the long-running
discussion of support for additional key management methods, e.g. as
described in draft-ohlsson-rtcweb-sdes-support and the meeting
materials supporting draft-ietf-avtcore-srtp-ekt.   Working group
discussion of this matter is, of course, welcome, but the chairs are
also considering a virtual interim meeting in January of 2013, focused
on this specific topic.  Comments on this plan or on the relevant
approaches is welcome.

regards,

Ted, Cullen, Magnus

From fluffy@cisco.com  Thu Oct  4 08:44:10 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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With my individual contributor hat on =85

Justin and I would like to have some time for JSEP. I think there are a bun=
ch of issues, particularly around the use of SDP and timing of various thin=
gs that need discussion.=20

Thanks, Cullen



On Oct 2, 2012, at 9:08 AM, Magnus Westerlund <magnus.westerlund@ericsson.c=
om> wrote:

> WG,
>=20
> We chairs like to have input into what you see should be on the Agenda
> for our two meeting slots in Atlanta. Both document editors/authors and
> participant are welcome to request or inform the WG or the chairs only
> about issues that needs to be discussed in the meeting.
>=20
> Two items we chairs believe should be on the agenda are:
>=20
> - Video codec selection
> - JSEP
>=20
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From ekr@rtfm.com  Thu Oct  4 10:02:45 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 4 Oct 2012 10:02:03 -0700
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Subject: Re: [rtcweb] Resolving whether to support additional key management methods
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A virtual meeting seems OK.

I'm totally heads-down in code for the next while so I won't really
be able to contribute to much list discussion on this until after IETF...

Best,
-Ekr


On Thu, Oct 4, 2012 at 8:32 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
> The Chairs would like to foster resolution of the long-running
> discussion of support for additional key management methods, e.g. as
> described in draft-ohlsson-rtcweb-sdes-support and the meeting
> materials supporting draft-ietf-avtcore-srtp-ekt.   Working group
> discussion of this matter is, of course, welcome, but the chairs are
> also considering a virtual interim meeting in January of 2013, focused
> on this specific topic.  Comments on this plan or on the relevant
> approaches is welcome.
>
> regards,
>
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From martin.thomson@gmail.com  Thu Oct  4 11:00:55 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 4 October 2012 09:57, Harald Alvestrand <harald@alvestrand.no> wrote:
> For the Web use case, I think PRANSWER is an unnecessary distraction; if
> we're writing both initiator and responder, multiple O/A rounds, or use of
> multiple PeerConnections, is a much cleaner solution for the use cases I can
> wrap my head around.

I'd remove that qualifier.  We currently have a draft that describes
both cloning AND PRANSWER, both of which are more cleanly addressed by
either multiple O/A rounds or multiple independent sessions.

The two things that these provide is

PRANSWER:
The ability to describe a session that includes receipt capabilities
that are a superset of send capabilities.  SDP O/A doesn't really
provide a way for me to describe this because the default assumption
is that items not appearing in the answer are no longer needed.
With something like bundle involved, we could split m= lines into
sendonly and recvonly portions to allow for this case to be provided.

CLONING:
The ability to use the same transport primitives to interact with
multiple peers.  Ostensibly, this is to prepare for a session, with
only one continuing to completion, mainly to reduce clipping.
By adding candidates from all serial offenders to the same answer, you
can get ICE setup, though I doubt that DTLS handshakes will be able to
complete unless we permit multiple a=fingerprint lines and a few other
things.  I'm still disinclined to support this feature.  Let the
forkers have clipping.

Both of these complicate the API significantly.  I'd rather see the
complexity pushed to the applications that need this complexity.

That is, unless you are willing to consider a better API that makes
these issues SEP.

From christer.holmberg@ericsson.com  Thu Oct  4 11:21:27 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Harald Alvestrand <harald@alvestrand.no>
Date: Thu, 4 Oct 2012 20:21:22 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

>> For the Web use case, I think PRANSWER is an unnecessary distraction; if
>> we're writing both initiator and responder, multiple O/A rounds, or use =
of
>> multiple PeerConnections, is a much cleaner solution for the use cases I=
 can
>> wrap my head around.
>
> I'd remove that qualifier.  We currently have a draft that describes
> both cloning AND PRANSWER, both of which are more cleanly addressed by
> either multiple O/A rounds or multiple independent sessions.
>
>PRANSWER:
>The ability to describe a session that includes receipt capabilities
>that are a superset of send capabilities.  SDP O/A doesn't really
>provide a way for me to describe this because the default assumption
>is that items not appearing in the answer are no longer needed.
>With something like bundle involved, we could split m=3D lines into
>sendonly and recvonly portions to allow for this case to be provided.

I don't think the reason for PRANSWER comes from SDP O/A as such. It's more=
 about supporting serial forking.

>CLONING:
>The ability to use the same transport primitives to interact with
>multiple peers.  Ostensibly, this is to prepare for a session, with
>only one continuing to completion, mainly to reduce clipping.
>By adding candidates from all serial offenders to the same answer, you
>can get ICE setup, though I doubt that DTLS handshakes will be able to
>complete unless we permit multiple a=3Dfingerprint lines and a few other
>things.  I'm still disinclined to support this feature.  Let the
>forkers have clipping.

If you want to support parallel forking, you need to be able to simultanous=
ly perform ICE etc with all remote peers.

>Both of these complicate the API significantly.  I'd rather see the
>complexity pushed to the applications that need this complexity.

I want to be able to support forking. Exactly HOW it's done can be discusse=
d.

Also, I can live without support of parallel forking, and I don't really ca=
re if serial forking support is provided using cloning or being able to upd=
ate the remote descriptor (using PRANSWER, or something else).

>That is, unless you are willing to consider a better API that makes
>these issues SEP.

You have one in mind? ;)

Regards,

Christer=

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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 4 October 2012 11:21, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> I don't think the reason for PRANSWER comes from SDP O/A as such. It's more about supporting serial forking.

Not my point.  My point is that the desired functionality (apply the
answer without invalidating the offer) cannot be easily provided using
SDP O/A.  Most scenarios leave the possibility that the offerer sends
the (provisional) answerer stuff that the answerer doesn't want.

> If you want to support parallel forking, you need to be able to simultanously perform ICE etc with all remote peers.

Right.  You can almost get that by pretending that it's just one
remote peer with lots of candidates.  You can at least get to the
consent point by doing that.  But you aren't going to nominate more
than one pair.  Similarly, this will only generate a single DTLS
session.

Media is a complete mess if you manage to get that far.  Unless you
like scenarios were A and hear B and C, but B and C can't hear each
other and other such joys (there's a consent issue too, but it's a
minor issue of scale).

>>Both of these complicate the API significantly.  I'd rather see the
>>complexity pushed to the applications that need this complexity.
>
> I want to be able to support forking. Exactly HOW it's done can be discussed.

One approach is to create as many RTCPeerConnection objects as you
feel might be needed.  Then it becomes YOUR problem, not our problem.
And those are the sorts of problem I like ;)  (I might be interested
in forking too, but I'm happy to keep my problems private.)

> Also, I can live without support of parallel forking, and I don't really care if serial forking support is provided using cloning or being able to update the remote descriptor (using PRANSWER, or something else).

Serial forking is potentially far easier to support by doing a series
of O/A exchanges:
10: setLocal(offer)
20: get next answer
30: setRemote(answer)
40: goto 10
There's the small matter of having no guarantee that the offer will
set successfully, but I'd consider that an unlikely scenario in
practice.

>>That is, unless you are willing to consider a better API that makes
>>these issues SEP.
>
> You have one in mind? ;)

I am not at liberty to disclose that information.

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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I'm going to get back to this thread at some point but as we pointed out be=
fore, some of the text on this in the draft is just wrong. I'm trying to so=
rt out some of the key things on what basic SDP looks like for rtcweb befor=
e getting back to the PRANSWER stuff.=20


On Oct 2, 2012, at 12:00 PM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> =20
> Hi,
> =20
> One of the arguments for the Microsoft API proposal was regarding the JSE=
P usage of SDP Offer/Answer.
> =20
> There has also been ideas about JSEP "relaxing" the SDP O/A rules.
> =20
> What is the current status of that? Is the intention to "relax" something=
?
> =20
> Regards,
> =20
> Christer
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Hi,

>> I don't think the reason for PRANSWER comes from SDP O/A as such. It's m=
ore about supporting serial forking.
>
> Not my point.  My point is that the desired functionality (apply the
> answer without invalidating the offer) cannot be easily provided using
> SDP O/A.  Most scenarios leave the possibility that the offerer sends
> the (provisional) answerer stuff that the answerer doesn't want.
>
>> If you want to support parallel forking, you need to be able to simultan=
ously perform ICE etc with all remote peers.
>
> Right.  You can almost get that by pretending that it's just one
> remote peer with lots of candidates.  You can at least get to the
> consent point by doing that.  But you aren't going to nominate more
> than one pair.  Similarly, this will only generate a single DTLS
> session.

I was thinking about the same once, but I think I found some issues (unfort=
unately they don't come to my mind at the moment).

But, it's for sure an alternative we could look into.

That would allow ICE to take place during the early session, but media migh=
t be more tricky - unless you also add the m- lines of each remote peer to =
the "pretended" single peer.

> Media is a complete mess if you manage to get that far.  Unless you
> like scenarios were A and hear B and C, but B and C can't hear each
> other and other such joys (there's a consent issue too, but it's a
> minor issue of scale).
>
>>>Both of these complicate the API significantly.  I'd rather see the
>>>complexity pushed to the applications that need this complexity.
>>
>> I want to be able to support forking. Exactly HOW it's done can be discu=
ssed.
>
> One approach is to create as many RTCPeerConnection objects as you
> feel might be needed.  Then it becomes YOUR problem, not our problem.
> And those are the sorts of problem I like ;)  (I might be interested
> in forking too, but I'm happy to keep my problems private.)
>
>> Also, I can live without support of parallel forking, and I don't really=
 care if serial forking support is provided using cloning or being able to =
update the remote descriptor (using PRANSWER, or something else).
>
> Serial forking is potentially far easier to support by doing a series
> of O/A exchanges:
> 10: setLocal(offer)
> 20: get next answer
> 30: setRemote(answer)
> 40: goto 10
> There's the small matter of having no guarantee that the offer will
> set successfully, but I'd consider that an unlikely scenario in
> practice.

Can I use the same local descriptor for every setLocal() call?

>>>That is, unless you are willing to consider a better API that makes
>>>these issues SEP.
>>
>> You have one in mind? ;)
>
> I am not at liberty to disclose that information.

Can you then at least tell me when I'll be able to buy the Microsoft Surfac=
e? :)

Regards,

Christer


From christer.holmberg@ericsson.com  Fri Oct  5 05:41:45 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, mmusic WG <mmusic@ietf.org>, Harald Tveit Alvestrand <harald@alvestrand.no>, Justin Uberti <juberti@google.com>, Eric Rescorla <ekr@rtfm.com>, Bernard Aboba <bernard_aboba@hotmail.com>, Jonathan Lennox <jonathan@vidyo.com>, Randell Jesup <rjesup@mozilla.com>
Date: Fri, 5 Oct 2012 14:41:40 +0200
Thread-Topic: path forward on bundle 
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] path forward on bundle
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Hi,

>I'm trying to figure out how to move forward on bundle stuff. Let me start=
 by summarizing what I think the three proposals are then we can try and so=
rt out pros/cons of each. I think we agree that the goals here is to have s=
omething that gets the SDP so=20
>that we can negotiate using less ports.=20
>
>I see it as we have three proposed solutions that I will call "a=3Dbundle =
same port", "a=3Dbundle different port", and  "m=3Dbundle". I'll try and de=
scribe these below to make sure we are on same page of what the three are t=
hen we can try and figurer out=20
>pros/cons of each and what to do. I'll give examples of the SDP offer for =
each but I tried and make everything simple, I'm going to ignore the RTCP m=
ux and such but I think you can see how that would get added to all the exa=
mples.=20
>
>
>First lets set the baseline of an offer that does not want to multiplex th=
e audio and video on same port=20
 >        v=3D0
 >        o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
 >        s=3D
 >        c=3DIN IP4 host.atlanta.example.com
 >        t=3D0 0
 >        m=3Daudio 49170 RTP/AVP 0=20
 >        a=3Dmid:foo
 >        a=3Drtpmap:0 PCMU/8000
 >        m=3Dvideo 49172 RTP/AVP 31=20
 >        a=3Dmid:bar
 >       a=3Drtpmap:31 H261/90000

Correct.


>Proposal "a=3Dbundle same port"
>        v=3D0
>         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>         s=3D
>         c=3DIN IP4 host.atlanta.example.com
>         t=3D0 0
>         a=3Dgroup:BUNDLE foo bar
>         m=3Daudio 49170 RTP/AVP 0=20
>         a=3Dmid:foo
>         a=3Drtpmap:0 PCMU/8000
>         m=3Dvideo 49170 RTP/AVP 31=20
>        a=3Dmid:bar
>         a=3Drtpmap:31 H261/90000
>
> In the above example, note that all the m lines for a mid identified in t=
he group:BUNLDE have the same port (49170)

Correct. This is what is currently described in draft-bundle.



>Proposal "a=3Dbundle different port"
>         v=3D0
>         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>         s=3D
>         c=3DIN IP4 host.atlanta.example.com
>         t=3D0 0
>         a=3Dgroup:BUNDLE foo bar
>         m=3Daudio 49170 RTP/AVP 0=20
>         a=3Dmid:foo
>         a=3Drtpmap:0 PCMU/8000
>         m=3Dvideo 49172 RTP/AVP 31=20
>         a=3Dmid:bar
>         a=3Drtpmap:31 H261/90000

> In the above example, note that the port numbers are the same as baseline=
 - if you device receiving this offer does not support group:BUNLDE, then t=
his is identical to the baseline. So one m line has a port of 49170 and the=
 other has a different port.=20
> Other than that, it is the same as the "a=3Dbundle same port". The semant=
ics of group:BUNLDE would be defined to be that if the SDP receiver support=
s BUNDLE and wants to use it, then it places a group:BUNDLE line in the ans=
wer and it send all the media=20
> to the mids in the in BUNDLE group to the port number identified for the =
m-line corresponding to the first mid in the BUNDLE group. In this example,=
 the first mid on the a=3Dgroup:BUNDLE lines is foo, which has a m line wit=
h a port of 49170, so both the=20
> H261 and PCMU (mids foo and bar) would go to port 49170 if the device cre=
ating the answer wanted to use BUNDLE. We don't have a draft yet that says =
exactly this thought it is very close to the one-rtp draft ( and for that m=
atter  close to bundle draft )

This is more or less Harald's original proposal.



>Proposal "m=3Dbundle"
>         v=3D0
>         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>         s=3D
>         c=3DIN IP4 host.atlanta.example.com
>         t=3D0 0
>         a=3Dgroup:BUNDLE foo bar baz=20
>         m=3Daudio 49170 RTP/AVP 0=20
>         a=3Dmid:foo
>         a=3Drtpmap:0 PCMU/8000
>         m=3Dvideo 49172 RTP/AVP 31=20
>         a=3Dmid:bar
>         a=3Drtpmap:31 H261/90000
>         m=3Dbundle 10000 RTP/AVP 0 8 97 31 32
>         a=3Dmid:baz
>         a=3Dfull-rtpmap:0 audio/PCMU/8000
>         a=3Dfull-rtpmap:31 video/H261/90000
>
>
> Note in the above example the SDP above the m=3Dbundle line is pretty muc=
h the same as the baseline + the group:BUNDLE line. The m=3Dbundle is new m=
edia type that indicates many things are bundled together on port 10000. A =
device that understood=20
> bundle would know to ignore the m lines corresponding to foo and bar mids=
 and would just use the stuff from the bas mid.=20

Correct. Later today, we are planning to submit an individual draft (ie *no=
t* a new version of draft-bundle) on this alternative, so that people have =
some text to look at.

Regards,

Christer








From harald@alvestrand.no  Fri Oct  5 05:55:41 2012
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Cc: Jonathan Lennox <jonathan@vidyo.com>, mmusic WG <mmusic@ietf.org>, Randell Jesup <rjesup@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] path forward on bundle
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On 10/05/2012 02:28 PM, Cullen Jennings (fluffy) wrote:
> I'm trying to figure out how to move forward on bundle stuff. Let me st=
art by summarizing what I think the three proposals are then we can try a=
nd sort out pros/cons of each. I think we agree that the goals here is to=
 have something that gets the SDP so that we can negotiate using less por=
ts.
>
> I see it as we have three proposed solutions that I will call "a=3Dbund=
le same port", "a=3Dbundle different port", and  "m=3Dbundle". I'll try a=
nd describe these below to make sure we are on same page of what the thre=
e are then we can try and figurer out pros/cons of each and what to do. I=
'll give examples of the SDP offer for each but I tried and make everythi=
ng simple, I'm going to ignore the RTCP mux and such but I think you can =
see how that would get added to all the examples.
>
>
> First lets set the baseline of an offer that does not want to multiplex=
 the audio and video on same port
>           v=3D0
>           o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.c=
om
>           s=3D
>           c=3DIN IP4 host.atlanta.example.com
>           t=3D0 0
>           m=3Daudio 49170 RTP/AVP 0
>           a=3Dmid:foo
>           a=3Drtpmap:0 PCMU/8000
>           m=3Dvideo 49172 RTP/AVP 31
>           a=3Dmid:bar
>           a=3Drtpmap:31 H261/90000
>
>
>
> Proposal "a=3Dbundle same port"
>           v=3D0
>           o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.c=
om
>           s=3D
>           c=3DIN IP4 host.atlanta.example.com
>           t=3D0 0
>           a=3Dgroup:BUNDLE foo bar
>           m=3Daudio 49170 RTP/AVP 0
>           a=3Dmid:foo
>           a=3Drtpmap:0 PCMU/8000
>           m=3Dvideo 49170 RTP/AVP 31
>           a=3Dmid:bar
>           a=3Drtpmap:31 H261/90000
>
> In the above example, note that all the m lines for a mid identified in=
 the group:BUNLDE have the same port (49170)
No issue so far.
>
>
>
> Proposal "a=3Dbundle different port"
>           v=3D0
>           o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.c=
om
>           s=3D
>           c=3DIN IP4 host.atlanta.example.com
>           t=3D0 0
>           a=3Dgroup:BUNDLE foo bar
>           m=3Daudio 49170 RTP/AVP 0
>           a=3Dmid:foo
>           a=3Drtpmap:0 PCMU/8000
>           m=3Dvideo 49172 RTP/AVP 31
>           a=3Dmid:bar
>           a=3Drtpmap:31 H261/90000
>
> In the above example, note that the port numbers are the same as baseli=
ne - if you device receiving this offer does not support group:BUNLDE, th=
en this is identical to the baseline. So one m line has a port of 49170 a=
nd the other has a different port. Other than that, it is the same as the=
 "a=3Dbundle same port". The semantics of group:BUNLDE would be defined t=
o be that if the SDP receiver supports BUNDLE and wants to use it, then i=
t places a group:BUNDLE line in the answer and it send all the media to t=
he mids in the in BUNDLE group to the port number identified for the m-li=
ne corresponding to the first mid in the BUNDLE group. In this example, t=
he first mid on the a=3Dgroup:BUNDLE lines is foo, which has a m line wit=
h a port of 49170, so both the H261 and PCMU (mids foo and bar) would go =
to port 49170 if the device creating the answer wanted to use BUNDLE. We =
don't have a draft yet that says exactly this thought it is very close to=
 the one-rtp draft ( and for that matter  close to bundle draft )
I believe this is the mechanism described in=20
draft-alvestrand-one-rtp-02, or at least what it was intended to say. So =

we can use the name "a=3Dgroup:together" for this without risk of confusi=
on.
>
>
>
> Proposal "m=3Dbundle"
>           v=3D0
>           o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.c=
om
>           s=3D
>           c=3DIN IP4 host.atlanta.example.com
>           t=3D0 0
>           a=3Dgroup:BUNDLE foo bar baz
>           m=3Daudio 49170 RTP/AVP 0
>           a=3Dmid:foo
>           a=3Drtpmap:0 PCMU/8000
>           m=3Dvideo 49172 RTP/AVP 31
>           a=3Dmid:bar
>           a=3Drtpmap:31 H261/90000
>           m=3Dbundle 10000 RTP/AVP 0 8 97 31 32
>           a=3Dmid:baz
>           a=3Dfull-rtpmap:0 audio/PCMU/8000
>           a=3Dfull-rtpmap:31 video/H261/90000
>
>
> Note in the above example the SDP above the m=3Dbundle line is pretty m=
uch the same as the baseline + the group:BUNDLE line. The m=3Dbundle is n=
ew media type that indicates many things are bundled together on port 100=
00. A device that understood bundle would know to ignore the m lines corr=
esponding to foo and bar mids and would just use the stuff from the bas m=
id.
Yep. I've suggested calling it "m=3Danytype", but Christer hasn't adopted=
=20
that so far :-)
>
>
> So before we start discussing the pros / cons of theses three approache=
s, let's spend a few days and make sure that I have correctly characteriz=
ed the three approaches under discussion.
>
> Did I get it right? Do we need more explanation of any of these to see =
how they work?
>
> Lets keep clarifications only on this thread and I'll start a separate =
thread for pro/cons once we get this a bit clearer.
Remember to change the subject line to a descriptive one for the next=20
thread :-)
>
>
>
>
>
>
>
>
>



From fluffy@cisco.com  Fri Oct  5 05:28:36 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: mmusic WG <mmusic@ietf.org>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Harald Tveit Alvestrand <harald@alvestrand.no>, "Justin Uberti" <juberti@google.com>, Eric Rescorla <ekr@rtfm.com>, Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard_aboba@hotmail.com>, Jonathan Lennox <jonathan@vidyo.com>, Randell Jesup <rjesup@mozilla.com>
Thread-Topic: path forward on bundle 
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] path forward on bundle
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I'm trying to figure out how to move forward on bundle stuff. Let me start =
by summarizing what I think the three proposals are then we can try and sor=
t out pros/cons of each. I think we agree that the goals here is to have so=
mething that gets the SDP so that we can negotiate using less ports.=20

I see it as we have three proposed solutions that I will call "a=3Dbundle s=
ame port", "a=3Dbundle different port", and  "m=3Dbundle". I'll try and des=
cribe these below to make sure we are on same page of what the three are th=
en we can try and figurer out pros/cons of each and what to do. I'll give e=
xamples of the SDP offer for each but I tried and make everything simple, I=
'm going to ignore the RTCP mux and such but I think you can see how that w=
ould get added to all the examples.=20


First lets set the baseline of an offer that does not want to multiplex the=
 audio and video on same port=20
         v=3D0
         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
         s=3D
         c=3DIN IP4 host.atlanta.example.com
         t=3D0 0
         m=3Daudio 49170 RTP/AVP 0=20
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49172 RTP/AVP 31=20
         a=3Dmid:bar
         a=3Drtpmap:31 H261/90000



Proposal "a=3Dbundle same port"
         v=3D0
         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
         s=3D
         c=3DIN IP4 host.atlanta.example.com
         t=3D0 0
         a=3Dgroup:BUNDLE foo bar
         m=3Daudio 49170 RTP/AVP 0=20
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49170 RTP/AVP 31=20
         a=3Dmid:bar
         a=3Drtpmap:31 H261/90000

In the above example, note that all the m lines for a mid identified in the=
 group:BUNLDE have the same port (49170)



Proposal "a=3Dbundle different port"
         v=3D0
         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
         s=3D
         c=3DIN IP4 host.atlanta.example.com
         t=3D0 0
         a=3Dgroup:BUNDLE foo bar
         m=3Daudio 49170 RTP/AVP 0=20
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49172 RTP/AVP 31=20
         a=3Dmid:bar
         a=3Drtpmap:31 H261/90000

In the above example, note that the port numbers are the same as baseline -=
 if you device receiving this offer does not support group:BUNLDE, then thi=
s is identical to the baseline. So one m line has a port of 49170 and the o=
ther has a different port. Other than that, it is the same as the "a=3Dbund=
le same port". The semantics of group:BUNLDE would be defined to be that if=
 the SDP receiver supports BUNDLE and wants to use it, then it places a gro=
up:BUNDLE line in the answer and it send all the media to the mids in the i=
n BUNDLE group to the port number identified for the m-line corresponding t=
o the first mid in the BUNDLE group. In this example, the first mid on the =
a=3Dgroup:BUNDLE lines is foo, which has a m line with a port of 49170, so =
both the H261 and PCMU (mids foo and bar) would go to port 49170 if the dev=
ice creating the answer wanted to use BUNDLE. We don't have a draft yet tha=
t says exactly this thought it is very close to the one-rtp draft ( and for=
 that matter  close to bundle draft )



Proposal "m=3Dbundle"
         v=3D0
         o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
         s=3D
         c=3DIN IP4 host.atlanta.example.com
         t=3D0 0
         a=3Dgroup:BUNDLE foo bar baz=20
         m=3Daudio 49170 RTP/AVP 0=20
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49172 RTP/AVP 31=20
         a=3Dmid:bar
         a=3Drtpmap:31 H261/90000
         m=3Dbundle 10000 RTP/AVP 0 8 97 31 32
         a=3Dmid:baz
         a=3Dfull-rtpmap:0 audio/PCMU/8000
         a=3Dfull-rtpmap:31 video/H261/90000


Note in the above example the SDP above the m=3Dbundle line is pretty much =
the same as the baseline + the group:BUNDLE line. The m=3Dbundle is new med=
ia type that indicates many things are bundled together on port 10000. A de=
vice that understood bundle would know to ignore the m lines corresponding =
to foo and bar mids and would just use the stuff from the bas mid.=20


So before we start discussing the pros / cons of theses three approaches, l=
et's spend a few days and make sure that I have correctly characterized the=
 three approaches under discussion.=20

Did I get it right? Do we need more explanation of any of these to see how =
they work?=20

Lets keep clarifications only on this thread and I'll start a separate thre=
ad for pro/cons once we get this a bit clearer.=20










From harald@alvestrand.no  Fri Oct  5 10:11:10 2012
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Subject: Re: [rtcweb] Resolving whether to support additional key management methods
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On 10/04/2012 05:32 PM, Ted Hardie wrote:
> The Chairs would like to foster resolution of the long-running
> discussion of support for additional key management methods, e.g. as
> described in draft-ohlsson-rtcweb-sdes-support and the meeting
> materials supporting draft-ietf-avtcore-srtp-ekt.   Working group
> discussion of this matter is, of course, welcome, but the chairs are
> also considering a virtual interim meeting in January of 2013, focused
> on this specific topic.  Comments on this plan or on the relevant
> approaches is welcome.

What is the proposed outcome space of this discussion?
We already have a decision of "MUST do DTLS(-SRTP)", but I'm not sure 
what the other questions are.

I could see the result of the SDES discussion being one of:

- MUST do SDES
- SHOULD do SDES
- MAY do SDES
- MUST NOT do SDES

I also see another discussion around EKT, which could conclude

- MUST do EKT
- MAY do EKT
- MUST NOT do EKT

(I don't see much point in "SHOULD do EKT" - it doesn't give 
interoperability, so is not much more useful than MAY)

Are there other things on the table? Are both of the above on the table?
>
> regards,
>
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Fri Oct  5 10:13:14 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 10/04/2012 08:21 PM, Christer Holmberg wrote:
> Hi,
>
>>> For the Web use case, I think PRANSWER is an unnecessary distraction; if
>>> we're writing both initiator and responder, multiple O/A rounds, or use of
>>> multiple PeerConnections, is a much cleaner solution for the use cases I can
>>> wrap my head around.
>> I'd remove that qualifier.  We currently have a draft that describes
>> both cloning AND PRANSWER, both of which are more cleanly addressed by
>> either multiple O/A rounds or multiple independent sessions.
>>
>> PRANSWER:
>> The ability to describe a session that includes receipt capabilities
>> that are a superset of send capabilities.  SDP O/A doesn't really
>> provide a way for me to describe this because the default assumption
>> is that items not appearing in the answer are no longer needed.
>> With something like bundle involved, we could split m= lines into
>> sendonly and recvonly portions to allow for this case to be provided.
> I don't think the reason for PRANSWER comes from SDP O/A as such. It's more about supporting serial forking.
I've asked that question before, but I don't remember seeing an answer 
from people who know what they're talking about:

Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, 
or does it not violate the SDP O/A model?

I don't understand how it, in the form that has been described as "what 
we have to support", can be SDP O/A compatible, but then there are many 
things about SIP that I don't understand, so I may understand it when 
it's explained to me.

               Harald


From martin.thomson@gmail.com  Fri Oct  5 10:14:18 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 5 October 2012 00:39, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>>> If you want to support parallel forking, you need to be able to simultanously perform ICE etc with all remote peers.
>>
>> Right.  You can almost get that by pretending that it's just one
>> remote peer with lots of candidates.  You can at least get to the
>> consent point by doing that.  But you aren't going to nominate more
>> than one pair.  Similarly, this will only generate a single DTLS
>> session.
>
> I was thinking about the same once, but I think I found some issues (unfortunately they don't come to my mind at the moment).

It's probably not going to be flawless, true.  As you say, media would
be a little crazy.  You might be able to get one media stream out of
it - not multiple.

> Can I use the same local descriptor for every setLocal() call?

My experience suggests that you can.  However, that's not stipulated
in the API specification, so it's not an ironclad guarantee.

> Can you then at least tell me when I'll be able to buy the Microsoft Surface? :)

Yes.  When they release it.  October 25 apparently.

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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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--f46d042f9f2a4639ea04cb5373d7
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On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

> I've asked that question before, but I don't remember seeing an answer
> from people who know what they're talking about:
>
> Does serial forking, as practiced by SIP UAs, violate the SDP O/A model,
> or does it not violate the SDP O/A model?
>
> I don't understand how it, in the form that has been described as "what we
> have to support", can be SDP O/A compatible, but then there are many things
> about SIP that I don't understand, so I may understand it when it's
> explained to me.
>
>
O/A does not describe multiple answers to a single offer. Nothing
explicitly prohibits it there but I would argue this is not part of this
specification. No standard document that I am aware of describes how serial
O/A is supposed to work. Serial forking as practiced by SIP UA does violate
SIP RFC 3261 which states that each dialog can only have one answer to each
offer. If answer is sent in both provisional and final response, it should
be the same. You can, however, create multiple dialogs with the same offer.
This normally implies parallel forking, but the most common use case is
with early media, where you end up with multiple early dialogs. For
instance you call a phone number, phone network sends you SDP in early
media and plays a dial tone, then the called number does not answer, and
you get connected to a voice mail which uses a completely different SDP in
final answer. According to SIP these answers should come with different
to-tags and technically would be parts of two different dialogs. What makes
this a bit tricky is when the phone network and the voice mail are behind
SBC (or some other sort of B2BUA) you only see one dialog which send you
multiple answers to the same offer. This scenario is so common that it is
more likely to be implemented then parallel forking. This is why PRANSWER
was suggested.

If cloning can be implemented in WebRTC it would be more standard compliant
then PRANSWER and would allow for a lot more use cases. Typical difficulty
in parallel forking implementation in SIP is due to RTP from multiple
sources coming to the same IP and port with no consent or notification to
the receiving side. This makes it very difficult to present this media to
the user. There is no way to tie media to actual dialogs since RTP can come
from different IP and ports then specified in answer SDPs. This is not an
issue with WebRTC where consent to receive media is required. I would argue
let's implement cloning and not waste any more time on PRANSWER which in my
opinion will always stay a half working hack.
_____________
Roman Shpount

--f46d042f9f2a4639ea04cb5373d7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Oct 5, 2012 at 1:13 PM=
, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestra=
nd.no" target=3D"_blank">harald@alvestrand.no</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">
I&#39;ve asked that question before, but I don&#39;t remember seeing an ans=
wer from people who know what they&#39;re talking about:<br>
<br>
Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, or=
 does it not violate the SDP O/A model?<br>
<br>
I don&#39;t understand how it, in the form that has been described as &quot=
;what we have to support&quot;, can be SDP O/A compatible, but then there a=
re many things about SIP that I don&#39;t understand, so I may understand i=
t when it&#39;s explained to me.<span class=3D"HOEnZb"><font color=3D"#8888=
88"><br>

</font></span><br></blockquote><div><br>O/A does not describe multiple answ=
ers to a single offer. Nothing explicitly prohibits it there but I would ar=
gue this is not part of this specification. No standard document that I am =
aware of describes how serial O/A is supposed to work. Serial forking as pr=
acticed by SIP UA does violate SIP RFC 3261 which states that each dialog c=
an only have one answer to each offer. If answer is sent in both provisiona=
l and final response, it should be the same. You can, however, create multi=
ple dialogs with the same offer. This normally implies parallel forking, bu=
t the most common use case is with early media, where you end up with multi=
ple early dialogs. For instance you call a phone number, phone network send=
s you SDP in early media and plays a dial tone, then the called number does=
 not answer, and you get connected to a voice mail which uses a completely =
different SDP in final answer. According to SIP these answers should come w=
ith different to-tags and technically would be parts of two different dialo=
gs. What makes this a bit tricky is when the phone network and the voice ma=
il are behind SBC (or some other sort of B2BUA) you only see one dialog whi=
ch send you multiple answers to the same offer. This scenario is so common =
that it is more likely to be implemented then parallel forking. This is why=
 PRANSWER was suggested. <br>
<br>If cloning can be implemented in WebRTC it would be more standard compl=
iant then PRANSWER and would allow for a lot more use cases. Typical diffic=
ulty in parallel forking implementation in SIP is due to RTP from multiple =
sources coming to the same IP and port with no consent or notification to t=
he receiving side. This makes it very difficult to present this media to th=
e user. There is no way to tie media to actual dialogs since RTP can come f=
rom different IP and ports then specified in answer SDPs. This is not an is=
sue with WebRTC where consent to receive media is required. I would argue l=
et&#39;s implement cloning and not waste any more time on PRANSWER which in=
 my opinion will always stay a half working hack.<br>
</div></div>_____________<br>Roman Shpount<br>
<br>

--f46d042f9f2a4639ea04cb5373d7--

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From: Ted Hardie <ted.ietf@gmail.com>
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On Fri, Oct 5, 2012 at 10:11 AM, Harald Alvestrand <harald@alvestrand.no> wrote:
> On 10/04/2012 05:32 PM, Ted Hardie wrote:

> What is the proposed outcome space of this discussion?
> We already have a decision of "MUST do DTLS(-SRTP)",

Yes, and we are not proposing to re-discuss whether this a MUST; we
want to focus on whether there are additional proposals which have
consensus.


> but I'm not sure what
> the other questions are.
>
> I could see the result of the SDES discussion being one of:
>
> - MUST do SDES
> - SHOULD do SDES
> - MAY do SDES
> - MUST NOT do SDES
>
> I also see another discussion around EKT, which could conclude
>
> - MUST do EKT
> - MAY do EKT
> - MUST NOT do EKT
>
> (I don't see much point in "SHOULD do EKT" - it doesn't give
> interoperability, so is not much more useful than MAY)
>
> Are there other things on the table? Are both of the above on the table?
>

Both of the above discussions have already started, and they have not
concluded; we want to conclude them.  If there are other proposals,
they can be brought forward in the context of this discussion,
provided they do not start with "let's ditch DTLS-(SRTP)".

regards,

Ted


>>
>> regards,
>>
>> Ted, Cullen, Magnus
>> _______________________________________________
>> 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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To: Harald Alvestrand <harald@alvestrand.no>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, Jonathan Lennox <jonathan@vidyo.com>, Randell Jesup <rjesup@mozilla.com>, mmusic WG <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] path forward on bundle
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"Note in the above example the SDP above the m=bundle line is pretty
much the same as the baseline + the group:BUNDLE line. The m=bundle is
new media type that indicates many things are bundled together on port
10000. A device that understood bundle would know to ignore the m
lines corresponding to foo and bar mids and would just use the stuff
from the bas mid."
^^^

Should it be baz mid?

/Kaiduan

On Fri, Oct 5, 2012 at 8:55 AM, Harald Alvestrand <harald@alvestrand.no> wrote:
> On 10/05/2012 02:28 PM, Cullen Jennings (fluffy) wrote:
>>
>> I'm trying to figure out how to move forward on bundle stuff. Let me start
>> by summarizing what I think the three proposals are then we can try and sort
>> out pros/cons of each. I think we agree that the goals here is to have
>> something that gets the SDP so that we can negotiate using less ports.
>>
>> I see it as we have three proposed solutions that I will call "a=bundle
>> same port", "a=bundle different port", and  "m=bundle". I'll try and
>> describe these below to make sure we are on same page of what the three are
>> then we can try and figurer out pros/cons of each and what to do. I'll give
>> examples of the SDP offer for each but I tried and make everything simple,
>> I'm going to ignore the RTCP mux and such but I think you can see how that
>> would get added to all the examples.
>>
>>
>> First lets set the baseline of an offer that does not want to multiplex
>> the audio and video on same port
>>           v=0
>>           o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>>           s=
>>           c=IN IP4 host.atlanta.example.com
>>           t=0 0
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49172 RTP/AVP 31
>>           a=mid:bar
>>           a=rtpmap:31 H261/90000
>>
>>
>>
>> Proposal "a=bundle same port"
>>           v=0
>>           o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>>           s=
>>           c=IN IP4 host.atlanta.example.com
>>           t=0 0
>>           a=group:BUNDLE foo bar
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49170 RTP/AVP 31
>>           a=mid:bar
>>           a=rtpmap:31 H261/90000
>>
>> In the above example, note that all the m lines for a mid identified in
>> the group:BUNLDE have the same port (49170)
>
> No issue so far.
>
>>
>>
>>
>> Proposal "a=bundle different port"
>>           v=0
>>           o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>>           s=
>>           c=IN IP4 host.atlanta.example.com
>>           t=0 0
>>           a=group:BUNDLE foo bar
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49172 RTP/AVP 31
>>           a=mid:bar
>>           a=rtpmap:31 H261/90000
>>
>> In the above example, note that the port numbers are the same as baseline
>> - if you device receiving this offer does not support group:BUNLDE, then
>> this is identical to the baseline. So one m line has a port of 49170 and the
>> other has a different port. Other than that, it is the same as the "a=bundle
>> same port". The semantics of group:BUNLDE would be defined to be that if the
>> SDP receiver supports BUNDLE and wants to use it, then it places a
>> group:BUNDLE line in the answer and it send all the media to the mids in the
>> in BUNDLE group to the port number identified for the m-line corresponding
>> to the first mid in the BUNDLE group. In this example, the first mid on the
>> a=group:BUNDLE lines is foo, which has a m line with a port of 49170, so
>> both the H261 and PCMU (mids foo and bar) would go to port 49170 if the
>> device creating the answer wanted to use BUNDLE. We don't have a draft yet
>> that says exactly this thought it is very close to the one-rtp draft ( and
>> for that matter  clo
>
> se to bundle draft )
> I believe this is the mechanism described in draft-alvestrand-one-rtp-02, or
> at least what it was intended to say. So we can use the name
> "a=group:together" for this without risk of confusion.
>
>>
>>
>>
>> Proposal "m=bundle"
>>           v=0
>>           o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
>>           s=
>>           c=IN IP4 host.atlanta.example.com
>>           t=0 0
>>           a=group:BUNDLE foo bar baz
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49172 RTP/AVP 31
>>           a=mid:bar
>>           a=rtpmap:31 H261/90000
>>           m=bundle 10000 RTP/AVP 0 8 97 31 32
>>           a=mid:baz
>>           a=full-rtpmap:0 audio/PCMU/8000
>>           a=full-rtpmap:31 video/H261/90000
>>
>>
>> Note in the above example the SDP above the m=bundle line is pretty much
>> the same as the baseline + the group:BUNDLE line. The m=bundle is new media
>> type that indicates many things are bundled together on port 10000. A device
>> that understood bundle would know to ignore the m lines corresponding to foo
>> and bar mids and would just use the stuff from the bas mid.
>
> Yep. I've suggested calling it "m=anytype", but Christer hasn't adopted that
> so far :-)
>
>>
>>
>> So before we start discussing the pros / cons of theses three approaches,
>> let's spend a few days and make sure that I have correctly characterized the
>> three approaches under discussion.
>>
>> Did I get it right? Do we need more explanation of any of these to see how
>> they work?
>>
>> Lets keep clarifications only on this thread and I'll start a separate
>> thread for pro/cons once we get this a bit clearer.
>
> Remember to change the subject line to a descriptive one for the next thread
> :-)
>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Kaiduan Xie <kaiduanx@gmail.com>
Thread-Topic: [rtcweb] path forward on bundle
Thread-Index: AQHNoyaEp4nhhdwyxEuzOdZlbOlWD5erYa2A
Date: Fri, 5 Oct 2012 18:48:59 +0000
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References: <C5E08FE080ACFD4DAE31E4BDBF944EB111869DCB@xmb-aln-x02.cisco.com> <506ED8C4.6080409@alvestrand.no> <CACKRbQejJDt0nd4OuZHrHcC48PZzOJRvKbiQUR1=ee4CxmcOZQ@mail.gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Randell Jesup <rjesup@mozilla.com>, mmusic WG <mmusic@ietf.org>, Jonathan Lennox <jonathan@vidyo.com>
Subject: Re: [rtcweb] path forward on bundle
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On Oct 5, 2012, at 12:23 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

> "Note in the above example the SDP above the m=3Dbundle line is pretty
> much the same as the baseline + the group:BUNDLE line. The m=3Dbundle is
> new media type that indicates many things are bundled together on port
> 10000. A device that understood bundle would know to ignore the m
> lines corresponding to foo and bar mids and would just use the stuff
> from the bas mid."
> ^^^
>=20
> Should it be baz mid?

oops - yes - that should be baz , sorry



>=20
> /Kaiduan
>=20
> On Fri, Oct 5, 2012 at 8:55 AM, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>> On 10/05/2012 02:28 PM, Cullen Jennings (fluffy) wrote:
>>>=20
>>> I'm trying to figure out how to move forward on bundle stuff. Let me st=
art
>>> by summarizing what I think the three proposals are then we can try and=
 sort
>>> out pros/cons of each. I think we agree that the goals here is to have
>>> something that gets the SDP so that we can negotiate using less ports.
>>>=20
>>> I see it as we have three proposed solutions that I will call "a=3Dbund=
le
>>> same port", "a=3Dbundle different port", and  "m=3Dbundle". I'll try an=
d
>>> describe these below to make sure we are on same page of what the three=
 are
>>> then we can try and figurer out pros/cons of each and what to do. I'll =
give
>>> examples of the SDP offer for each but I tried and make everything simp=
le,
>>> I'm going to ignore the RTCP mux and such but I think you can see how t=
hat
>>> would get added to all the examples.
>>>=20
>>>=20
>>> First lets set the baseline of an offer that does not want to multiplex
>>> the audio and video on same port
>>>          v=3D0
>>>          o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.co=
m
>>>          s=3D
>>>          c=3DIN IP4 host.atlanta.example.com
>>>          t=3D0 0
>>>          m=3Daudio 49170 RTP/AVP 0
>>>          a=3Dmid:foo
>>>          a=3Drtpmap:0 PCMU/8000
>>>          m=3Dvideo 49172 RTP/AVP 31
>>>          a=3Dmid:bar
>>>          a=3Drtpmap:31 H261/90000
>>>=20
>>>=20
>>>=20
>>> Proposal "a=3Dbundle same port"
>>>          v=3D0
>>>          o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.co=
m
>>>          s=3D
>>>          c=3DIN IP4 host.atlanta.example.com
>>>          t=3D0 0
>>>          a=3Dgroup:BUNDLE foo bar
>>>          m=3Daudio 49170 RTP/AVP 0
>>>          a=3Dmid:foo
>>>          a=3Drtpmap:0 PCMU/8000
>>>          m=3Dvideo 49170 RTP/AVP 31
>>>          a=3Dmid:bar
>>>          a=3Drtpmap:31 H261/90000
>>>=20
>>> In the above example, note that all the m lines for a mid identified in
>>> the group:BUNLDE have the same port (49170)
>>=20
>> No issue so far.
>>=20
>>>=20
>>>=20
>>>=20
>>> Proposal "a=3Dbundle different port"
>>>          v=3D0
>>>          o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.co=
m
>>>          s=3D
>>>          c=3DIN IP4 host.atlanta.example.com
>>>          t=3D0 0
>>>          a=3Dgroup:BUNDLE foo bar
>>>          m=3Daudio 49170 RTP/AVP 0
>>>          a=3Dmid:foo
>>>          a=3Drtpmap:0 PCMU/8000
>>>          m=3Dvideo 49172 RTP/AVP 31
>>>          a=3Dmid:bar
>>>          a=3Drtpmap:31 H261/90000
>>>=20
>>> In the above example, note that the port numbers are the same as baseli=
ne
>>> - if you device receiving this offer does not support group:BUNLDE, the=
n
>>> this is identical to the baseline. So one m line has a port of 49170 an=
d the
>>> other has a different port. Other than that, it is the same as the "a=
=3Dbundle
>>> same port". The semantics of group:BUNLDE would be defined to be that i=
f the
>>> SDP receiver supports BUNDLE and wants to use it, then it places a
>>> group:BUNDLE line in the answer and it send all the media to the mids i=
n the
>>> in BUNDLE group to the port number identified for the m-line correspond=
ing
>>> to the first mid in the BUNDLE group. In this example, the first mid on=
 the
>>> a=3Dgroup:BUNDLE lines is foo, which has a m line with a port of 49170,=
 so
>>> both the H261 and PCMU (mids foo and bar) would go to port 49170 if the
>>> device creating the answer wanted to use BUNDLE. We don't have a draft =
yet
>>> that says exactly this thought it is very close to the one-rtp draft ( =
and
>>> for that matter  clo
>>=20
>> se to bundle draft )
>> I believe this is the mechanism described in draft-alvestrand-one-rtp-02=
, or
>> at least what it was intended to say. So we can use the name
>> "a=3Dgroup:together" for this without risk of confusion.
>>=20
>>>=20
>>>=20
>>>=20
>>> Proposal "m=3Dbundle"
>>>          v=3D0
>>>          o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.co=
m
>>>          s=3D
>>>          c=3DIN IP4 host.atlanta.example.com
>>>          t=3D0 0
>>>          a=3Dgroup:BUNDLE foo bar baz
>>>          m=3Daudio 49170 RTP/AVP 0
>>>          a=3Dmid:foo
>>>          a=3Drtpmap:0 PCMU/8000
>>>          m=3Dvideo 49172 RTP/AVP 31
>>>          a=3Dmid:bar
>>>          a=3Drtpmap:31 H261/90000
>>>          m=3Dbundle 10000 RTP/AVP 0 8 97 31 32
>>>          a=3Dmid:baz
>>>          a=3Dfull-rtpmap:0 audio/PCMU/8000
>>>          a=3Dfull-rtpmap:31 video/H261/90000
>>>=20
>>>=20
>>> Note in the above example the SDP above the m=3Dbundle line is pretty m=
uch
>>> the same as the baseline + the group:BUNDLE line. The m=3Dbundle is new=
 media
>>> type that indicates many things are bundled together on port 10000. A d=
evice
>>> that understood bundle would know to ignore the m lines corresponding t=
o foo
>>> and bar mids and would just use the stuff from the bas mid.
>>=20
>> Yep. I've suggested calling it "m=3Danytype", but Christer hasn't adopte=
d that
>> so far :-)
>>=20
>>>=20
>>>=20
>>> So before we start discussing the pros / cons of theses three approache=
s,
>>> let's spend a few days and make sure that I have correctly characterize=
d the
>>> three approaches under discussion.
>>>=20
>>> Did I get it right? Do we need more explanation of any of these to see =
how
>>> they work?
>>>=20
>>> Lets keep clarifications only on this thread and I'll start a separate
>>> thread for pro/cons once we get this a bit clearer.
>>=20
>> Remember to change the subject line to a descriptive one for the next th=
read
>> :-)
>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Fri Oct  5 14:26:41 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Harald Alvestrand <harald@alvestrand.no>
Date: Fri, 5 Oct 2012 23:26:38 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

Roman is correct.

The SDP O/A doesn't consider forking, because each forked SIP leg creates a=
 unique early dialog, and each early dialog has its own O/A state machine.

Regards,

Christer


________________________________
From: Roman Shpount [roman@telurix.com]
Sent: Friday, October 05, 2012 8:46 PM
To: Harald Alvestrand
Cc: Christer Holmberg; rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?


On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand <harald@alvestrand.no<mai=
lto:harald@alvestrand.no>> wrote:
I've asked that question before, but I don't remember seeing an answer from=
 people who know what they're talking about:

Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, or=
 does it not violate the SDP O/A model?

I don't understand how it, in the form that has been described as "what we =
have to support", can be SDP O/A compatible, but then there are many things=
 about SIP that I don't understand, so I may understand it when it's explai=
ned to me.


O/A does not describe multiple answers to a single offer. Nothing explicitl=
y prohibits it there but I would argue this is not part of this specificati=
on. No standard document that I am aware of describes how serial O/A is sup=
posed to work. Serial forking as practiced by SIP UA does violate SIP RFC 3=
261 which states that each dialog can only have one answer to each offer. I=
f answer is sent in both provisional and final response, it should be the s=
ame. You can, however, create multiple dialogs with the same offer. This no=
rmally implies parallel forking, but the most common use case is with early=
 media, where you end up with multiple early dialogs. For instance you call=
 a phone number, phone network sends you SDP in early media and plays a dia=
l tone, then the called number does not answer, and you get connected to a =
voice mail which uses a completely different SDP in final answer. According=
 to SIP these answers should come with different to-tags and technically wo=
uld be parts of two different dialogs. What makes this a bit tricky is when=
 the phone network and the voice mail are behind SBC (or some other sort of=
 B2BUA) you only see one dialog which send you multiple answers to the same=
 offer. This scenario is so common that it is more likely to be implemented=
 then parallel forking. This is why PRANSWER was suggested.

If cloning can be implemented in WebRTC it would be more standard compliant=
 then PRANSWER and would allow for a lot more use cases. Typical difficulty=
 in parallel forking implementation in SIP is due to RTP from multiple sour=
ces coming to the same IP and port with no consent or notification to the r=
eceiving side. This makes it very difficult to present this media to the us=
er. There is no way to tie media to actual dialogs since RTP can come from =
different IP and ports then specified in answer SDPs. This is not an issue =
with WebRTC where consent to receive media is required. I would argue let's=
 implement cloning and not waste any more time on PRANSWER which in my opin=
ion will always stay a half working hack.
_____________
Roman Shpount


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Date: Fri, 5 Oct 2012 23:26:59 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

>>>> If you want to support parallel forking, you need to be able to simult=
anously perform ICE etc with all remote peers.
>>>
>>> Right.  You can almost get that by pretending that it's just one
>>> remote peer with lots of candidates.  You can at least get to the
>>> consent point by doing that.  But you aren't going to nominate more
>>> than one pair.  Similarly, this will only generate a single DTLS
>>> session.
>>
>> I was thinking about the same once, but I think I found some issues (unf=
ortunately they don't come to my mind at the moment).
>
> It's probably not going to be flawless, true.  As you say, media would
> be a little crazy.  You might be able to get one media stream out of
> it - not multiple.

And, I guess it would still require to be able to update the answer, as add=
itional early dialogs (with associated remote ICE candidates) may be create=
d.

>> Can I use the same local descriptor for every setLocal() call?
>
> My experience suggests that you can.  However, that's not stipulated
> in the API specification, so it's not an ironclad guarantee.

If it doesn't, and a new local descripor is created, you would need to send=
 that one to the remote endpoint.

Regards,

Christer=

From martin.thomson@gmail.com  Fri Oct  5 16:19:59 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 5 October 2012 14:26, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>>> Can I use the same local descriptor for every setLocal() call?
>>
>> My experience suggests that you can.  However, that's not stipulated
>> in the API specification, so it's not an ironclad guarantee.
>
> If it doesn't, and a new local descripor is created, you would need to send that one to the remote endpoint.

You don't create a new description, you just set one.  The only
problem arises when you can't set the local description because
something broke since you last set it.  (Maybe setting the answer
caused the browser to release something that you need to set the
offer.)

If that happens, you are hosed.  You could try to make a new
PeerConnection, but you'll have to resend your INVITE because you will
get a new set of candidates, ufrag, pwd, etc...

From partha@parthasarathi.co.in  Fri Oct  5 22:56:33 2012
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We discussed earlier that PRANSWER states complicates SIP offer/answer
mechanism as the mapping with SIP offer/answer is not well defined. For
Example: "UPDATE with SDP from SIP UAS after sending 18x with SDP has to be
considered as PRANSWER or OFFER or ANSWER in originating JSEP side" ?

I agree with Roman that Cloning is the solution for serial forking as well.

Thanks
Partha  

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Christer Holmberg
Sent: Saturday, October 06, 2012 2:57 AM
To: Roman Shpount; Harald Alvestrand
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?

Hi,

Roman is correct.

The SDP O/A doesn't consider forking, because each forked SIP leg creates a
unique early dialog, and each early dialog has its own O/A state machine.

Regards,

Christer


________________________________
From: Roman Shpount [roman@telurix.com]
Sent: Friday, October 05, 2012 8:46 PM
To: Harald Alvestrand
Cc: Christer Holmberg; rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?


On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand
<harald@alvestrand.no<mailto:harald@alvestrand.no>> wrote:
I've asked that question before, but I don't remember seeing an answer from
people who know what they're talking about:

Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, or
does it not violate the SDP O/A model?

I don't understand how it, in the form that has been described as "what we
have to support", can be SDP O/A compatible, but then there are many things
about SIP that I don't understand, so I may understand it when it's
explained to me.


O/A does not describe multiple answers to a single offer. Nothing explicitly
prohibits it there but I would argue this is not part of this specification.
No standard document that I am aware of describes how serial O/A is supposed
to work. Serial forking as practiced by SIP UA does violate SIP RFC 3261
which states that each dialog can only have one answer to each offer. If
answer is sent in both provisional and final response, it should be the
same. You can, however, create multiple dialogs with the same offer. This
normally implies parallel forking, but the most common use case is with
early media, where you end up with multiple early dialogs. For instance you
call a phone number, phone network sends you SDP in early media and plays a
dial tone, then the called number does not answer, and you get connected to
a voice mail which uses a completely different SDP in final answer.
According to SIP these answers should come with different to-tags and
technically would be parts of
  two different dialogs. What makes this a bit tricky is when the phone
network and the voice mail are behind SBC (or some other sort of B2BUA) you
only see one dialog which send you multiple answers to the same offer. This
scenario is so common that it is more likely to be implemented then parallel
forking. This is why PRANSWER was suggested.

If cloning can be implemented in WebRTC it would be more standard compliant
then PRANSWER and would allow for a lot more use cases. Typical difficulty
in parallel forking implementation in SIP is due to RTP from multiple
sources coming to the same IP and port with no consent or notification to
the receiving side. This makes it very difficult to present this media to
the user. There is no way to tie media to actual dialogs since RTP can come
from different IP and ports then specified in answer SDPs. This is not an
issue with WebRTC where consent to receive media is required. I would argue
let's implement cloning and not waste any more time on PRANSWER which in my
opinion will always stay a half working hack.
_____________
Roman Shpount

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


From christer.holmberg@ericsson.com  Sat Oct  6 11:28:57 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, 'Roman Shpount' <roman@telurix.com>, 'Harald Alvestrand' <harald@alvestrand.no>
Date: Sat, 6 Oct 2012 20:24:40 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

>We discussed earlier that PRANSWER states complicates SIP offer/answer
>mechanism as the mapping with SIP offer/answer is not well defined. For
>Example: "UPDATE with SDP from SIP UAS after sending 18x with SDP has to b=
e
>considered as PRANSWER or OFFER or ANSWER in originating JSEP side" ?
>
>I agree with Roman that Cloning is the solution for serial forking as well=
.

If we support cloning (no matter whether it's a separate function call, or =
simply creating a new session with the same local descriptor), I agree that=
 PRANSWER probably could be removed.

At least I have never thought of any other use of it than to support serial=
 forking.

Regards,

Christer




-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Christer Holmberg
Sent: Saturday, October 06, 2012 2:57 AM
To: Roman Shpount; Harald Alvestrand
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?

Hi,

Roman is correct.

The SDP O/A doesn't consider forking, because each forked SIP leg creates a
unique early dialog, and each early dialog has its own O/A state machine.

Regards,

Christer


________________________________
From: Roman Shpount [roman@telurix.com]
Sent: Friday, October 05, 2012 8:46 PM
To: Harald Alvestrand
Cc: Christer Holmberg; rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?


On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand
<harald@alvestrand.no<mailto:harald@alvestrand.no>> wrote:
I've asked that question before, but I don't remember seeing an answer from
people who know what they're talking about:

Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, or
does it not violate the SDP O/A model?

I don't understand how it, in the form that has been described as "what we
have to support", can be SDP O/A compatible, but then there are many things
about SIP that I don't understand, so I may understand it when it's
explained to me.


O/A does not describe multiple answers to a single offer. Nothing explicitl=
y
prohibits it there but I would argue this is not part of this specification=
.
No standard document that I am aware of describes how serial O/A is suppose=
d
to work. Serial forking as practiced by SIP UA does violate SIP RFC 3261
which states that each dialog can only have one answer to each offer. If
answer is sent in both provisional and final response, it should be the
same. You can, however, create multiple dialogs with the same offer. This
normally implies parallel forking, but the most common use case is with
early media, where you end up with multiple early dialogs. For instance you
call a phone number, phone network sends you SDP in early media and plays a
dial tone, then the called number does not answer, and you get connected to
a voice mail which uses a completely different SDP in final answer.
According to SIP these answers should come with different to-tags and
technically would be parts of
  two different dialogs. What makes this a bit tricky is when the phone
network and the voice mail are behind SBC (or some other sort of B2BUA) you
only see one dialog which send you multiple answers to the same offer. This
scenario is so common that it is more likely to be implemented then paralle=
l
forking. This is why PRANSWER was suggested.

If cloning can be implemented in WebRTC it would be more standard compliant
then PRANSWER and would allow for a lot more use cases. Typical difficulty
in parallel forking implementation in SIP is due to RTP from multiple
sources coming to the same IP and port with no consent or notification to
the receiving side. This makes it very difficult to present this media to
the user. There is no way to tie media to actual dialogs since RTP can come
from different IP and ports then specified in answer SDPs. This is not an
issue with WebRTC where consent to receive media is required. I would argue
let's implement cloning and not waste any more time on PRANSWER which in my
opinion will always stay a half working hack.
_____________
Roman Shpount

_______________________________________________
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: Martin Thomson <martin.thomson@gmail.com>
Date: Sat, 6 Oct 2012 20:38:49 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

>>>> Can I use the same local descriptor for every setLocal() call?
>>>
>>> My experience suggests that you can.  However, that's not stipulated
>>> in the API specification, so it's not an ironclad guarantee.
>>
>> If it doesn't, and a new local descripor is created, you would need to s=
end that one to the remote=20
>> endpoint.
>
> You don't create a new description, you just set one.

Correct.

> The only problem arises when you can't set the local description because
> something broke since you last set it.  (Maybe setting the answer
> caused the browser to release something that you need to set the
> offer.)

That is the main difference between PRANSWER and ANSWER: PRANSWER does not =
release local resources based on the answer.

> If that happens, you are hosed.  You could try to make a new PeerConnecti=
on, but you'll have to resend your INVITE because you will
> get a new set of candidates, ufrag, pwd, etc...

Or, you could set the new local descriptor BEFORE you apply the answer to t=
he previous one. I think that is how the cloning is currently described in =
JSEP: the cloning happens first (eventhough it may never be used, if no add=
itional remote peers are contacted).

Regards,

Christer=

From salvatore.loreto@ericsson.com  Mon Oct  8 01:22:34 2012
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: [rtcweb] SDP and datachanel
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Hi there,

I have just sent an email to the mmusic mailing list to restart the 
discussion
of what should go in SDP for the SCTP (over DTLS) association setup:
http://www.ietf.org/mail-archive/web/mmusic/current/msg09667.html

In particular what is general enough to be inserted in the current wg item
(http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp)
and what is specific of the Datachannel protocol but needs to be 
eventually specified
in a separate draft.

Please go to the mmusic mailing list and discuss the (SDP related 
)issues there
without cross-posting

thanks
Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 8 Oct 2012 10:56:53 +0200
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Subject: [rtcweb] FW: Draft new: draft-holmberg-mmusic-sdp-mmt-negotiation-00
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FYI,

For those of you interested in media multiplexing.

Regards,

Christer

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: 8. lokakuuta 2012 11:55
To: mmusic@ietf.org
Subject: [MMUSIC] Draft new: draft-holmberg-mmusic-sdp-mmt-negotiation-00

Hi,

We've submitted a new draft, which describes the 'm=3Dbundle' mechanism.

In order to avoid confusion, we're not using bundle terminology in the draf=
t.

Regards,

Christer

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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'c=
olor:#1F497D'>FYI,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D=
'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'color:#1F497D'>For those of you interested in media multiplexing.<o:p>=
</o:p></span></p><p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&n=
bsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Reg=
ards,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'color:#1=
F497D'>Christer<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'co=
lor:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div style=3D'border:none;bor=
der-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal=
><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From=
:</span></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-seri=
f"'> mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf =
Of </b>Christer Holmberg<br><b>Sent:</b> 8. lokakuuta 2012 11:55<br><b>To:<=
/b> mmusic@ietf.org<br><b>Subject:</b> [MMUSIC] Draft new: draft-holmberg-m=
music-sdp-mmt-negotiation-00<o:p></o:p></span></p></div></div><p class=3DMs=
oNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span lang=3DFI>Hi,<o:p><=
/o:p></span></p><p class=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</o:p></spa=
n></p><p class=3DMsoNormal>We&#8217;ve submitted a new draft, which describ=
es the &#8217;m=3Dbundle&#8217; mechanism.<o:p></o:p></p><p class=3DMsoNorm=
al><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>In order to avoid confusion, w=
e&#8217;re not using bundle terminology in the draft.<o:p></o:p></p><p clas=
s=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p></o:p>=
</p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Christer=
<o:p></o:p></p></div></body></html>=

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From christer.holmberg@ericsson.com  Mon Oct  8 02:47:25 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 8 Oct 2012 11:47:20 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Hi,

I would like to discuss the different alternatives in order to support fork=
ing, e.g. whether we use cloning, whether we simply set additional local de=
scriptor, and whether we can get rid of PRANSWER.

As it is related to JSEP and SDP, I guess it should be part of the JSEP dis=
cussion. I am willing to help with material.

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings (fluffy)
Sent: 4. lokakuuta 2012 18:44
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting


With my individual contributor hat on .

Justin and I would like to have some time for JSEP. I think there are a bun=
ch of issues, particularly around the use of SDP and timing of various thin=
gs that need discussion.=20

Thanks, Cullen



On Oct 2, 2012, at 9:08 AM, Magnus Westerlund <magnus.westerlund@ericsson.c=
om> wrote:

> WG,
>=20
> We chairs like to have input into what you see should be on the Agenda=20
> for our two meeting slots in Atlanta. Both document editors/authors=20
> and participant are welcome to request or inform the WG or the chairs=20
> only about issues that needs to be discussed in the meeting.
>=20
> Two items we chairs believe should be on the agenda are:
>=20
> - Video codec selection
> - JSEP
>=20
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On Oct 6, 2012, at 11:24 , Christer Holmberg <christer.holmberg@ericsson.co=
m> wrote:

>=20
> At least I have never thought of any other use of it than to support seri=
al forking.

Other use cases were raised when connecting to a conference bridge where th=
e initial contact point wanted to keep the options for video codecs open un=
til after the actually MCU was connected. I think there were other uses cas=
es when gatewaying to H.323.=20



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Date: Mon, 8 Oct 2012 22:51:31 +0000
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Uh, sort of  - if you send an offer that says you could get early media, yo=
u may get early media from two places even thought you have not received an=
y answer. Yes, I realize you can't do ICE till you get the answer but I thi=
nk the above is still the case in non ICE case an with ICE case you may end=
 up with two dialogs formed but dialogs are really a Sip level concept not =
an RTP concept.=20



On Oct 5, 2012, at 14:26 , Christer Holmberg <christer.holmberg@ericsson.co=
m> wrote:

> Hi,
>=20
> Roman is correct.
>=20
> The SDP O/A doesn't consider forking, because each forked SIP leg creates=
 a unique early dialog, and each early dialog has its own O/A state machine=
.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> ________________________________
> From: Roman Shpount [roman@telurix.com]
> Sent: Friday, October 05, 2012 8:46 PM
> To: Harald Alvestrand
> Cc: Christer Holmberg; rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
>=20
>=20
> On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand <harald@alvestrand.no<m=
ailto:harald@alvestrand.no>> wrote:
> I've asked that question before, but I don't remember seeing an answer fr=
om people who know what they're talking about:
>=20
> Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, =
or does it not violate the SDP O/A model?
>=20
> I don't understand how it, in the form that has been described as "what w=
e have to support", can be SDP O/A compatible, but then there are many thin=
gs about SIP that I don't understand, so I may understand it when it's expl=
ained to me.
>=20
>=20
> O/A does not describe multiple answers to a single offer. Nothing explici=
tly prohibits it there but I would argue this is not part of this specifica=
tion. No standard document that I am aware of describes how serial O/A is s=
upposed to work. Serial forking as practiced by SIP UA does violate SIP RFC=
 3261 which states that each dialog can only have one answer to each offer.=
 If answer is sent in both provisional and final response, it should be the=
 same. You can, however, create multiple dialogs with the same offer. This =
normally implies parallel forking, but the most common use case is with ear=
ly media, where you end up with multiple early dialogs. For instance you ca=
ll a phone number, phone network sends you SDP in early media and plays a d=
ial tone, then the called number does not answer, and you get connected to =
a voice mail which uses a completely different SDP in final answer. Accordi=
ng to SIP these answers should come with different to-tags and technically =
would be parts of
>  two different dialogs. What makes this a bit tricky is when the phone ne=
twork and the voice mail are behind SBC (or some other sort of B2BUA) you o=
nly see one dialog which send you multiple answers to the same offer. This =
scenario is so common that it is more likely to be implemented then paralle=
l forking. This is why PRANSWER was suggested.
>=20
> If cloning can be implemented in WebRTC it would be more standard complia=
nt then PRANSWER and would allow for a lot more use cases. Typical difficul=
ty in parallel forking implementation in SIP is due to RTP from multiple so=
urces coming to the same IP and port with no consent or notification to the=
 receiving side. This makes it very difficult to present this media to the =
user. There is no way to tie media to actual dialogs since RTP can come fro=
m different IP and ports then specified in answer SDPs. This is not an issu=
e with WebRTC where consent to receive media is required. I would argue let=
's implement cloning and not waste any more time on PRANSWER which in my op=
inion will always stay a half working hack.
> _____________
> Roman Shpount
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
Thread-Index: AQHNpad+aenXLuIV20+kkVz1nsB6Xg==
Date: Mon, 8 Oct 2012 22:51:46 +0000
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On Oct 6, 2012, at 11:38 , Christer Holmberg <christer.holmberg@ericsson.co=
m> wrote:

> Hi,
>=20
>>>>> Can I use the same local descriptor for every setLocal() call?
>>>>=20
>>>> My experience suggests that you can.  However, that's not stipulated
>>>> in the API specification, so it's not an ironclad guarantee.
>>>=20
>>> If it doesn't, and a new local descripor is created, you would need to =
send that one to the remote=20
>>> endpoint.
>>=20
>> You don't create a new description, you just set one.
>=20
> Correct.
>=20
>> The only problem arises when you can't set the local description because
>> something broke since you last set it.  (Maybe setting the answer
>> caused the browser to release something that you need to set the

If you send an answer that has not relation to the offer, you are equally h=
osed. We are keeping the pairing of the offers and answers in the JS state =
so there are ways that the JS can mess things up if it is buggy but that no=
t really a PRANSWER problem - it's just as much a issue with offers and ans=
wers.=20

>> offer.)
>=20
> That is the main difference between PRANSWER and ANSWER: PRANSWER does no=
t release local resources based on the answer.

Exactly - and the important implication of this is that the far end can kee=
p using the resources that were in the  Offer. Sequential forking is a post=
er child for one of the cases where you need to do this but there are other=
s.  =20

>=20
>> If that happens, you are hosed.  You could try to make a new PeerConnect=
ion, but you'll have to resend your INVITE because you will
>> get a new set of candidates, ufrag, pwd, etc...
>=20
> Or, you could set the new local descriptor BEFORE you apply the answer to=
 the previous one. I think that is how the cloning is currently described i=
n JSEP: the cloning happens first (eventhough it may never be used, if no a=
dditional remote peers are contacted).
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP-02: SDP line
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Subject: Re: [rtcweb] JSEP-02: SDP line
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On Oct 4, 2012, at 7:33 , Christer Holmberg <christer.holmberg@ericsson.com=
> wrote:

> Hi,
> =20
> What is a =94SDP line=94? The same thing as a =93SDP blob=94?
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

this is poorly defined and we need to fix that - it's probably the wrong te=
rm=20

I think it was trying to get at the m-line matching issues in SDP so the te=
rm=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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On Oct 8, 2012, at 2:47 , Christer Holmberg <christer.holmberg@ericsson.com=
> wrote:

> Hi,
>=20
> I would like to discuss the different alternatives in order to support fo=
rking, e.g. whether we use cloning, whether we simply set additional local =
descriptor, and whether we can get rid of PRANSWER.

Seriously? we have discussed this so many times and always come to the same=
 conclusion. I have not seen anything on the list that suggests why we need=
 to remove this or how mapping to SIP 180 with sequential forking is going =
to work without it. It also has other important uses. There are a bunch of =
changes that are needed to the JSEP draft to remove some of the inconsisten=
cies in this and clarify some parts but I'd rather wait till we had that up=
dated before we got into a whole discussion about exploding it yet again.=20

Why don't we have a phone call to try and outline what the problems you are=
 trying to solve that the current solution does not work for then figure ou=
t how much we want to explode this.=20

I'll note that current clone text has lots of "miracles happen here, insert=
 supper fluffy hand wave" in it and plenty of weasel room on failure to all=
ocate required resources on the clone. It's more or less a sketch of an ide=
a at this point. I'm perfectly happy to see people try and sort out the det=
ails on clone but using it explode the consensus we have come to around PRA=
NSWER seems like a really bad idea at this point.=20

Cullen


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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On Oct 8, 2012, at 2:47 , Christer Holmberg <christer.holmberg@ericsson.com=
> wrote:

> Hi,
>=20
> I would like to discuss the different alternatives in order to support fo=
rking, e.g. whether we use cloning, whether we simply set additional local =
descriptor, and whether we can get rid of PRANSWER.

Seriously? we have discussed this so many times and always come to the same=
 conclusion. I have not seen anything on the list that suggests why we need=
 to remove this or how mapping to SIP 180 with sequential forking is going =
to work without it. It also has other important uses. There are a bunch of =
changes that are needed to the JSEP draft to remove some of the inconsisten=
cies in this and clarify some parts but I'd rather wait till we had that up=
dated before we got into a whole discussion about exploding it yet again.=20

Why don't we have a phone call to try and outline what the problems you are=
 trying to solve that the current solution does not work for then figure ou=
t how much we want to explode this.=20

I'll note that current clone text has lots of "miracles happen here, insert=
 supper fluffy hand wave" in it and plenty of weasel room on failure to all=
ocate required resources on the clone. It's more or less a sketch of an ide=
a at this point. I'm perfectly happy to see people try and sort out the det=
ails on clone but using it explode the consensus we have come to around PRA=
NSWER seems like a really bad idea at this point.=20

Cullen


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Cullen,

I have an existing interop problem with SIP serial forking that I cannot
solve with the proposed schema. With PRANSWER, how would I handle SIP
UPDATE sith SDP received in early dialog?

Regards,
_____________
Roman Shpount


On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Oct 8, 2012, at 2:47 , Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> > Hi,
> >
> > I would like to discuss the different alternatives in order to support
> forking, e.g. whether we use cloning, whether we simply set additional
> local descriptor, and whether we can get rid of PRANSWER.
>
> Seriously? we have discussed this so many times and always come to the
> same conclusion. I have not seen anything on the list that suggests why we
> need to remove this or how mapping to SIP 180 with sequential forking is
> going to work without it. It also has other important uses. There are a
> bunch of changes that are needed to the JSEP draft to remove some of the
> inconsistencies in this and clarify some parts but I'd rather wait till we
> had that updated before we got into a whole discussion about exploding it
> yet again.
>
> Why don't we have a phone call to try and outline what the problems you
> are trying to solve that the current solution does not work for then figure
> out how much we want to explode this.
>
> I'll note that current clone text has lots of "miracles happen here,
> insert supper fluffy hand wave" in it and plenty of weasel room on failure
> to allocate required resources on the clone. It's more or less a sketch of
> an idea at this point. I'm perfectly happy to see people try and sort out
> the details on clone but using it explode the consensus we have come to
> around PRANSWER seems like a really bad idea at this point.
>
> Cullen
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b2e79887dcc9004cb942afd
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Cullen,<div><br></div><div>I have an existing interop problem with SIP seri=
al forking that I cannot solve with the proposed schema. With PRANSWER, how=
 would I handle SIP UPDATE sith SDP received in early dialog?</div><div>
<br></div><div>Regards,<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at 6:53 PM, Cullen J=
ennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" =
target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Oct 8, 2012, at 2:47 , Christer Holmberg &lt;<a href=3D"mailto:christer.=
holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; I would like to discuss the different alternatives in order to support=
 forking, e.g. whether we use cloning, whether we simply set additional loc=
al descriptor, and whether we can get rid of PRANSWER.<br>
<br>
Seriously? we have discussed this so many times and always come to the same=
 conclusion. I have not seen anything on the list that suggests why we need=
 to remove this or how mapping to SIP 180 with sequential forking is going =
to work without it. It also has other important uses. There are a bunch of =
changes that are needed to the JSEP draft to remove some of the inconsisten=
cies in this and clarify some parts but I&#39;d rather wait till we had tha=
t updated before we got into a whole discussion about exploding it yet agai=
n.<br>

<br>
Why don&#39;t we have a phone call to try and outline what the problems you=
 are trying to solve that the current solution does not work for then figur=
e out how much we want to explode this.<br>
<br>
I&#39;ll note that current clone text has lots of &quot;miracles happen her=
e, insert supper fluffy hand wave&quot; in it and plenty of weasel room on =
failure to allocate required resources on the clone. It&#39;s more or less =
a sketch of an idea at this point. I&#39;m perfectly happy to see people tr=
y and sort out the details on clone but using it explode the consensus we h=
ave come to around PRANSWER seems like a really bad idea at this point.<br>

<br>
Cullen<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>

--047d7b2e79887dcc9004cb942afd--

From ekr@rtfm.com  Mon Oct  8 16:01:40 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 8 Oct 2012 16:00:58 -0700
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I would attend such a phone call if it were held.

My immediate reaction is that from an implementor's perspective. PRANSWER
is a lot easier to address than cloning. I feel like cloning pulls in a lot=
 of
difficult lifetime and ownership semantics. By contrast, supporting PRANSWE=
R
is very easy. Before deciding that cloning is the answer I'd like to see tw=
o
things:

1. A clear set of use cases that shows what cloning does well that PRANSWER
does not.
2. A fairly complete defn of the clone semantics.

Otherwise I worry that we'll detour to cloning and then only after 6 months
discover it doesn't work.

Best,
-Ekr



On Mon, Oct 8, 2012 at 3:52 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com> wrote:
>
> On Oct 8, 2012, at 2:47 , Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:
>
>> Hi,
>>
>> I would like to discuss the different alternatives in order to support f=
orking, e.g. whether we use cloning, whether we simply set additional local=
 descriptor, and whether we can get rid of PRANSWER.
>
> Seriously? we have discussed this so many times and always come to the sa=
me conclusion. I have not seen anything on the list that suggests why we ne=
ed to remove this or how mapping to SIP 180 with sequential forking is goin=
g to work without it. It also has other important uses. There are a bunch o=
f changes that are needed to the JSEP draft to remove some of the inconsist=
encies in this and clarify some parts but I'd rather wait till we had that =
updated before we got into a whole discussion about exploding it yet again.
>
> Why don't we have a phone call to try and outline what the problems you a=
re trying to solve that the current solution does not work for then figure =
out how much we want to explode this.
>
> I'll note that current clone text has lots of "miracles happen here, inse=
rt supper fluffy hand wave" in it and plenty of weasel room on failure to a=
llocate required resources on the clone. It's more or less a sketch of an i=
dea at this point. I'm perfectly happy to see people try and sort out the d=
etails on clone but using it explode the consensus we have come to around P=
RANSWER seems like a really bad idea at this point.
>
> Cullen
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From roman@telurix.com  Mon Oct  8 16:04:11 2012
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On Mon, Oct 8, 2012 at 6:51 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> Uh, sort of  - if you send an offer that says you could get early media,
> you may get early media from two places even thought you have not received
> any answer. Yes, I realize you can't do ICE till you get the answer but I
> think the above is still the case in non ICE case an with ICE case you may
> end up with two dialogs formed but dialogs are really a Sip level concept
> not an RTP concept.
>
>
Dialogs are entirely SIP level concept, but what I am arguing is that with
ICE and WebRTC you can associate RTP streams with a particular answer. This
means you can clone connection, provide it with the answer and each
connection would be able to filter out only the media that is intended for
it. If I am missing something and ICE is not enough then msid should be.
_____________
Roman Shpount

--047d7b2e7988b0201604cb943cb5
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<br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at 6:51 PM, Cullen Jenni=
ngs (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" targ=
et=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<br>
Uh, sort of =A0- if you send an offer that says you could get early media, =
you may get early media from two places even thought you have not received =
any answer. Yes, I realize you can&#39;t do ICE till you get the answer but=
 I think the above is still the case in non ICE case an with ICE case you m=
ay end up with two dialogs formed but dialogs are really a Sip level concep=
t not an RTP concept.<br>

<div class=3D"HOEnZb"><div class=3D"h5"><br></div></div></blockquote><div><=
br></div><div>Dialogs=A0are=A0entirely SIP level concept, but what I am arg=
uing is that with ICE and WebRTC you can associate RTP streams with a parti=
cular answer. This means you can clone connection, provide it with the answ=
er and each connection would be able to filter out only the media that is i=
ntended for it.=A0If I am missing something and ICE is not enough then msid=
 should be.</div>
_____________<br>Roman Shpount<br><div>=A0</div></div>

--047d7b2e7988b0201604cb943cb5--

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Walk me through what the call flow would like it both endpoints were SIP =
UA's so I understand it better.=20


Note - I'm not against folks trying to figure out clone. I'm saying that =
we should not get rid of PRANSWER.


On Oct 8, 2012, at 15:58 , Roman Shpount <roman@telurix.com> wrote:

> Cullen,
>=20
> I have an existing interop problem with SIP serial forking that I =
cannot solve with the proposed schema. With PRANSWER, how would I handle =
SIP UPDATE sith SDP received in early dialog?
>=20
> Regards,
> _____________
> Roman Shpount
>=20
>=20
> On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
>=20
> On Oct 8, 2012, at 2:47 , Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> > Hi,
> >
> > I would like to discuss the different alternatives in order to =
support forking, e.g. whether we use cloning, whether we simply set =
additional local descriptor, and whether we can get rid of PRANSWER.
>=20
> Seriously? we have discussed this so many times and always come to the =
same conclusion. I have not seen anything on the list that suggests why =
we need to remove this or how mapping to SIP 180 with sequential forking =
is going to work without it. It also has other important uses. There are =
a bunch of changes that are needed to the JSEP draft to remove some of =
the inconsistencies in this and clarify some parts but I'd rather wait =
till we had that updated before we got into a whole discussion about =
exploding it yet again.
>=20
> Why don't we have a phone call to try and outline what the problems =
you are trying to solve that the current solution does not work for then =
figure out how much we want to explode this.
>=20
> I'll note that current clone text has lots of "miracles happen here, =
insert supper fluffy hand wave" in it and plenty of weasel room on =
failure to allocate required resources on the clone. It's more or less a =
sketch of an idea at this point. I'm perfectly happy to see people try =
and sort out the details on clone but using it explode the consensus we =
have come to around PRANSWER seems like a really bad idea at this point.
>=20
> Cullen
>=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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--f46d042de40ba70a3404cb947ca7
Content-Type: text/plain; charset=ISO-8859-1

The scenario is:

WebRTC is placing a call through some sort of gateway to a SIP end point.
It sends an offer that gets translated into INVITE with SDP. SIP end point
sends back 183 with SDP that gets delivered to the WebRTC end point as
PRANSWER. Then SIP end point sends an UPDATE with SDP in early dialog. And
then finally SIP end point sends 200 OK with different SDP. I am not sure
how to translate the SDP in UPDATE and the final answer in WebRTC API
calls. I would guess this is not something that can be supported without
cloning.
_____________
Roman Shpount


On Mon, Oct 8, 2012 at 7:09 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Walk me through what the call flow would like it both endpoints were SIP
> UA's so I understand it better.
>
>
> Note - I'm not against folks trying to figure out clone. I'm saying that
> we should not get rid of PRANSWER.
>
>
> On Oct 8, 2012, at 15:58 , Roman Shpount <roman@telurix.com> wrote:
>
> > Cullen,
> >
> > I have an existing interop problem with SIP serial forking that I cannot
> solve with the proposed schema. With PRANSWER, how would I handle SIP
> UPDATE sith SDP received in early dialog?
> >
> > Regards,
> > _____________
> > Roman Shpount
> >
> >
> > On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy) <
> fluffy@cisco.com> wrote:
> >
> > On Oct 8, 2012, at 2:47 , Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >
> > > Hi,
> > >
> > > I would like to discuss the different alternatives in order to support
> forking, e.g. whether we use cloning, whether we simply set additional
> local descriptor, and whether we can get rid of PRANSWER.
> >
> > Seriously? we have discussed this so many times and always come to the
> same conclusion. I have not seen anything on the list that suggests why we
> need to remove this or how mapping to SIP 180 with sequential forking is
> going to work without it. It also has other important uses. There are a
> bunch of changes that are needed to the JSEP draft to remove some of the
> inconsistencies in this and clarify some parts but I'd rather wait till we
> had that updated before we got into a whole discussion about exploding it
> yet again.
> >
> > Why don't we have a phone call to try and outline what the problems you
> are trying to solve that the current solution does not work for then figure
> out how much we want to explode this.
> >
> > I'll note that current clone text has lots of "miracles happen here,
> insert supper fluffy hand wave" in it and plenty of weasel room on failure
> to allocate required resources on the clone. It's more or less a sketch of
> an idea at this point. I'm perfectly happy to see people try and sort out
> the details on clone but using it explode the consensus we have come to
> around PRANSWER seems like a really bad idea at this point.
> >
> > Cullen
> >
> > _______________________________________________
> > 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
>
>

--f46d042de40ba70a3404cb947ca7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

The scenario is:<div><br></div><div>WebRTC is placing a call through some s=
ort of gateway to a SIP end point. It sends an offer that gets translated i=
nto INVITE with SDP. SIP end point sends back 183 with SDP that gets delive=
red to the WebRTC end point as PRANSWER. Then SIP end point sends an UPDATE=
 with SDP in early dialog. And then finally SIP end point sends 200 OK with=
 different SDP. I am not sure how to translate the SDP in UPDATE and the fi=
nal answer in WebRTC API calls. I would guess this is not something that ca=
n be supported without cloning.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at 7:09 PM, Cullen J=
ennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_b=
lank">fluffy@iii.ca</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"=
>
<br>
Walk me through what the call flow would like it both endpoints were SIP UA=
&#39;s so I understand it better.<br>
<br>
<br>
Note - I&#39;m not against folks trying to figure out clone. I&#39;m saying=
 that we should not get rid of PRANSWER.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On Oct 8, 2012, at 15:58 , Roman Shpount &lt;<a href=3D"mailto:roman@teluri=
x.com">roman@telurix.com</a>&gt; wrote:<br>
<br>
&gt; Cullen,<br>
&gt;<br>
&gt; I have an existing interop problem with SIP serial forking that I cann=
ot solve with the proposed schema. With PRANSWER, how would I handle SIP UP=
DATE sith SDP received in early dialog?<br>
&gt;<br>
&gt; Regards,<br>
&gt; _____________<br>
&gt; Roman Shpount<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy) &lt;<a href=
=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On Oct 8, 2012, at 2:47 , Christer Holmberg &lt;<a href=3D"mailto:chri=
ster.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:<b=
r>
&gt;<br>
&gt; &gt; Hi,<br>
&gt; &gt;<br>
&gt; &gt; I would like to discuss the different alternatives in order to su=
pport forking, e.g. whether we use cloning, whether we simply set additiona=
l local descriptor, and whether we can get rid of PRANSWER.<br>
&gt;<br>
&gt; Seriously? we have discussed this so many times and always come to the=
 same conclusion. I have not seen anything on the list that suggests why we=
 need to remove this or how mapping to SIP 180 with sequential forking is g=
oing to work without it. It also has other important uses. There are a bunc=
h of changes that are needed to the JSEP draft to remove some of the incons=
istencies in this and clarify some parts but I&#39;d rather wait till we ha=
d that updated before we got into a whole discussion about exploding it yet=
 again.<br>

&gt;<br>
&gt; Why don&#39;t we have a phone call to try and outline what the problem=
s you are trying to solve that the current solution does not work for then =
figure out how much we want to explode this.<br>
&gt;<br>
&gt; I&#39;ll note that current clone text has lots of &quot;miracles happe=
n here, insert supper fluffy hand wave&quot; in it and plenty of weasel roo=
m on failure to allocate required resources on the clone. It&#39;s more or =
less a sketch of an idea at this point. I&#39;m perfectly happy to see peop=
le try and sort out the details on clone but using it explode the consensus=
 we have come to around PRANSWER seems like a really bad idea at this point=
.<br>

&gt;<br>
&gt; Cullen<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>
</div></div></blockquote></div><br></div>

--f46d042de40ba70a3404cb947ca7--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Date: Tue, 9 Oct 2012 00:42:48 +0000
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Check out text from RFC 3311 that says=20

      o  If the UPDATE is being sent before the completion of the INVITE
         transaction, and the initial INVITE contained an offer, the
         UPDATE cannot be sent with an offer unless the callee has
         generated an answer in a reliable provisional response, has
         received a PRACK for that reliable provisional response, has
         not received any requests (PRACK or UPDATE) with offers that it
         has not answered, and has not sent any UPDATE requests
         containing offers that have not been answered.

So the 183 would be need to be sent reliably at which case I would expect t=
he gateways to map it to a final offer in RTCWeb not a provisional one. At =
that point when the update comes along with an offer, it just mapped to RTC=
web offer.=20


On Oct 8, 2012, at 16:21 , Roman Shpount <roman@telurix.com>
 wrote:

> The scenario is:
>=20
> WebRTC is placing a call through some sort of gateway to a SIP end point.=
 It sends an offer that gets translated into INVITE with SDP. SIP end point=
 sends back 183 with SDP that gets delivered to the WebRTC end point as PRA=
NSWER. Then SIP end point sends an UPDATE with SDP in early dialog. And the=
n finally SIP end point sends 200 OK with different SDP. I am not sure how =
to translate the SDP in UPDATE and the final answer in WebRTC API calls. I =
would guess this is not something that can be supported without cloning.
> _____________
> Roman Shpount
>=20
>=20
> On Mon, Oct 8, 2012 at 7:09 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
> Walk me through what the call flow would like it both endpoints were SIP =
UA's so I understand it better.
>=20
>=20
> Note - I'm not against folks trying to figure out clone. I'm saying that =
we should not get rid of PRANSWER.
>=20
>=20
> On Oct 8, 2012, at 15:58 , Roman Shpount <roman@telurix.com> wrote:
>=20
> > Cullen,
> >
> > I have an existing interop problem with SIP serial forking that I canno=
t solve with the proposed schema. With PRANSWER, how would I handle SIP UPD=
ATE sith SDP received in early dialog?
> >
> > Regards,
> > _____________
> > Roman Shpount
> >
> >
> > On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy) <fluffy@cisco.=
com> wrote:
> >
> > On Oct 8, 2012, at 2:47 , Christer Holmberg <christer.holmberg@ericsson=
.com> wrote:
> >
> > > Hi,
> > >
> > > I would like to discuss the different alternatives in order to suppor=
t forking, e.g. whether we use cloning, whether we simply set additional lo=
cal descriptor, and whether we can get rid of PRANSWER.
> >
> > Seriously? we have discussed this so many times and always come to the =
same conclusion. I have not seen anything on the list that suggests why we =
need to remove this or how mapping to SIP 180 with sequential forking is go=
ing to work without it. It also has other important uses. There are a bunch=
 of changes that are needed to the JSEP draft to remove some of the inconsi=
stencies in this and clarify some parts but I'd rather wait till we had tha=
t updated before we got into a whole discussion about exploding it yet agai=
n.
> >
> > Why don't we have a phone call to try and outline what the problems you=
 are trying to solve that the current solution does not work for then figur=
e out how much we want to explode this.
> >
> > I'll note that current clone text has lots of "miracles happen here, in=
sert supper fluffy hand wave" in it and plenty of weasel room on failure to=
 allocate required resources on the clone. It's more or less a sketch of an=
 idea at this point. I'm perfectly happy to see people try and sort out the=
 details on clone but using it explode the consensus we have come to around=
 PRANSWER seems like a really bad idea at this point.
> >
> > Cullen
> >
> > _______________________________________________
> > 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


From fluffy@iii.ca  Mon Oct  8 17:47:09 2012
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From: Cullen Jennings <fluffy@iii.ca>
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One other thought on this. I think this just points out that it would be =
really nice to have a draft that showed how to build a RTCWeb / SIP =
gateway.=20

=20
On Oct 8, 2012, at 17:42 , Cullen Jennings <fluffy@cisco.com> wrote:

>=20
> Check out text from RFC 3311 that says=20
>=20
>      o  If the UPDATE is being sent before the completion of the =
INVITE
>         transaction, and the initial INVITE contained an offer, the
>         UPDATE cannot be sent with an offer unless the callee has
>         generated an answer in a reliable provisional response, has
>         received a PRACK for that reliable provisional response, has
>         not received any requests (PRACK or UPDATE) with offers that =
it
>         has not answered, and has not sent any UPDATE requests
>         containing offers that have not been answered.
>=20
> So the 183 would be need to be sent reliably at which case I would =
expect the gateways to map it to a final offer in RTCWeb not a =
provisional one. At that point when the update comes along with an =
offer, it just mapped to RTCweb offer.=20
>=20
>=20
> On Oct 8, 2012, at 16:21 , Roman Shpount <roman@telurix.com>
> wrote:
>=20
>> The scenario is:
>>=20
>> WebRTC is placing a call through some sort of gateway to a SIP end =
point. It sends an offer that gets translated into INVITE with SDP. SIP =
end point sends back 183 with SDP that gets delivered to the WebRTC end =
point as PRANSWER. Then SIP end point sends an UPDATE with SDP in early =
dialog. And then finally SIP end point sends 200 OK with different SDP. =
I am not sure how to translate the SDP in UPDATE and the final answer in =
WebRTC API calls. I would guess this is not something that can be =
supported without cloning.
>> _____________
>> Roman Shpount
>>=20
>>=20
>> On Mon, Oct 8, 2012 at 7:09 PM, Cullen Jennings <fluffy@iii.ca> =
wrote:
>>=20
>> Walk me through what the call flow would like it both endpoints were =
SIP UA's so I understand it better.
>>=20
>>=20
>> Note - I'm not against folks trying to figure out clone. I'm saying =
that we should not get rid of PRANSWER.
>>=20
>>=20
>> On Oct 8, 2012, at 15:58 , Roman Shpount <roman@telurix.com> wrote:
>>=20
>>> Cullen,
>>>=20
>>> I have an existing interop problem with SIP serial forking that I =
cannot solve with the proposed schema. With PRANSWER, how would I handle =
SIP UPDATE sith SDP received in early dialog?
>>>=20
>>> Regards,
>>> _____________
>>> Roman Shpount
>>>=20
>>>=20
>>> On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
>>>=20
>>> On Oct 8, 2012, at 2:47 , Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>>=20
>>>> Hi,
>>>>=20
>>>> I would like to discuss the different alternatives in order to =
support forking, e.g. whether we use cloning, whether we simply set =
additional local descriptor, and whether we can get rid of PRANSWER.
>>>=20
>>> Seriously? we have discussed this so many times and always come to =
the same conclusion. I have not seen anything on the list that suggests =
why we need to remove this or how mapping to SIP 180 with sequential =
forking is going to work without it. It also has other important uses. =
There are a bunch of changes that are needed to the JSEP draft to remove =
some of the inconsistencies in this and clarify some parts but I'd =
rather wait till we had that updated before we got into a whole =
discussion about exploding it yet again.
>>>=20
>>> Why don't we have a phone call to try and outline what the problems =
you are trying to solve that the current solution does not work for then =
figure out how much we want to explode this.
>>>=20
>>> I'll note that current clone text has lots of "miracles happen here, =
insert supper fluffy hand wave" in it and plenty of weasel room on =
failure to allocate required resources on the clone. It's more or less a =
sketch of an idea at this point. I'm perfectly happy to see people try =
and sort out the details on clone but using it explode the consensus we =
have come to around PRANSWER seems like a really bad idea at this point.
>>>=20
>>> Cullen
>>>=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
>>=20
>=20


From roman@telurix.com  Mon Oct  8 18:24:12 2012
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--e89a8ff1c03eb2966904cb963184
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On Mon, Oct 8, 2012 at 8:42 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> Check out text from RFC 3311 that says
>
>       o  If the UPDATE is being sent before the completion of the INVITE
>          transaction, and the initial INVITE contained an offer, the
>          UPDATE cannot be sent with an offer unless the callee has
>          generated an answer in a reliable provisional response, has
>          received a PRACK for that reliable provisional response, has
>          not received any requests (PRACK or UPDATE) with offers that it
>          has not answered, and has not sent any UPDATE requests
>          containing offers that have not been answered.
>
> So the 183 would be need to be sent reliably at which case I would expect
> the gateways to map it to a final offer in RTCWeb not a provisional one. At
> that point when the update comes along with an offer, it just mapped to
> RTCweb offer.
>
>
And what happens with the next 183 or 200 OK sent reliably in a different
dialog? When we are dealing with serial forking we are dealing with
different sequential dialogs created by the same offer, aren't we?
Otherwise this is not really standard compliant.
_____________
Roman Shpount

--e89a8ff1c03eb2966904cb963184
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at 8:4=
2 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluf=
fy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
<br>
Check out text from RFC 3311 that says<br>
<br>
=A0 =A0 =A0 o =A0If the UPDATE is being sent before the completion of the I=
NVITE<br>
=A0 =A0 =A0 =A0 =A0transaction, and the initial INVITE contained an offer, =
the<br>
=A0 =A0 =A0 =A0 =A0UPDATE cannot be sent with an offer unless the callee ha=
s<br>
=A0 =A0 =A0 =A0 =A0generated an answer in a reliable provisional response, =
has<br>
=A0 =A0 =A0 =A0 =A0received a PRACK for that reliable provisional response,=
 has<br>
=A0 =A0 =A0 =A0 =A0not received any requests (PRACK or UPDATE) with offers =
that it<br>
=A0 =A0 =A0 =A0 =A0has not answered, and has not sent any UPDATE requests<b=
r>
=A0 =A0 =A0 =A0 =A0containing offers that have not been answered.<br>
<br>
So the 183 would be need to be sent reliably at which case I would expect t=
he gateways to map it to a final offer in RTCWeb not a provisional one. At =
that point when the update comes along with an offer, it just mapped to RTC=
web offer.<br>

<br></blockquote><div><br></div><div>And what happens with the next 183 or =
200 OK sent reliably in a different dialog? When we are dealing with serial=
 forking we are dealing with different sequential dialogs created by the sa=
me offer, aren&#39;t we? Otherwise this is not really standard compliant.=
=A0</div>
_____________<br>Roman Shpount<br><br><div>=A0</div></div>

--e89a8ff1c03eb2966904cb963184--

From fluffy@cisco.com  Mon Oct  8 18:26:52 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
Thread-Index: AQHNpaetkG2ADNG+3EO5+NGGgdEEepewWX6AgAAC9ICAAAN2gIAAFs4AgAALWACAAAD2AA==
Date: Tue, 9 Oct 2012 01:26:50 +0000
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Well, what's the end user experience you're trying to achieve  - lets say w=
e are talking about audio. Do you want to play the first one or the last on=
e? If you want to bridge them into a party line, this is probably not how y=
ou want the SIP side to happen.=20



On Oct 8, 2012, at 18:24 , Roman Shpount <roman@telurix.com>
 wrote:

>=20
>=20
> On Mon, Oct 8, 2012 at 8:42 PM, Cullen Jennings (fluffy) <fluffy@cisco.co=
m> wrote:
>=20
> Check out text from RFC 3311 that says
>=20
>       o  If the UPDATE is being sent before the completion of the INVITE
>          transaction, and the initial INVITE contained an offer, the
>          UPDATE cannot be sent with an offer unless the callee has
>          generated an answer in a reliable provisional response, has
>          received a PRACK for that reliable provisional response, has
>          not received any requests (PRACK or UPDATE) with offers that it
>          has not answered, and has not sent any UPDATE requests
>          containing offers that have not been answered.
>=20
> So the 183 would be need to be sent reliably at which case I would expect=
 the gateways to map it to a final offer in RTCWeb not a provisional one. A=
t that point when the update comes along with an offer, it just mapped to R=
TCweb offer.
>=20
>=20
> And what happens with the next 183 or 200 OK sent reliably in a different=
 dialog? When we are dealing with serial forking we are dealing with differ=
ent sequential dialogs created by the same offer, aren't we? Otherwise this=
 is not really standard compliant.=20
> _____________
> Roman Shpount
>=20
> =20


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--047d7b1121a581984504cb964d2a
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On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> Well, what's the end user experience you're trying to achieve  - lets say
> we are talking about audio. Do you want to play the first one or the last
> one? If you want to bridge them into a party line, this is probably not how
> you want the SIP side to happen.
>
>
This is still serial forking. Imagine the following scenario: You are
calling a phone number, you dial first provider who starts playing a dial
tone then uses an UPDATE to connect you to a media server to play an
announcement after which your call goes to voice mail (another dialog and
another SDP). This is all fairly legal and not unusual for SIP scenarios.
Nothing to do with party lines.
_____________
Roman Shpount

--047d7b1121a581984504cb964d2a
Content-Type: text/html; charset=ISO-8859-1
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On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">=
&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com<=
/a>&gt;</span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">
<br>
Well, what&#39;s the end user experience you&#39;re trying to achieve =A0- =
lets say we are talking about audio. Do you want to play the first one or t=
he last one? If you want to bridge them into a party line, this is probably=
 not how you want the SIP side to happen.<br>

<br></blockquote><div><br></div>This is still serial forking. Imagine the f=
ollowing scenario: You are calling a phone number, you dial first provider =
who starts playing a dial tone then uses an UPDATE to connect you to a medi=
a server to play an announcement after which your call goes to voice mail (=
another dialog and another SDP). This is all fairly legal and not unusual f=
or SIP scenarios. Nothing to do with party lines.<br clear=3D"all">
_____________<br>Roman Shpount<br><br><div>=A0</div></div>

--047d7b1121a581984504cb964d2a--

From fluffy@iii.ca  Mon Oct  8 18:41:01 2012
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On Oct 8, 2012, at 18:31 , Roman Shpount <roman@telurix.com> wrote:

> On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
>=20
> Well, what's the end user experience you're trying to achieve  - lets =
say we are talking about audio. Do you want to play the first one or the =
last one? If you want to bridge them into a party line, this is probably =
not how you want the SIP side to happen.
>=20
>=20
> This is still serial forking. Imagine the following scenario: You are =
calling a phone number, you dial first provider who starts playing a =
dial tone then uses an UPDATE to connect you to a media server to play =
an announcement after which your call goes to voice mail (another dialog =
and another SDP). This is all fairly legal and not unusual for SIP =
scenarios. Nothing to do with party lines.
> _____________
> Roman Shpount
>=20

Ok, serial forking with early media seems like it should work. Seems =
like the browser would make the offer, that gateway turned into SIP =
invite that then went to sequential forking proxy that send sent the =
invite to UA A. UA A returns a 180 that gets mapped to PRANSWER, the =
proxy then forks to UA B that sends a 200 on a different dialog that =
gets mapped to answer.=20=

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On Mon, Oct 8, 2012 at 9:41 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Oct 8, 2012, at 18:31 , Roman Shpount <roman@telurix.com> wrote:
>
> > On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) <
> fluffy@cisco.com> wrote:
> >
> > Well, what's the end user experience you're trying to achieve  - lets
> say we are talking about audio. Do you want to play the first one or the
> last one? If you want to bridge them into a party line, this is probably
> not how you want the SIP side to happen.
> >
> >
> > This is still serial forking. Imagine the following scenario: You are
> calling a phone number, you dial first provider who starts playing a dial
> tone then uses an UPDATE to connect you to a media server to play an
> announcement after which your call goes to voice mail (another dialog and
> another SDP). This is all fairly legal and not unusual for SIP scenarios.
> Nothing to do with party lines.
>
> Ok, serial forking with early media seems like it should work. Seems like
> the browser would make the offer, that gateway turned into SIP invite that
> then went to sequential forking proxy that send sent the invite to UA A. UA
> A returns a 180 that gets mapped to PRANSWER, the proxy then forks to UA B
> that sends a 200 on a different dialog that gets mapped to answer.


 The only issue is the there is currently no way to map an UPDATE received
in the early dialog to WebRTC API calls. I am probably OK with it not being
handled until the cloning is implemented, but this shows that PRANSWER is
not very useful even for the serial forking.
_____________
Roman Shpount

--14dae9cdc6c16bfa2e04cb96a5b4
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On Mon, Oct 8, 2012 at 9:41 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> =
wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Oct 8, 2012, at 18:31 , Roman Shpount &lt;<a href=3D"mailto:roman@teluri=
x.com">roman@telurix.com</a>&gt; wrote:<br>
<br>
&gt; On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) &lt;<a href=
=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Well, what&#39;s the end user experience you&#39;re trying to achieve =
=A0- lets say we are talking about audio. Do you want to play the first one=
 or the last one? If you want to bridge them into a party line, this is pro=
bably not how you want the SIP side to happen.<br>

&gt;<br>
&gt;<br>
&gt; This is still serial forking. Imagine the following scenario: You are =
calling a phone number, you dial first provider who starts playing a dial t=
one then uses an UPDATE to connect you to a media server to play an announc=
ement after which your call goes to voice mail (another dialog and another =
SDP). This is all fairly legal and not unusual for SIP scenarios. Nothing t=
o do with party lines.<br>

<br>
</div></div>Ok, serial forking with early media seems like it should work. =
Seems like the browser would make the offer, that gateway turned into SIP i=
nvite that then went to sequential forking proxy that send sent the invite =
to UA A. UA A returns a 180 that gets mapped to PRANSWER, the proxy then fo=
rks to UA B that sends a 200 on a different dialog that gets mapped to answ=
er. </blockquote>
</div><br>=A0The only issue is the there is currently no way to map an UPDA=
TE received in the early dialog to WebRTC API calls. I am probably OK with =
it not being handled until the cloning is implemented, but this shows that =
PRANSWER is not very useful even for the serial forking.<br>
_____________<br>Roman Shpount<br>
<br><br>

--14dae9cdc6c16bfa2e04cb96a5b4--

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From: Cullen Jennings <fluffy@iii.ca>
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On Oct 8, 2012, at 18:56 , Roman Shpount <roman@telurix.com> wrote:

> On Mon, Oct 8, 2012 at 9:41 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
> On Oct 8, 2012, at 18:31 , Roman Shpount <roman@telurix.com> wrote:
>=20
> > On Mon, Oct 8, 2012 at 9:26 PM, Cullen Jennings (fluffy) =
<fluffy@cisco.com> wrote:
> >
> > Well, what's the end user experience you're trying to achieve  - =
lets say we are talking about audio. Do you want to play the first one =
or the last one? If you want to bridge them into a party line, this is =
probably not how you want the SIP side to happen.
> >
> >
> > This is still serial forking. Imagine the following scenario: You =
are calling a phone number, you dial first provider who starts playing a =
dial tone then uses an UPDATE to connect you to a media server to play =
an announcement after which your call goes to voice mail (another dialog =
and another SDP). This is all fairly legal and not unusual for SIP =
scenarios. Nothing to do with party lines.
>=20
> Ok, serial forking with early media seems like it should work. Seems =
like the browser would make the offer, that gateway turned into SIP =
invite that then went to sequential forking proxy that send sent the =
invite to UA A. UA A returns a 180 that gets mapped to PRANSWER, the =
proxy then forks to UA B that sends a 200 on a different dialog that =
gets mapped to answer.
>=20
>  The only issue is the there is currently no way to map an UPDATE =
received in the early dialog to WebRTC API calls. I am probably OK with =
it not being handled until the cloning is implemented, but this shows =
that PRANSWER is not very useful even for the serial forking.
>=20

How do you think this work in a SIP only system? Lets say A sends the =
invite with offer to proxy that forks to B. Then B does 180Rel followed =
by an update. Then the proxy forks to C on a new dialog. Do you think C =
can send send early media to A and A is actually going to play it? And =
what if B use still sending media - it that case A will be getting media =
from B and C at the same time. Do you have devices that do that?  Most =
people just seem to make voicemail work fine without needing to =
introduce this type of complexity. If you know what you want A to do in =
this case, I think we can figure out how to make that happen - there's =
been a lot of discussion about what a SIP device receiving early media =
from two places should do and there was never any really great answer =
other than don't do that. =20

The meta issue here is thought this is serial forking at the signalling =
level in SIP, it is actually parallel forking from a SDP / media  point =
of view.=20

The case that started this all was use UPDATE in awaited RFC says is not =
allowed. I'm not arguing there are not cases where clone would be =
useful, but I do think all the voicemail systems I am familiar with =
would work fine with the the simple PRANSWER.=20




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--047d7bf0d174ff06d204cb97577c
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Changing subject

On Mon, Oct 8, 2012 at 10:30 PM, Cullen Jennings <fluffy@iii.ca> wrote:

> >  The only issue is the there is currently no way to map an UPDATE
> received in the early dialog to WebRTC API calls. I am probably OK with it
> not being handled until the cloning is implemented, but this shows that
> PRANSWER is not very useful even for the serial forking.
> >
>
> How do you think this work in a SIP only system? Lets say A sends the
> invite with offer to proxy that forks to B. Then B does 180Rel followed by
> an update. Then the proxy forks to C on a new dialog. Do you think C can
> send send early media to A and A is actually going to play it? And what if
> B use still sending media - it that case A will be getting media from B and
> C at the same time. Do you have devices that do that?  Most people just
> seem to make voicemail work fine without needing to introduce this type of
> complexity. If you know what you want A to do in this case, I think we can
> figure out how to make that happen - there's been a lot of discussion about
> what a SIP device receiving early media from two places should do and there
> was never any really great answer other than don't do that.
>
> The meta issue here is thought this is serial forking at the signalling
> level in SIP, it is actually parallel forking from a SDP / media  point of
> view.
>
> The case that started this all was use UPDATE in awaited RFC says is not
> allowed. I'm not arguing there are not cases where clone would be useful,
> but I do think all the voicemail systems I am familiar with would work fine
> with the the simple PRANSWER.
>
>
This actually works fairly well in the current systems. The fact that
UPDATE is used does not mean there are multiple media streams flowing at
any given time. And, I am familiar with a couple of systems that do exactly
what I have described. Typically some sort of announcement or
color-ring-back system followed by a voice mail. I think the problem is
that this group picked one interop scenario and added something to support
it while arbitrarily deciding that other scenarios should not be supported.
SIP does not have serial forking. Offer/Answer does not describe something
like this. We either need to define it and make it something that describes
some set of existing scenarios or add support for something that is defined.
_____________
Roman Shpount

--047d7bf0d174ff06d204cb97577c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Changing subject<div><br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at =
10:30 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@ii=
i.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">&gt; =A0The only issue is the there=
 is currently no way to map an UPDATE received in the early dialog to WebRT=
C API calls. I am probably OK with it not being handled until the cloning i=
s implemented, but this shows that PRANSWER is not very useful even for the=
 serial forking.<br>

&gt;<br>
<br>
</div></div>How do you think this work in a SIP only system? Lets say A sen=
ds the invite with offer to proxy that forks to B. Then B does 180Rel follo=
wed by an update. Then the proxy forks to C on a new dialog. Do you think C=
 can send send early media to A and A is actually going to play it? And wha=
t if B use still sending media - it that case A will be getting media from =
B and C at the same time. Do you have devices that do that? =A0Most people =
just seem to make voicemail work fine without needing to introduce this typ=
e of complexity. If you know what you want A to do in this case, I think we=
 can figure out how to make that happen - there&#39;s been a lot of discuss=
ion about what a SIP device receiving early media from two places should do=
 and there was never any really great answer other than don&#39;t do that.<=
br>

<br>
The meta issue here is thought this is serial forking at the signalling lev=
el in SIP, it is actually parallel forking from a SDP / media =A0point of v=
iew.<br>
<br>
The case that started this all was use UPDATE in awaited RFC says is not al=
lowed. I&#39;m not arguing there are not cases where clone would be useful,=
 but I do think all the voicemail systems I am familiar with would work fin=
e with the the simple PRANSWER.<br>

<br></blockquote><div><br></div><div>This actually works fairly well in the=
 current systems. The fact that UPDATE is used does not mean there are mult=
iple media streams flowing at any given time. And, I am familiar with a cou=
ple of systems that do exactly what I have described. Typically some sort o=
f announcement or color-ring-back system followed by a voice mail. I think =
the problem is that this group picked one interop scenario and added someth=
ing to support it while arbitrarily deciding that other scenarios should no=
t be supported. SIP does not have serial forking. Offer/Answer does not des=
cribe something like this. We either need to define it and make it somethin=
g that describes some set of existing scenarios or add support for somethin=
g that is defined.</div>
_____________<br>Roman Shpount<br><br><div>=A0</div></div></div>

--047d7bf0d174ff06d204cb97577c--

From christer.holmberg@ericsson.com  Mon Oct  8 23:23:36 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Tue, 9 Oct 2012 08:23:29 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

> Uh, sort of  - if you send an offer that says you could get early media, =
you may get early media from two places even thought you have not received =
any answer.=20

Sure, but that's separate from the O/A state machine, which is still per di=
alog.

Regards,

Christer



On Oct 5, 2012, at 14:26 , Christer Holmberg <christer.holmberg@ericsson.co=
m> wrote:

> Hi,
>=20
> Roman is correct.
>=20
> The SDP O/A doesn't consider forking, because each forked SIP leg creates=
 a unique early dialog, and each early dialog has its own O/A state machine=
.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> ________________________________
> From: Roman Shpount [roman@telurix.com]
> Sent: Friday, October 05, 2012 8:46 PM
> To: Harald Alvestrand
> Cc: Christer Holmberg; rtcweb@ietf.org
> Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
>=20
>=20
> On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand <harald@alvestrand.no<m=
ailto:harald@alvestrand.no>> wrote:
> I've asked that question before, but I don't remember seeing an answer fr=
om people who know what they're talking about:
>=20
> Does serial forking, as practiced by SIP UAs, violate the SDP O/A model, =
or does it not violate the SDP O/A model?
>=20
> I don't understand how it, in the form that has been described as "what w=
e have to support", can be SDP O/A compatible, but then there are many thin=
gs about SIP that I don't understand, so I may understand it when it's expl=
ained to me.
>=20
>=20
> O/A does not describe multiple answers to a single offer. Nothing=20
> explicitly prohibits it there but I would argue this is not part of this =
specification. No standard document that I am aware of describes how serial=
 O/A is supposed to work. Serial forking as practiced by SIP UA does violat=
e SIP RFC 3261 which states that each dialog can only have one answer to ea=
ch offer. If answer is sent in both provisional and final response, it shou=
ld be the same. You can, however, create multiple dialogs with the same off=
er. This normally implies parallel forking, but the most common use case is=
 with early media, where you end up with multiple early dialogs. For instan=
ce you call a phone number, phone network sends you SDP in early media and =
plays a dial tone, then the called number does not answer, and you get conn=
ected to a voice mail which uses a completely different SDP in final answer=
. According to SIP these answers should come with different to-tags and tec=
hnically would be parts of  two different dialogs. What makes this a bit tr=
icky is when the phone network and the voice mail are behind SBC (or some o=
ther sort of B2BUA) you only see one dialog which send you multiple answers=
 to the same offer. This scenario is so common that it is more likely to be=
 implemented then parallel forking. This is why PRANSWER was suggested.
>=20
> If cloning can be implemented in WebRTC it would be more standard complia=
nt then PRANSWER and would allow for a lot more use cases. Typical difficul=
ty in parallel forking implementation in SIP is due to RTP from multiple so=
urces coming to the same IP and port with no consent or notification to the=
 receiving side. This makes it very difficult to present this media to the =
user. There is no way to tie media to actual dialogs since RTP can come fro=
m different IP and ports then specified in answer SDPs. This is not an issu=
e with WebRTC where consent to receive media is required. I would argue let=
's implement cloning and not waste any more time on PRANSWER which in my op=
inion will always stay a half working hack.
> _____________
> Roman Shpount
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Tue, 9 Oct 2012 08:26:37 +0200
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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Hi,

>>>>>> Can I use the same local descriptor for every setLocal() call?
>>>>>=20
>>>>> My experience suggests that you can.  However, that's not=20
>>>>> stipulated in the API specification, so it's not an ironclad guarante=
e.
>>>>=20
>>>> If it doesn't, and a new local descripor is created, you would need=20
>>>> to send that one to the remote endpoint.
>>>=20
>>> You don't create a new description, you just set one.
>>=20
>> Correct.
>
>> The only problem arises when you can't set the local description=20
>> because something broke since you last set it.  (Maybe setting the=20
>> answer caused the browser to release something that you need to set=20
>> the
>
> If you send an answer that has not relation to the offer, you are equally=
 hosed. We are keeping the pairing of the offers and answers in the JS stat=
e so there are ways that the JS can mess things up if it is buggy but that =
not really a PRANSWER problem -=20
> it's just as much a issue with offers and answers.=20

I am not sure what you mean by "an answer that has not relation to the offe=
r". Answers are always related to the Offers.

Regards,

Christer

From christer.holmberg@ericsson.com  Mon Oct  8 23:28:43 2012
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Date: Tue, 9 Oct 2012 08:28:39 +0200
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Hi Ekr,

>I would attend such a phone call if it were held.
>
>My immediate reaction is that from an implementor's perspective. PRANSWER =
is a lot easier to address than cloning. I feel like cloning pulls in a lot=
 of difficult lifetime and ownership semantics. By contrast, supporting PRA=
NSWER is very easy. Before >deciding that cloning is the answer I'd like to=
 see two
>things:
>
>1. A clear set of use cases that shows what cloning does well that PRANSWE=
R does not.
>2. A fairly complete defn of the clone semantics.
>
>Otherwise I worry that we'll detour to cloning and then only after 6 month=
s discover it doesn't work.

Please note that JSEP-02 already supports cloning (the details on how it's =
done are missing, though).

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Cullen Jennings <fluffy@iii.ca>
Date: Tue, 9 Oct 2012 08:40:59 +0200
Thread-Topic: [rtcweb] PRANSWER is unusable for serial forking
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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Hi,

I THINK Roman is talking about the following flow:


BROWSER		JS APP			FORKING PROXY			SIP UA 1			SIP UA 2

------- OFFER --------------->
                                                  ----- INVITE (OFFER 1) --=
------->
							----- INVITE (OFFER 1) --------->
							----- INVITE (OFFER 1) ---------------------------------------------=
------->
							<----180 (ANSWER 1a) ----------
			<----180 (ANSWER 1a) ----------
<----PRANSWER -----------
							<----UPDATE (OFFER 2) --------
			<----UPDATE (OFFER 2) ---------
<----???? --------------------
							<----200 (ANSWER 1b) -----------------------------------------------=
-------
			<----200 (ANSWER 1b) ----------
<----ANSWER ---------------


What is "????"?

Note that this is NOT about parallel forking, and which media is played to =
the user - the JS APP can make that decision based on whatever policy. The =
question is how the UPDATE is mapped to the JSEP API.

Regards,

Christer



From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: 9. lokakuuta 2012 5:46
To: Cullen Jennings
Cc: rtcweb@ietf.org
Subject: [rtcweb] PRANSWER is unusable for serial forking

Changing subject

On Mon, Oct 8, 2012 at 10:30 PM, Cullen Jennings <fluffy@iii.ca> wrote:
> =A0The only issue is the there is currently no way to map an UPDATE recei=
ved in the early dialog to WebRTC API calls. I am probably OK with it not b=
eing handled until the cloning is implemented, but this shows that PRANSWER=
 is not very useful even for the serial forking.
>
How do you think this work in a SIP only system? Lets say A sends the invit=
e with offer to proxy that forks to B. Then B does 180Rel followed by an up=
date. Then the proxy forks to C on a new dialog. Do you think C can send se=
nd early media to A and A is actually going to play it? And what if B use s=
till sending media - it that case A will be getting media from B and C at t=
he same time. Do you have devices that do that? =A0Most people just seem to=
 make voicemail work fine without needing to introduce this type of complex=
ity. If you know what you want A to do in this case, I think we can figure =
out how to make that happen - there's been a lot of discussion about what a=
 SIP device receiving early media from two places should do and there was n=
ever any really great answer other than don't do that.

The meta issue here is thought this is serial forking at the signalling lev=
el in SIP, it is actually parallel forking from a SDP / media =A0point of v=
iew.

The case that started this all was use UPDATE in awaited RFC says is not al=
lowed. I'm not arguing there are not cases where clone would be useful, but=
 I do think all the voicemail systems I am familiar with would work fine wi=
th the the simple PRANSWER.

This actually works fairly well in the current systems. The fact that UPDAT=
E is used does not mean there are multiple media streams flowing at any giv=
en time. And, I am familiar with a couple of systems that do exactly what I=
 have described. Typically some sort of announcement or color-ring-back sys=
tem followed by a voice mail. I think the problem is that this group picked=
 one interop scenario and added something to support it while arbitrarily d=
eciding that other scenarios should not be supported. SIP does not have ser=
ial forking. Offer/Answer does not describe something like this. We either =
need to define it and make it something that describes some set of existing=
 scenarios or add support for something that is defined.
_____________
Roman Shpount
=A0

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Date: Tue, 9 Oct 2012 08:41:17 +0200
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Hi,

>> I would like to discuss the different alternatives in order to support f=
orking, e.g. whether we use cloning, whether we simply set additional local=
 descriptor, and whether we can get rid of PRANSWER.
>
> Seriously? we have discussed this so many times and always come to the sa=
me conclusion. I have not seen anything on the list that suggests why we ne=
ed to remove this or how mapping to SIP 180 with sequential forking is goin=
g to work without it.

You CAN do serial forking also with cloning (or, as suggested by Martin, si=
mply setting additional local descriptors).

I am not blindly suggesting to remove PRANSWER (I have been one of the defe=
nders for having it). It's part of the more general "JSEP Offer/Answer Usag=
e" issue, which we DO need to make sure we get right.

> It also has other important uses. There are a bunch of changes that are n=
eeded to the JSEP draft to remove some of the inconsistencies in this and c=
larify some parts but I'd rather wait till we had that updated before we go=
t into a whole discussion about=20
> exploding it yet again.=20

I have no problem with that. But, again, we do need to get the JSEP O/A sor=
ted out. My suggesting is still that the base should be RFC 3264, and if we=
 need to "relax" things we should very carefully look at those cases.

> Why don't we have a phone call to try and outline what the problems you a=
re trying to solve that the current solution does not work for then figure =
out how much we want to explode this.=20

Sure. I'd attend such call.

Regards,

Christer


From paalh@ifi.uio.no  Tue Oct  9 01:30:32 2012
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Subject: [rtcweb] PhD position available in the area of in Time-Dependent Networking
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AVAILABLE PHD POSITION

At Simula Research Laboratory (Norway) there is an available PhD =
position in the area of in Time-Dependent Networking. The PhD Student =
will work in the area of networking support for time-dependent =
applications, focusing on thin streams. Thin streams are generated by =
applications that send data with high interarrival time and also often =
small packet sizes. These traffic patterns are in stark contrast to =
greedy streams where the application always has data to send and the =
sending rate is limited by congestion control. Typical for thin streams =
is that they are generated by interactive applications with a need for =
low latency communication.

The following topics are central for the work in this position:

=95 Develop mechanisms to improve thin-stream latency when transmitted =
via different buffer types with varying drop schemes.
=95 Investigate the congestion control support of conditionally thin =
streams (i.e. not everything must be sent at all times) via per-packet =
priorities.
=95 Work towards a formal definition of the concept of thin streams.
=95 Work towards defining what TCP fairness must mean in thin-stream =
scenarios.



The position is connected to the "Traffic behaviour of interactive =
time-dependent thin streams on the modern Internet" (TimeIn) research =
project cooperating with partners like UNINETT, University of Oslo, =
Karlstad University, University of Kaiserslautern, CAIDA San Diego, =
Cisco and Funcom.

It is planned to align the research with, and possibly contribute to, =
ongoing standardization efforts in the RTP Media Congestion Avoidance =
Techniques (RMCAT) working group in the IETF.


For more information about the project, the requirements, Simula, =
salary, etc.,=20
please see
  http://simula.no/jobs/PhDStudent-TimeIn

or contact:
  Postdoc Andreas Petlund, e-mail: apetlund@simula.no or
  Professor P=E5l Halvorsen, e-mail: paalh@simula.no
(where you include the following phrase in the subject field: =93TimeIn =
PhD position=94)

The application deadline is as soon as possible, but no later than 20 =
October 2012.

To apply for this position, click here:=20
=
https://emea3.recruitmentplatform.com/appproc/index.cfm?event=3DcreateSess=
ionAfterSessionClear&ID=3DQL5FK026203F3VBQBV779QWAD&jobboard=3D0&nPTID=3D2=
8&bSessionClear=3Dtrue=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 9 Oct 2012 11:24:38 +0200
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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2012/10/9 Roman Shpount <roman@telurix.com>:
> SIP does not have serial forking.

SIp has serial forking but just at signaling level, not at media level :)


> Offer/Answer does not describe something
> like this.

+1


> We either need to define it and make it something that describes
> some set of existing scenarios or add support for something that is defin=
ed.

IMHO the solution is described in other mail thread (maybe in rtcweb
WG): implement cloning. Example:

- My JS SIP app sends an INVITE with SDP and receives a 183 response
with To-tag "AAAA" and SDP A, so I already have a media stream.

- Later the voicemail system answers the call so replies a 200 with
To-tag "BBBB" and SDP B. Since To-tag is different I do know (as per
RFC 3261/3264) that I should create another media "session" sharing my
local SDP (but cloning it). And I also want to close the previous one.

Of course, for this we need some API method like
"cloneLocalDescription(new_remote_sdp_object)" or whatever.

AFAIR this has been recently discussed in rtcweb WG.



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

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On Oct 8, 2012, at 23:26 , Christer Holmberg <christer.holmberg@ericsson.co=
m>
 wrote:

> Hi,
>=20
>>>>>>> Can I use the same local descriptor for every setLocal() call?
>>>>>>=20
>>>>>> My experience suggests that you can.  However, that's not=20
>>>>>> stipulated in the API specification, so it's not an ironclad guarant=
ee.
>>>>>=20
>>>>> If it doesn't, and a new local descripor is created, you would need=20
>>>>> to send that one to the remote endpoint.
>>>>=20
>>>> You don't create a new description, you just set one.
>>>=20
>>> Correct.
>>=20
>>> The only problem arises when you can't set the local description=20
>>> because something broke since you last set it.  (Maybe setting the=20
>>> answer caused the browser to release something that you need to set=20
>>> the
>>=20
>> If you send an answer that has not relation to the offer, you are equall=
y hosed. We are keeping the pairing of the offers and answers in the JS sta=
te so there are ways that the JS can mess things up if it is buggy but that=
 not really a PRANSWER problem -=20
>> it's just as much a issue with offers and answers.=20
>=20
> I am not sure what you mean by "an answer that has not relation to the of=
fer". Answers are always related to the Offers.

Sorry - that was poorly written. Lets say you had two PeerConnection call t=
hem A and B and they both create an offer and far sides sends answers to bo=
th offers. Now if you take the answer from offer B and use it as the answer=
 for A, all bets are off and bad things will happen. That is what I meant b=
y if you use an answer that had no relation to offer it needed to be associ=
ated with.=20

>=20
> Regards,
>=20
> Christer


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On Oct 8, 2012, at 23:41 , Christer Holmberg <christer.holmberg@ericsson.co=
m>
 wrote:

> Hi,
>=20
>>> I would like to discuss the different alternatives in order to support =
forking, e.g. whether we use cloning, whether we simply set additional loca=
l descriptor, and whether we can get rid of PRANSWER.
>>=20
>> Seriously? we have discussed this so many times and always come to the s=
ame conclusion. I have not seen anything on the list that suggests why we n=
eed to remove this or how mapping to SIP 180 with sequential forking is goi=
ng to work without it.
>=20
> You CAN do serial forking also with cloning (or, as suggested by Martin, =
simply setting additional local descriptors).
>=20
> I am not blindly suggesting to remove PRANSWER (I have been one of the de=
fenders for having it). It's part of the more general "JSEP Offer/Answer Us=
age" issue, which we DO need to make sure we get right.

agree - sounds like we are on same page=20

>=20
>> It also has other important uses. There are a bunch of changes that are =
needed to the JSEP draft to remove some of the inconsistencies in this and =
clarify some parts but I'd rather wait till we had that updated before we g=
ot into a whole discussion about=20
>> exploding it yet again.=20
>=20
> I have no problem with that. But, again, we do need to get the JSEP O/A s=
orted out. My suggesting is still that the base should be RFC 3264, and if =
we need to "relax" things we should very carefully look at those cases.

I have not seen any reason to relax 3264 yet but if something comes up, agr=
ee we should carefully look at the cases. I think we can just do straight u=
p 3264. Arguments that SIP early media in a 180 is not compliant with 3264 =
are just wrong.

>=20
>> Why don't we have a phone call to try and outline what the problems you =
are trying to solve that the current solution does not work for then figure=
 out how much we want to explode this.=20
>=20
> Sure. I'd attend such call.
>=20
> Regards,
>=20
> Christer
>=20


From martin.thomson@gmail.com  Tue Oct  9 09:09:53 2012
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From: Martin Thomson <martin.thomson@gmail.com>
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On 9 October 2012 07:16, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> I have not seen any reason to relax 3264 yet but if something comes up, agree we should carefully look at the cases. I think we can just do straight up 3264.

RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.

   The offerer MAY immediately cease listening for media formats that
   were listed in the initial offer, but not present in the answer.

"the" answer.

> Arguments that SIP early media in a 180 is not compliant with 3264 are just wrong.

I'd be pleased if you could find something concrete that supports
this, because I just re-read 3264 and I can't find anything supporting
your conclusion.

I have no problem with the idea of a well-established status quo that
does not conform to the documents.  That would be consistent with many
other RFCs.  However, if that is the case, perhaps someone could write
this behavior down so that we can use the same definitions.  Clearly,
that isn't happening right now.

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From: "Richard Shockey" <richard@shockey.us>
To: "'Cullen Jennings'" <fluffy@iii.ca>, "'Roman Shpount'" <roman@telurix.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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+1 to that idea.

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
<mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Cullen Jennings
Sent: Monday, October 08, 2012 8:48 PM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting


One other thought on this. I think this just points out that it would be
really nice to have a draft that showed how to build a RTCWeb / SIP gateway.


 
On Oct 8, 2012, at 17:42 , Cullen Jennings <fluffy@cisco.com> wrote:

> 
> Check out text from RFC 3311 that says 
> 
>      o  If the UPDATE is being sent before the completion of the INVITE
>         transaction, and the initial INVITE contained an offer, the
>         UPDATE cannot be sent with an offer unless the callee has
>         generated an answer in a reliable provisional response, has
>         received a PRACK for that reliable provisional response, has
>         not received any requests (PRACK or UPDATE) with offers that it
>         has not answered, and has not sent any UPDATE requests
>         containing offers that have not been answered.
> 
> So the 183 would be need to be sent reliably at which case I would expect
the gateways to map it to a final offer in RTCWeb not a provisional one. At
that point when the update comes along with an offer, it just mapped to
RTCweb offer. 
> 
> 
> On Oct 8, 2012, at 16:21 , Roman Shpount <roman@telurix.com>
> wrote:
> 
>> The scenario is:
>> 
>> WebRTC is placing a call through some sort of gateway to a SIP end point.
It sends an offer that gets translated into INVITE with SDP. SIP end point
sends back 183 with SDP that gets delivered to the WebRTC end point as
PRANSWER. Then SIP end point sends an UPDATE with SDP in early dialog. And
then finally SIP end point sends 200 OK with different SDP. I am not sure
how to translate the SDP in UPDATE and the final answer in WebRTC API calls.
I would guess this is not something that can be supported without cloning.
>> _____________
>> Roman Shpount
>> 
>> 
>> On Mon, Oct 8, 2012 at 7:09 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>> 
>> Walk me through what the call flow would like it both endpoints were SIP
UA's so I understand it better.
>> 
>> 
>> Note - I'm not against folks trying to figure out clone. I'm saying that
we should not get rid of PRANSWER.
>> 
>> 
>> On Oct 8, 2012, at 15:58 , Roman Shpount <roman@telurix.com> wrote:
>> 
>>> Cullen,
>>> 
>>> I have an existing interop problem with SIP serial forking that I cannot
solve with the proposed schema. With PRANSWER, how would I handle SIP UPDATE
sith SDP received in early dialog?
>>> 
>>> Regards,
>>> _____________
>>> Roman Shpount
>>> 
>>> 
>>> On Mon, Oct 8, 2012 at 6:53 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com> wrote:
>>> 
>>> On Oct 8, 2012, at 2:47 , Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>>> 
>>>> Hi,
>>>> 
>>>> I would like to discuss the different alternatives in order to support
forking, e.g. whether we use cloning, whether we simply set additional local
descriptor, and whether we can get rid of PRANSWER.
>>> 
>>> Seriously? we have discussed this so many times and always come to the
same conclusion. I have not seen anything on the list that suggests why we
need to remove this or how mapping to SIP 180 with sequential forking is
going to work without it. It also has other important uses. There are a
bunch of changes that are needed to the JSEP draft to remove some of the
inconsistencies in this and clarify some parts but I'd rather wait till we
had that updated before we got into a whole discussion about exploding it
yet again.
>>> 
>>> Why don't we have a phone call to try and outline what the problems you
are trying to solve that the current solution does not work for then figure
out how much we want to explode this.
>>> 
>>> I'll note that current clone text has lots of "miracles happen here,
insert supper fluffy hand wave" in it and plenty of weasel room on failure
to allocate required resources on the clone. It's more or less a sketch of
an idea at this point. I'm perfectly happy to see people try and sort out
the details on clone but using it explode the consensus we have come to
around PRANSWER seems like a really bad idea at this point.
>>> 
>>> Cullen
>>> 
>>> _______________________________________________
>>> 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: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Tue, 9 Oct 2012 21:04:11 +0200
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Hi,

>> I have not seen any reason to relax 3264 yet but if something comes up, =
agree we should carefully look at the cases. I think we can just do straigh=
t up 3264.
>
>RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>
>   The offerer MAY immediately cease listening for media formats that
>   were listed in the initial offer, but not present in the answer.
>
> "the" answer.

I agree with Martin. 3264 O/A is always per dialog, and forking is supporte=
d by generating multiple dialogs. JSEP, OTOH, in order to support forking w=
ith a single "dialog" (peerConnection local descriptor), now defines O/A as=
 offer+any number of pranswers + answer.

So, we would e.g. have to define what happens if a new offer is received fr=
om the remote side while the browser is in pranswer-received state (see my =
call flow in another reply).

Regards,

Christer=

From fluffy@cisco.com  Tue Oct  9 13:05:48 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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On Oct 9, 2012, at 12:04 , Christer Holmberg <christer.holmberg@ericsson.co=
m>
 wrote:

> Hi,
>=20
>>> I have not seen any reason to relax 3264 yet but if something comes up,=
 agree we should carefully look at the cases. I think we can just do straig=
ht up 3264.
>>=20
>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>>=20
>>  The offerer MAY immediately cease listening for media formats that
>>  were listed in the initial offer, but not present in the answer.
>>=20
>> "the" answer.
>=20
> I agree with Martin. 3264 O/A is always per dialog, and forking is suppor=
ted by generating multiple dialogs. JSEP, OTOH, in order to support forking=
 with a single "dialog" (peerConnection local descriptor), now defines O/A =
as offer+any number of pranswers + answer.
>=20
> So, we would e.g. have to define what happens if a new offer is received =
from the remote side while the browser is in pranswer-received state (see m=
y call flow in another reply).


3264, SDP, 3261, and related documents are dealing with a bunch of things i=
ncluding what happens at media plane and signaling plane.=20

I'll note that though O/A is per dialog, there is only one O shared across =
multiple legs and when you create the O you don't know how many dialogs the=
re will be. So from the media point of view (covered more in SDP spec than =
3264) there is one O with a bunch of A. From signaling point of view there =
are a bunch of O/A pairs).=20

The dialog is SIP signaling concept not a media plane level concept. We mov=
ed the signaling part out of the browser and into the JS. But the media par=
t is still in the browser. So as 3264 says, after the offer is constructed,=
 we have to be willing to receive media for all the codec type in the offer=
. When we get the answer 3264 makes it clear that one MAY stop receiving th=
e codecs that were in the offer but not selected in the answer. However, th=
is can not be done until the signaling layer is sure that no more offers wi=
ll be honored. Since that signalling part of Offer/Answer is in the JS, the=
 API need to to have a way to signal what the MAY part should do and that i=
s the PRANSWER vs ANSWER.=20

People keep trying to make this some complex weird argument invoking the SI=
P deities of the past and quoting incomprehensible phrases from various RFC=
 caved in stone but the bottom line in the code is very simple to understan=
d. When creating the offer you alloced some resources ( like a port to rece=
ive video on ). When you get an answer that does not use that resource, you=
 need to tell the media stack if it should free the resources or not. O/A h=
as situation where you need to keep the resources available (like there are=
 more dialogs coming) and situation where you need to free the resource.  S=
ince we split the signalling out of the browser and left the media in the b=
rowser, we need be able to allow the JS that is dealing with signaling to t=
ell the browser when to free the resource.=20







From martin.thomson@gmail.com  Tue Oct  9 15:58:50 2012
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From: Martin Thomson <martin.thomson@gmail.com>
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On 9 October 2012 13:05, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> 3264, SDP, 3261, and related documents are dealing with a bunch of things=
 including what happens at media plane and signaling plane.
>
> I'll note that though O/A is per dialog, there is only one O shared acros=
s multiple legs and when you create the O you don't know how many dialogs t=
here will be. So from the media point of view (covered more in SDP spec tha=
n 3264) there is one O with a bunch of A. From signaling point of view ther=
e are a bunch of O/A pairs).
>
> The dialog is SIP signaling concept not a media plane level concept. We m=
oved the signaling part out of the browser and into the JS. But the media p=
art is still in the browser. So as 3264 says, after the offer is constructe=
d, we have to be willing to receive media for all the codec type in the off=
er. When we get the answer 3264 makes it clear that one MAY stop receiving =
the codecs that were in the offer but not selected in the answer. However, =
this can not be done until the signaling layer is sure that no more offers =
will be honored. Since that signalling part of Offer/Answer is in the JS, t=
he API need to to have a way to signal what the MAY part should do and that=
 is the PRANSWER vs ANSWER.

That's a consistent and logical view.  It is not what RFC 3264 says.
It may even be the only reasonable solution.

> People keep trying to make this some complex weird argument invoking the =
SIP deities of the past and quoting incomprehensible phrases from various R=
FC caved in stone [...]

Quit beating around the bush: it's me you are talking about.

It's not a complex weird argument, it's this:
  PRANSWER is not described in RFC 3264.

draft-*-jsep might say something about it, but I'd have to guess about
what to do if I were to implement the feature and that doesn't help
with interoperation.

If there was text that described the above, plus the implications in
some detail (including what resources can be released as they relate
to the SDP), then we're good.  I don't even think that writing this is
especially hard. As it stands, the definition for cloning is on par
with the definition for PRANSWER.

From harald@alvestrand.no  Tue Oct  9 23:38:44 2012
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Subject: [rtcweb] Fwd: WG poll for consensus MSID draft
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If people feel that this should be progressed, now is a good time to say 
so in MMUSIC.

           Harald


-------- Original Message --------
Subject: 	WG poll for consensus MSID draft
Date: 	Tue, 9 Oct 2012 15:24:04 +0200
From: 	Miguel A. Garcia <Miguel.A.Garcia@ericsson.com>
To: 	mmusic <mmusic@ietf.org>
CC: 	Flemming Andreasen <fandreas@cisco.com>, Harald Alvestrand 
<harald@alvestrand.no>



At the last IETF meeting we had a presentation of the following draft:

http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/

At that time, there was interest in solving the problem of cross session
stream identification.

We would like to poll the working group for consensus on adopting the
above mentioned draft as a working group item, once we get an approved
milestone.

If you support this work or have problems with the adoption of this
document as WG item, please indicate it by replying to this e-mail within
the next 5 days.

Flemming and Miguel (co-chairs)

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain





--------------020204040001050109050709
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    If people feel that this should be progressed, now is a good time to
    say so in MMUSIC.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>WG poll for consensus MSID draft</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 9 Oct 2012 15:24:04 +0200</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>Miguel A. Garcia <a class="moz-txt-link-rfc2396E" href="mailto:Miguel.A.Garcia@ericsson.com">&lt;Miguel.A.Garcia@ericsson.com&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>mmusic <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td>Flemming Andreasen <a class="moz-txt-link-rfc2396E" href="mailto:fandreas@cisco.com">&lt;fandreas@cisco.com&gt;</a>, Harald
              Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>At the last IETF meeting we had a presentation of the following draft:

<a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/">http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/</a>

At that time, there was interest in solving the problem of cross session 
stream identification.

We would like to poll the working group for consensus on adopting the 
above mentioned draft as a working group item, once we get an approved 
milestone.

If you support this work or have problems with the adoption of this 
document as WG item, please indicate it by replying to this e-mail within 
the next 5 days.

Flemming and Miguel (co-chairs)

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain
</pre>
      <br>
      <br>
    </div>
    <br>
  </body>
</html>

--------------020204040001050109050709--

From christer.holmberg@ericsson.com  Wed Oct 10 01:03:34 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Wed, 10 Oct 2012 10:03:30 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Hi,

>>>> I have not seen any reason to relax 3264 yet but if something comes up=
, agree we should carefully look at the cases. I think we can just do strai=
ght up 3264.
>>>=20
>>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>>>=20
>>>  The offerer MAY immediately cease listening for media formats that =20
>>> were listed in the initial offer, but not present in the answer.
>>>=20
>>> "the" answer.
>>=20
>> I agree with Martin. 3264 O/A is always per dialog, and forking is suppo=
rted by generating multiple dialogs. JSEP, OTOH, in order to support forkin=
g with a=20
>> single "dialog" (peerConnection local descriptor), now defines O/A as of=
fer+any number of pranswers + answer.
>>=20
>> So, we would e.g. have to define what happens if a new offer is received=
 from the remote side while the browser is in pranswer-received state (see =
my call flow in another reply).
>
> 3264, SDP, 3261, and related documents are dealing with a bunch of things=
 including what happens at media plane and signaling plane.=20
>
> I'll note that though O/A is per dialog, there is only one O shared acros=
s multiple legs and when you create the O you don't know how many dialogs t=
here will be.

The *initial* O is shared across multiple legs. Once a dialog has been crea=
ted, O/A transactions on that leg will not affect other legs.

> So from the media point of view (covered more in SDP spec than 3264) ther=
e is one O with a bunch of A. From signaling point of view there are a bunc=
h of O/A pairs).=20
>
> The dialog is SIP signaling concept not a media plane level concept. We m=
oved the signaling part out of the browser and into the JS. But the media p=
art is still in the=20
> browser. So as 3264 says, after the offer is constructed, we have to be w=
illing to receive media for all the codec type in the offer. When we get th=
e answer 3264 makes=20
> it clear that one MAY stop receiving the codecs that were in the offer bu=
t not selected in the answer. However, this can not be done until the signa=
ling layer is sure that=20
> no more offers will be honored. Since that signalling part of Offer/Answe=
r is in the JS, the API need to to have a way to signal what the MAY part s=
hould do and that is=20
> the PRANSWER vs ANSWER.=20
>
> People keep trying to make this some complex weird argument invoking the =
SIP deities of the past and quoting incomprehensible phrases from various R=
FC caved in=20
> stone but the bottom line in the code is very simple to understand. When =
creating the offer you alloced some resources ( like a port to receive vide=
o on ). When you
> get an answer that does not use that resource, you need to tell the media=
 stack if it should free the resources or not. O/A has situation where you =
need to keep the=20
> resources available (like there are more dialogs coming) and situation wh=
ere you need to free the resource.  Since we split the signalling out of th=
e browser and left=20
> the media in the browser, we need be able to allow the JS that is dealing=
 with signaling to tell the browser when to free the resource.=20

I don't know who "people" is, but at least I have only been talking about t=
he O/A *signaling* :) Media is a separate issue.

Regards,

Christer







From pkyzivat@alum.mit.edu  Wed Oct 10 11:42:45 2012
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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 10/5/12 1:46 PM, Roman Shpount wrote:
>
> On Fri, Oct 5, 2012 at 1:13 PM, Harald Alvestrand <harald@alvestrand.no
> <mailto:harald@alvestrand.no>> wrote:
>
>     I've asked that question before, but I don't remember seeing an
>     answer from people who know what they're talking about:
>
>     Does serial forking, as practiced by SIP UAs, violate the SDP O/A
>     model, or does it not violate the SDP O/A model?
>
>     I don't understand how it, in the form that has been described as
>     "what we have to support", can be SDP O/A compatible, but then there
>     are many things about SIP that I don't understand, so I may
>     understand it when it's explained to me.
>
>
> O/A does not describe multiple answers to a single offer. Nothing
> explicitly prohibits it there but I would argue this is not part of this
> specification. No standard document that I am aware of describes how
> serial O/A is supposed to work. Serial forking as practiced by SIP UA
> does violate SIP RFC 3261 which states that each dialog can only have
> one answer to each offer. If answer is sent in both provisional and
> final response, it should be the same. You can, however, create multiple
> dialogs with the same offer. This normally implies parallel forking, but
> the most common use case is with early media, where you end up with
> multiple early dialogs. For instance you call a phone number, phone
> network sends you SDP in early media and plays a dial tone, then the
> called number does not answer, and you get connected to a voice mail
> which uses a completely different SDP in final answer. According to SIP
> these answers should come with different to-tags and technically would
> be parts of two different dialogs.

Good summary. That is indeed the compliant way to handle this case.

> What makes this a bit tricky is when
> the phone network and the voice mail are behind SBC (or some other sort
> of B2BUA) you only see one dialog which send you multiple answers to the
> same offer. This scenario is so common that it is more likely to be
> implemented then parallel forking.

If you see this then you should complain to the vendor of the SBC/B2BUA 
that they are broken. There is nothing to prevent the SBC from 
generating multiple dialogs in order to do this case correctly.

> This is why PRANSWER was suggested.
>
> If cloning can be implemented in WebRTC it would be more standard
> compliant then PRANSWER and would allow for a lot more use cases.
> Typical difficulty in parallel forking implementation in SIP is due to
> RTP from multiple sources coming to the same IP and port with no consent
> or notification to the receiving side. This makes it very difficult to
> present this media to the user.

Yep.

> There is no way to tie media to actual
> dialogs since RTP can come from different IP and ports then specified in
> answer SDPs.

Known problem. There was a proposal *long* ago to solve this by the 
caller generating new O/As (with distinct addr/port) as soon as possible 
after discovering that forking has occurred. But that didn't go 
anywhere. And it would still leave a window where the association of 
address to dialog isn't known.

> This is not an issue with WebRTC where consent to receive
> media is required.

Yes. ICE helps a lot.

> I would argue let's implement cloning and not waste
> any more time on PRANSWER which in my opinion will always stay a half
> working hack.

Sounds good to me.

	Thanks,
	Paul


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Subject: Re: [rtcweb] JSEP: Relaxing SDP O/A rules?
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On 10/5/12 1:13 PM, Harald Alvestrand wrote:
> On 10/04/2012 08:21 PM, Christer Holmberg wrote:
>> Hi,
>>
>>>> For the Web use case, I think PRANSWER is an unnecessary
>>>> distraction; if
>>>> we're writing both initiator and responder, multiple O/A rounds, or
>>>> use of
>>>> multiple PeerConnections, is a much cleaner solution for the use
>>>> cases I can
>>>> wrap my head around.
>>> I'd remove that qualifier.  We currently have a draft that describes
>>> both cloning AND PRANSWER, both of which are more cleanly addressed by
>>> either multiple O/A rounds or multiple independent sessions.
>>>
>>> PRANSWER:
>>> The ability to describe a session that includes receipt capabilities
>>> that are a superset of send capabilities.  SDP O/A doesn't really
>>> provide a way for me to describe this because the default assumption
>>> is that items not appearing in the answer are no longer needed.
>>> With something like bundle involved, we could split m= lines into
>>> sendonly and recvonly portions to allow for this case to be provided.
>> I don't think the reason for PRANSWER comes from SDP O/A as such. It's
>> more about supporting serial forking.
> I've asked that question before, but I don't remember seeing an answer
> from people who know what they're talking about:
>
> Does serial forking, as practiced by SIP UAs, violate the SDP O/A model,
> or does it not violate the SDP O/A model?

In addition to what Roman already said, I'll add:

SIP (3261 & friends) doesn't say much about how to *do* forking. It sets 
down a bunch of rules, and developing algorithms that adhere to the 
rules is left as an exercise for the reader. There are also many things 
that are implicit but that must be inferred via long chains of 
reasoning. Much of this has become folklore. (RFC 6337 attempts to 
explain a lot of O/A, but doesn't directly address forking.)

The use of a separate dialog per fork falls out for free, by the nature 
of how to-tags (part of dialog id) are assigned.

But this is still a point of confusion among new (and some not so new) 
sip implementers. Questions about this come up regularly on the 
sip-implementers list. I think the frequency of them has been maintained 
for as long as I can remember.

So indeed there are ongoing violations, and some implementations are 
built to tolerate those violations.

	Thanks,
	Paul




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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Cullen,

As signaling is moved to JS (application layer), Offer-answer in JSEP has to
be generic enough. The underlying assumption in PRANSWER state is that
webserver knows whether the received SDP from remote is PRANSWER or final
ANSWER. Unfortunately, Webserver may not able to tell whether the receive in
lot of real time deployment which is an open issue now. UPDATE offer after
18x answer from remote is the typical example where PRANSWER state breaks.
The real-time usage of SIP UPDATE offer is to update the existing answer in
18x in the same dialog. Early dialog UPDATE callflow is well deployment
because of PSTN remote ringback (18x) from media server and then UPDATE with
SDP to update the media information of remote endpoing and then, call
connect (200 ok) from actual endpoint. The same PSTN callflow shall be
achieved by serial forking as well. Please note that the final answer or not
is not based on browser or originating SIP UA but it is based on the
intermediate SIP entities. 

>From RTCWeb perspective, The new OFFER is possible to be received in browser
as per RFC 3264 after the first answer is received. PRANSWER is the new
state in JSEP wherein JSEP is the extension of RFC 3264. It will be good in
case you explain how browser in PRANSWER has to handle new OFFER from the
remote side. 

Please include me in case any phone discussion if you are planning.

Thanks
Partha

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Cullen Jennings (fluffy)
Sent: Wednesday, October 10, 2012 1:35 AM
To: Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting


On Oct 9, 2012, at 12:04 , Christer Holmberg
<christer.holmberg@ericsson.com>
 wrote:

> Hi,
> 
>>> I have not seen any reason to relax 3264 yet but if something comes up,
agree we should carefully look at the cases. I think we can just do straight
up 3264.
>> 
>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>> 
>>  The offerer MAY immediately cease listening for media formats that
>>  were listed in the initial offer, but not present in the answer.
>> 
>> "the" answer.
> 
> I agree with Martin. 3264 O/A is always per dialog, and forking is
supported by generating multiple dialogs. JSEP, OTOH, in order to support
forking with a single "dialog" (peerConnection local descriptor), now
defines O/A as offer+any number of pranswers + answer.
> 
> So, we would e.g. have to define what happens if a new offer is received
from the remote side while the browser is in pranswer-received state (see my
call flow in another reply).


3264, SDP, 3261, and related documents are dealing with a bunch of things
including what happens at media plane and signaling plane. 

I'll note that though O/A is per dialog, there is only one O shared across
multiple legs and when you create the O you don't know how many dialogs
there will be. So from the media point of view (covered more in SDP spec
than 3264) there is one O with a bunch of A. From signaling point of view
there are a bunch of O/A pairs). 

The dialog is SIP signaling concept not a media plane level concept. We
moved the signaling part out of the browser and into the JS. But the media
part is still in the browser. So as 3264 says, after the offer is
constructed, we have to be willing to receive media for all the codec type
in the offer. When we get the answer 3264 makes it clear that one MAY stop
receiving the codecs that were in the offer but not selected in the answer.
However, this can not be done until the signaling layer is sure that no more
offers will be honored. Since that signalling part of Offer/Answer is in the
JS, the API need to to have a way to signal what the MAY part should do and
that is the PRANSWER vs ANSWER. 

People keep trying to make this some complex weird argument invoking the SIP
deities of the past and quoting incomprehensible phrases from various RFC
caved in stone but the bottom line in the code is very simple to understand.
When creating the offer you alloced some resources ( like a port to receive
video on ). When you get an answer that does not use that resource, you need
to tell the media stack if it should free the resources or not. O/A has
situation where you need to keep the resources available (like there are
more dialogs coming) and situation where you need to free the resource.
Since we split the signalling out of the browser and left the media in the
browser, we need be able to allow the JS that is dealing with signaling to
tell the browser when to free the resource. 






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


From worley@shell01.TheWorld.com  Wed Oct 10 13:27:10 2012
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Subject: [rtcweb] The SIP gateway use case
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I'm new to this discussion.  Being from the SIP community, my "default
use case" for WebRTC is when the web server routes the WebRTC media to
a SIP gateway, which I expect will be a significant fraction of early
usage of WebRTC.  Within this context, the WebRTC architecture needs
to be able to support the media stream operations that a SIP endpoint
("user agent", UA) must implement.  Those operations are not
immediately clear from a reading of the standards (as many a SIP
implementer has discovered), partly due to the complexity of the
situation and partly because a number of critical problems are not
manifested in the protocol operations, and hence the RFCs do not
address them.  The proper, indeed necessary, operations only become
clear from an extended discussion of various cases and what strategies
will and won't work in practice.

My apologies if any of this duplicates what others have already
described.  I've tried to leave out irrelevant parts of the SIP
signaling.  I may have made some mistakes.  In particular, I'm not
familiar with the details of ICE.  Please correct any mistakes.

I've divided the discussion into three sections
- Background
- The situation the SIP caller is faced with
- Architectural questions

* Background

Let me describe the actions by a SIP endpoint that initiates a call (a
"user agent client", UAC).  The operation of a SIP endpoint that
receives a call (a "user agent server", UAS) is similar but much
simpler, and I will leave it to the reader to construct that
discussion.  Similarly, I will assume that the UAC sends an initial
offer; when the UAC does not send an initial offer, the operations are
similar but simpler.

The UAC sends the INVITE containing the initial offer.  Over time,
this
INVITE reaches various UASs via request forwarding, serial forking,
and parallel forking.  Between the UAC and each UAS, SIP constructs a
"dialog", which is the stream of SIP requests and responses between
them through intermediary SIP entities, and the associated dialog
state.

Because of network delays, because some SIP responses are not reliably
transmitted, and because some SIP responses are absorbed by
intermediate SIP entities, the UAC does not have complete knowledge of
the dialogs it is participating in.

Connected with each dialog is a "session", which is a collection of
media streams.  Each session is described by the two SDPs which have
been sent from each UA to the other, and that description can change
over time due to SIP messages carrying SDP offers and answers.  The
sending and receiving of SDP between the UAC and each UAS is
separately governed by the offer/answer rules and has its own
offer/answer state.  (Excepting that all offer/answer states
necessarily start with the same initial offer.)

ICE negotiation is performed within any session for which SDP has been
sent and received.

In practice, the UAC uses a separate port for each media stream
(m-line) in its offer, but for any media stream, uses the same port to
communicate the clone of that media stream to/from each UAS.  In
practice, each UAS uses a different ports from every other UAS, and a
different port for each media stream.  The RTP of a media stream is
not labeled to match it with the governing SDP for its session; the
matching is done heuristically between the port used by the UAS and
the port listed in the SDP.

Because SIP responses may not reach the UAC promptly, the UAC can be
participating in sessions that it has received no SDP for.

A dialog and its session can be terminated in various ways:
- Final termination of the call by sending/receiving a BYE
- Termination of an early dialog by the UAC sending a BYE (uncommon,
but allowed)
- Termination of all but one of several 200 responses received in a
race situation by the UAC sending BYEs
- The UAS sending a failure final response.  (But some such responses
are received by intermediate proxies and not passed on, and trigger
serial forking.)
- The UAS sending a 199 response (or a proxy sending a 199 response on
behalf of the UAS), which may reach the UAC

* The situation the SIP caller is faced with

Thus, the SIP caller (UAC) is participating in a dynamically changing
collection of sessions.

Some of the sessions the UAC has complete knowledge of, and maintains
an offer/answer state for.  The current SDP for a session can be
updated by either UA.  The corresponding RTP streams are identified by
their UAS-end addresses.

Some of the sessions the UAC has no knowledge of, other than that it
is receiving RTP for the session.  The UAC may later receive SDP for
such a session.

The situation will be simplified to one dialog and session when:
- the dialog is ended by the UAC receiving a failure final response to
the initial INVITE
- the call is established by the UAC receiving a 200 final response
(but the UAC must be prepared to receive additional 200 final
responses and terminate those dialogs with BYE)

Regarding the media received in each particular session, there are
several alternatives:
- The media are being received, and are not silence.  This stream
should be mixed into what the user hears, as it contains information
from a UAS.
- The media are absent or silent, but the SIP signaling has indicated
that the far end is ringing (a 180 has been received on this dialog).
For this stream, a ringback signal should be mixed into what the user
hears.
- The media are absent or silent, and the SIP signaling has not
indicated that the far end is ringing.  For this stream, nothing
should be mixed into what the user hears.

Regarding the media sent to each particular session:
- If a session is for an established dialog, the user's voice should
be sent to the (necessarily unique) UAS.
- If a session is the only early dialog, the user's voice should be
sent to the unique UAS.  (This is because some legitimate callees
delay proving a 200 response and have extended information interchange
during the early dialog.)
- If there are multiple early dialogs, practice seems not to be
standardized.

* Architectural questions

In this context, the major architectural questions seem to be the
degree to which this messy reality is revealed to the WebRTC
browser-side code.  The choices on these questions are reflected in
both the wire protocol and the Javascript API.

Does WebRTC reveal the multiple sessions to the multiple UASs?  (It
seems that any sessions for which outgoing RTP can be generated (at
the browser), must be individually revealed to the browser, as each
UAS can provide answer SDP that has no codec in common with other
answer SDPs.)

Does WebRTC reveal/permit later offer/answer cycles for a session?
(Note that the SIP UA can prevent later offer/answer cycles by not
sending re-INVITEs/UPDATEs and giving failure responses to any
received re-INVITEs/UPDATEs.  But this is unlikely to be tolerated in
practice, as later offer/answer cycles are used in many SIP systems.)

Does WebRTC transport signaling-level ring and call failure
information (e.g., the SIP call progress and termination response
codes), or just the initiation and termination events of sessions?

Does WebRTC transport or reveal in the API the complete SDP or just
selected information from it?

Does WebRTC provide facilities for the Javascript to affect the
handling of multiple sessions (e.g., forcibly terminating an early
dialog with undesirable properties)?  (Note that in SIP phones, the
user is usually not provided with any such facility.)

Dale

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Subject: [rtcweb] Relaxing SDP O/A rules?
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> From: Roman Shpount [roman@telurix.com]
> 
> You can, however, create multiple dialogs with the same offer. This
> normally implies parallel forking, but the most common use case is
> with early media, where you end up with multiple early dialogs. For
> instance you call a phone number, phone network sends you SDP in
> early media and plays a dial tone, then the called number does not
> answer, and you get connected to a voice mail which uses a
> completely different SDP in final answer. According to SIP these
> answers should come with different to-tags and technically would be
> parts of two different dialogs.
>
> What makes this a bit tricky is when the
> phone network and the voice mail are behind SBC (or some other sort
> of B2BUA) you only see one dialog which send you multiple answers to
> the same offer. This scenario is so common that it is more likely to
> be implemented then parallel forking. This is why PRANSWER was
> suggested.

Strictly speaking, if the first answer is sent reliably (presumably in
a reliable provisional response), then the first O/A cycle is finished
and a second can be initiated.  What makes this situation interesting
is that the second O/A cycle can be done before any dialog is
established, that is, while further early dialogs may be created.

If the first answer is not sent reliably, then the B2BUA may not send
another offer without violating the O/A rules -- until a duplicate
answer is sent reliably (presumably in the 200 response).

> If cloning can be implemented in WebRTC it would be more standard
> compliant then PRANSWER and would allow for a lot more use
> cases. Typical difficulty in parallel forking implementation in SIP
> is due to RTP from multiple sources coming to the same IP and port
> with no consent or notification to the receiving side. This makes it
> very difficult to present this media to the user. There is no way to
> tie media to actual dialogs since RTP can come from different IP and
> ports then specified in answer SDPs. This is not an issue with
> WebRTC where consent to receive media is required. I would argue
> let's implement cloning and not waste any more time on PRANSWER
> which in my opinion will always stay a half working hack.

In practice, almost all SIP systems send media from the address/port
they specify in their SDP as the address/port from which they receive
media.  But if you implement RTP filtering or "consent to receive",
that does not help, as before call establishment, a SIP caller can
receive media from addresses/ports that it has not yet received SDP
for.

Dale

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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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On 10/09/2012 04:16 PM, Cullen Jennings (fluffy) wrote:
> On Oct 8, 2012, at 23:41 , Christer Holmberg <christer.holmberg@ericsson.com>
>   wrote:
>>
>> I have no problem with that. But, again, we do need to get the JSEP O/A sorted out. My suggesting is still that the base should be RFC 3264, and if we need to "relax" things we should very carefully look at those cases.
> I have not seen any reason to relax 3264 yet but if something comes up, agree we should carefully look at the cases. I think we can just do straight up 3264. Arguments that SIP early media in a 180 is not compliant with 3264 are just wrong.
>
This seems to be true as long as I consider the offer/180 exchange a 
separate SDP exchange from the offer/answer exchange, and abandon the 
concept that the final answer "updates" the 180 answer.

Is that what you are saying, or did I squint the wrong way?


From christer.holmberg@ericsson.com  Thu Oct 11 08:42:14 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 11 Oct 2012 17:38:18 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Hi,

>>> I have no problem with that. But, again, we do need to get the JSEP O/A=
 sorted out. My suggesting is still that the base should be RFC 3264, and i=
f we need to "relax" things we should very carefully look at those cases.
>> I have not seen any reason to relax 3264 yet but if something comes up, =
agree we should carefully look at the cases. I think we can just do straigh=
t up 3264. Arguments that SIP early media in a 180 is not compliant with 32=
64 are just wrong.
>
> This seems to be true as long as I consider the offer/180 exchange a
> separate SDP exchange from the offer/answer exchange, and abandon the
> concept that the final answer "updates" the 180 answer.
>
> Is that what you are saying, or did I squint the wrong way?

If you receive the SDP answer in a reliable 180 response, that IS an offer/=
answer exchange :)

That answer CANNOT be "updated" in another reliable 18x response, or in the=
 200 response.=20

(You can receive a "copy" of the previously received SDP answer in the 18x/=
200 responses, but it will not update the previously received SDP answer)

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu Oct 11 08:45:14 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 11 Oct 2012 17:44:18 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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References: <506B0367.4000103@ericsson.com> <C5E08FE080ACFD4DAE31E4BDBF944EB111867718@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD03A6@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB11187F8F1@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD087E@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1118810C3@xmb-aln-x02.cisco.com>, <5076E4CD.2020602@alvestrand.no>, <7F2072F1E0DE894DA4B517B93C6A0585340BAFAA7E@ESESSCMS0356.eemea.ericsson.se>
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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That answer CANNOT be "updated" in another reliable 18x response, or in the=
 200 response - on the same SIP early dialog, that is.


________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Christ=
er Holmberg [christer.holmberg@ericsson.com]
Sent: Thursday, October 11, 2012 6:38 PM
To: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting

Hi,

>>> I have no problem with that. But, again, we do need to get the JSEP O/A=
 sorted out. My suggesting is still that the base should be RFC 3264, and i=
f we need to "relax" things we should very carefully look at those cases.
>> I have not seen any reason to relax 3264 yet but if something comes up, =
agree we should carefully look at the cases. I think we can just do straigh=
t up 3264. Arguments that SIP early media in a 180 is not compliant with 32=
64 are just wrong.
>
> This seems to be true as long as I consider the offer/180 exchange a
> separate SDP exchange from the offer/answer exchange, and abandon the
> concept that the final answer "updates" the 180 answer.
>
> Is that what you are saying, or did I squint the wrong way?

If you receive the SDP answer in a reliable 180 response, that IS an offer/=
answer exchange :)

That answer CANNOT be "updated" in another reliable 18x response, or in the=
 200 response.

(You can receive a "copy" of the previously received SDP answer in the 18x/=
200 responses, but it will not update the previously received SDP answer)

Regards,

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

From bernard_aboba@hotmail.com  Thu Oct 11 11:58:08 2012
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Subject: [rtcweb] Agenda request: Overall Approach to defining SDP usage in WebRTC
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In addition to covering "fascinating" topics such as MTI codecs and JSEP
support of O/A at IETF 85, I'd like to suggest that we also set some time
aside at IETF 85 to discuss the overall approach to defining SDP usage in
WebRTC.  

 

In this "big picture" discussion, I'd like to try to figure out how we will
answer some of the following questions:

 

1.	What SDP functionality MUST/SHOULD/MAY be implemented/used in WebRTC
API implementations?
2.	What is the overall model for SDP usage in JSEP?
3.	How much of this discussion belongs in W3C, how much in IETF? 

 

The goal of this discussion is *not* to get into the details of the answer
to the questions, but rather to focus on what work needs to be done,  what
form it takes (adding material in existing docs, new docs, etc.), and what
is handled in what org (W3C, IETF, etc.).

 

To some extent, answers to question #1 are derivable from the RTP usage
draft, but I'm not clear that it makes sense to cover all aspects of SDP
usage there.  Also, it would be useful to discuss the plan for documenting
SDP in current usage that has no corresponding RFC or I-D (e.g. are all the
bases covered?). 

 

In question #2, I would include questions such as the role of the
constraints API (versus hacking SDP), what SDP lines/fields can or cannot be
edited by applications and how SDP errors are handled. 

 

While it would certainly be possible to write a -00 draft, without
understanding the overall plan, getting into detail probably doesn't make
much sense.  IMHO, slides should be sufficient, and I have some available to
stimulate discussion. 

 

 

 

 

 

 


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>In =
addition to covering &quot;fascinating&quot; topics such as MTI codecs =
and JSEP support of O/A at IETF 85, I'd like to suggest that we also set =
some time aside at IETF 85 to discuss the overall approach to defining =
SDP usage in WebRTC. &nbsp;<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>In this =
&quot;big picture&quot; discussion, I'd like to try to figure out how we =
will answer some of the following questions:<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><ol style=3D'margin-top:0in' =
start=3D1 type=3D1><li class=3DMsoNormal style=3D'mso-list:l0 level1 =
lfo1'>What SDP functionality MUST/SHOULD/MAY be implemented/used in =
WebRTC API implementations?<o:p></o:p></li><li class=3DMsoNormal =
style=3D'mso-list:l0 level1 lfo1'>What is the overall model for SDP =
usage in JSEP?<o:p></o:p></li><li class=3DMsoNormal style=3D'mso-list:l0 =
level1 lfo1'>How much of this discussion belongs in W3C, how much in =
IETF? <o:p></o:p></li></ol><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>The goal of this discussion is *<b>not</b>* to get =
into the details of the answer to the questions, but rather to focus on =
what work needs to be done,&nbsp; what form it takes (adding material in =
existing docs, new docs, etc.), and what is handled in what org (W3C, =
IETF, etc.).<o:p></o:p></p><p class=3DMsoNormal =
style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>To =
some extent, answers to question #1 are derivable from the RTP usage =
draft, but I'm not clear that it makes sense to cover all aspects of SDP =
usage there.&nbsp; Also, it would be useful to discuss the plan for =
documenting SDP in current usage that has no corresponding RFC or I-D =
(e.g. are all the bases covered?). <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>In question =
#2, I would include questions such as the role of the constraints API =
(versus hacking SDP), what SDP lines/fields can or cannot be edited by =
applications and how SDP errors are handled. <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>While it =
would certainly be possible to write a -00 draft, without understanding =
the overall plan, getting into detail probably doesn't make much sense. =
&nbsp;IMHO, slides should be sufficient, and I have some available to =
stimulate discussion. <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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From juberti@google.com  Thu Oct 11 22:42:33 2012
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From: Justin Uberti <juberti@google.com>
Date: Thu, 11 Oct 2012 22:42:11 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda request: Overall Approach to defining SDP usage in WebRTC
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--0016e6d260886b35d104cbd627a3
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I am planning to address some of what you mention in the JSEP -02, and also
have some slides on this for the JSEP discussion.


On Thu, Oct 11, 2012 at 11:58 AM, Bernard Aboba
<bernard_aboba@hotmail.com>wrote:

> In addition to covering "fascinating" topics such as MTI codecs and JSEP
> support of O/A at IETF 85, I'd like to suggest that we also set some time
> aside at IETF 85 to discuss the overall approach to defining SDP usage in
> WebRTC.  ****
>
> ** **
>
> In this "big picture" discussion, I'd like to try to figure out how we
> will answer some of the following questions:****
>
> ** **
>
>    1. What SDP functionality MUST/SHOULD/MAY be implemented/used in
>    WebRTC API implementations?****
>    2. What is the overall model for SDP usage in JSEP?****
>    3. How much of this discussion belongs in W3C, how much in IETF? ****
>
> ** **
>
> The goal of this discussion is **not** to get into the details of the
> answer to the questions, but rather to focus on what work needs to be
> done,  what form it takes (adding material in existing docs, new docs,
> etc.), and what is handled in what org (W3C, IETF, etc.).****
>
> ** **
>
> To some extent, answers to question #1 are derivable from the RTP usage
> draft, but I'm not clear that it makes sense to cover all aspects of SDP
> usage there.  Also, it would be useful to discuss the plan for documenting
> SDP in current usage that has no corresponding RFC or I-D (e.g. are all the
> bases covered?). ****
>
> ** **
>
> In question #2, I would include questions such as the role of the
> constraints API (versus hacking SDP), what SDP lines/fields can or cannot
> be edited by applications and how SDP errors are handled. ****
>
> ** **
>
> While it would certainly be possible to write a -00 draft, without
> understanding the overall plan, getting into detail probably doesn't make
> much sense.  IMHO, slides should be sufficient, and I have some available
> to stimulate discussion. ****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--0016e6d260886b35d104cbd627a3
Content-Type: text/html; charset=UTF-8
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I am planning to address some of what you mention in the JSEP -02, and also=
 have some slides on this for the JSEP discussion.<div class=3D"gmail_extra=
"><br><br><div class=3D"gmail_quote">On Thu, Oct 11, 2012 at 11:58 AM, Bern=
ard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com=
" target=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><p class=3D"MsoNormal">In addition to covering &quot;fascinatin=
g&quot; topics such as MTI codecs and JSEP support of O/A at IETF 85, I&#39=
;d like to suggest that we also set some time aside at IETF 85 to discuss t=
he overall approach to defining SDP usage in WebRTC. =C2=A0<u></u><u></u></=
p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal">In th=
is &quot;big picture&quot; discussion, I&#39;d like to try to figure out ho=
w we will answer some of the following questions:<u></u><u></u></p><p class=
=3D"MsoNormal">

<u></u>=C2=A0<u></u></p><ol style=3D"margin-top:0in" start=3D"1" type=3D"1"=
><li class=3D"MsoNormal">What SDP functionality MUST/SHOULD/MAY be implemen=
ted/used in WebRTC API implementations?<u></u><u></u></li><li class=3D"MsoN=
ormal">What is the overall model for SDP usage in JSEP?<u></u><u></u></li>

<li class=3D"MsoNormal">How much of this discussion belongs in W3C, how muc=
h in IETF? <u></u><u></u></li></ol><p class=3D"MsoNormal"><u></u>=C2=A0<u><=
/u></p><p class=3D"MsoNormal">The goal of this discussion is *<b>not</b>* t=
o get into the details of the answer to the questions, but rather to focus =
on what work needs to be done,=C2=A0 what form it takes (adding material in=
 existing docs, new docs, etc.), and what is handled in what org (W3C, IETF=
, etc.).<u></u><u></u></p>

<p class=3D"MsoNormal" style=3D"margin-left:.5in"><u></u>=C2=A0<u></u></p><=
p class=3D"MsoNormal">To some extent, answers to question #1 are derivable =
from the RTP usage draft, but I&#39;m not clear that it makes sense to cove=
r all aspects of SDP usage there.=C2=A0 Also, it would be useful to discuss=
 the plan for documenting SDP in current usage that has no corresponding RF=
C or I-D (e.g. are all the bases covered?). <u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal">In qu=
estion #2, I would include questions such as the role of the constraints AP=
I (versus hacking SDP), what SDP lines/fields can or cannot be edited by ap=
plications and how SDP errors are handled. <u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal">While=
 it would certainly be possible to write a -00 draft, without understanding=
 the overall plan, getting into detail probably doesn&#39;t make much sense=
. =C2=A0IMHO, slides should be sufficient, and I have some available to sti=
mulate discussion. <u></u><u></u></p>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal"><u></=
u>=C2=A0<u></u></p><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=
=3D"MsoNormal"><u></u>=C2=A0<u></u></p><p class=3D"MsoNormal"><u></u>=C2=A0=
<u></u></p><p class=3D"MsoNormal">

<u></u>=C2=A0<u></u></p></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>

--0016e6d260886b35d104cbd627a3--

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From: Justin Uberti <juberti@google.com>
Date: Thu, 11 Oct 2012 22:46:25 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] JSEP-02: SDP line
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"SDP line" meant, literally, each text line in the SDP. A SDP blob is
composed of SDP lines.

If there is better terminology to use here, please advise and we will
change it.


On Mon, Oct 8, 2012 at 3:51 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Oct 4, 2012, at 7:33 , Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> > Hi,
> >
> > What is a =E2=80=9DSDP line=E2=80=9D? The same thing as a =E2=80=9CSDP =
blob=E2=80=9D?
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> this is poorly defined and we need to fix that - it's probably the wrong
> term
>
> I think it was trying to get at the m-line matching issues in SDP so the
> term
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b6d89c2933a9b04cbd636bb
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&quot;SDP line&quot; meant, literally, each text line in the SDP. A SDP blo=
b is composed of SDP lines.<div><br></div><div>If there is better terminolo=
gy to use here, please advise and we will change it.</div><div class=3D"gma=
il_extra">

<br><br><div class=3D"gmail_quote">On Mon, Oct 8, 2012 at 3:51 PM, Cullen J=
ennings (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" =
target=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">

<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Oct 4, 2012, at 7:33 , Christer Holmberg &lt;<a href=3D"mailto:christer.=
holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; What is a =E2=80=9DSDP line=E2=80=9D? The same thing as a =E2=80=9CSDP=
 blob=E2=80=9D?<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
</div></div>&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>
this is poorly defined and we need to fix that - it&#39;s probably the wron=
g term<br>
<br>
I think it was trying to get at the m-line matching issues in SDP so the te=
rm<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>

--047d7b6d89c2933a9b04cbd636bb--

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Date: Thu, 11 Oct 2012 22:59:44 -0700
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Subject: Re: [rtcweb] JSEP-02: Clone comments
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The preceding paragraph of 4.7.2 mentions that each clone may end up with
its own peer-reflexive candidates (as would be the case if the local NAT
was endpoint-dependent). So while the local addresses and
server-reflexive/relay addresses should be the same, the overall set of
local candidates for each clone may be different.

I will try to clarify this for the next draft.


On Thu, Oct 4, 2012 at 5:47 AM, Jim Barnett <Jim.Barnett@genesyslab.com>wro=
te:

> If the local address isn=E2=80=99t the same, we shouldn=E2=80=99t call it=
 a =E2=80=98clone=E2=80=99.  I
> think that people will expect everything to be the same on a =E2=80=98clo=
ne=E2=80=99.  ***
> *
>
> ** **
>
> **-          **Jim******
>
> ** **
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Christer Holmberg
> *Sent:* Thursday, October 04, 2012 7:57 AM
> *To:* Harald Alvestrand; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] JSEP-02: Clone comments****
>
> ** **
>
> Hi,****
>
>  ****
>
> >> A couple of comments on the cloning stuff in JSEP:****
>
> >>** **
>
> >> Q1. The document doesn=E2=80=99t say how the cloning is performed. If =
that is
> outside the scope of JSEP, and belongs to the W3C API spec, I think it
> should be mentioned.****
>
> >> ****
>
> >> Q2. The text in section 4.7.2 says:****
>
> **>> ****
>
> >>                =E2=80=9CAs a result of this cloning, the application w=
ill end
> up with N****
>
> >>                parallel sessions, each with a local and remote
> description and their****
>
> >>                own local and remote addresses.=E2=80=9D****
>
> =E2=89=AB****
>
> >> I think the =E2=80=9Cown local addresses=E2=80=9D wording is a little =
misleading, as
> each clone will share the same local address.****
>
> >** **
>
> > Not sure what to say here.****
>
> >** **
>
> > I can see multiple ways to implement this - some will end up with
> different local ports, some won't.****
>
>  ****
>
> Earlier in the same section, the text says:****
>
>  ****
>
>         =E2=80=9CSince the *clone uses the same local description as its*=
****
>
> *      &nb! sp;  parent*, creating a clone will fail if it is not
> possible to reserve****
>
>          the same resources for the clone as have already been reserved b=
y
> the****
>
>          parent.=E2=80=9D****
>
>  ****
>
> > If the non-local (reflexive?) candidates are allocated using STUN on a
> per-port basis, addresses could be different too.****
>
>  ****
>
> My assumption, and understanding of the text, is that a clone is a=E2=80=
=A6 eeeeh=E2=80=A6
> clone =E2=80=93 meaning that the local address is the same :)****
>
>  ****
>
> Regards,****
>
>  ****
>
> Christer****
>
>  ****
>
>  ****
>
>  ****
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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The preceding paragraph of 4.7.2 mentions that each clone may end up with i=
ts own peer-reflexive candidates (as would be the case if the local NAT was=
 endpoint-dependent). So while the local addresses and server-reflexive/rel=
ay addresses should be the same, the overall set of local candidates for ea=
ch clone may be different.<div>

<br></div><div>I will try to clarify this for the next draft.</div><div cla=
ss=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Oct 4, 2012 a=
t 5:47 AM, Jim Barnett <span dir=3D"ltr">&lt;<a href=3D"mailto:Jim.Barnett@=
genesyslab.com" target=3D"_blank">Jim.Barnett@genesyslab.com</a>&gt;</span>=
 wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">If the local =
address isn=E2=80=99t the same, we shouldn=E2=80=99t call it a =E2=80=98clo=
ne=E2=80=99.=C2=A0 I think that people will expect everything to be the sam=
e on a =E2=80=98clone=E2=80=99.=C2=A0 <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p><p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&q=
uot;,&quot;sans-serif&quot;;color:#1f497d"><span>-<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=C2=A0 </span></span></span><u></u><span style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Jim<u></u>=
<u></u></span><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p><div><div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding=
:3.0pt 0in 0in 0in">

<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href=
=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank"=
>rtcweb-bounces@ietf.org</a>] <b>On Behalf Of </b>Christer Holmberg<br>

<b>Sent:</b> Thursday, October 04, 2012 7:57 AM<br><b>To:</b> Harald Alvest=
rand; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org<=
/a><br><b>Subject:</b> Re: [rtcweb] JSEP-02: Clone comments<u></u><u></u></=
span></p>

</div></div><div class=3D"im"><p class=3D"MsoNormal"><u></u>=C2=A0<u></u></=
p><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Hi,</span><span style=
=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span>=C2=A0</span><span><u></u><u></u></=
span></p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;=
font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt; A couple o=
f comments on the cloning stuff in JSEP:<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt;<u></u>=C2=A0<u></u>=
</span></p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt; Q1. The =
document doesn=E2=80=99t say how the cloning is performed. If that is outsi=
de the scope of JSEP, and belongs to the W3C API spec, I think it should be=
 mentioned.<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt;=C2=A0<u></u><u></u>=
</span></p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt; Q2. The =
text in section 4.7.2 says:<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><u></u>&gt;&gt;=C2=A0<u></u><u></u></p></=
div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family=
:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt;=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =E2=
=80=9CAs a result of this cloning, the application will end up with N<u></u=
><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt;=C2=A0=C2=A0 =C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 parallel=
 sessions, each with a local and remote description and their<u></u><u></u>=
</span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;&gt;=C2=A0=C2=A0 =C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 own loca=
l and remote addresses.=E2=80=9D<u></u><u></u></span></p></div><div><p clas=
s=3D"MsoNormal">

<span>=E2=89=AB</span><span style=3D"font-size:10.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;"><u></u><u></u></span></p></div><div><p c=
lass=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Calibr=
i&quot;,&quot;sans-serif&quot;">&gt;&gt; I think the =E2=80=9Cown local add=
resses=E2=80=9D wording is a little misleading, as each clone will share th=
e same local address.<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;<u></u>=C2=A0<u></u></sp=
an></p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt; Not sure what to=
 say here.<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt;<u></u>=C2=A0<u></u></sp=
an></p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&gt; I can see multip=
le ways to implement this - some will end up with different local ports, so=
me won&#39;t.<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span>=C2=A0<u></u><u></u></span></p></di=
v><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Earlier in the same se=
ction, the text says:</span><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0</span><span=
 style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0 =E2=80=9CSince the <b>clone uses the same local de=
scription as its</b></span><span style=3D"font-size:10.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;"><u></u><u></u></span></p>

</div></div><div><p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;=
font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0&amp;nb!
 sp;=C2=A0 parent</span></b><span>, creating a clone will fail if it is not=
 possible to reserve</span><span style=3D"font-size:10.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;"><u></u><u></u></span></p></div><div=
 class=3D"im">

<div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0 the same resources for the clone as have already b=
een reserved by the</span><span style=3D"font-size:10.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 parent.=E2=80=9D</span><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><u></u><=
u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">=C2=A0<u></u><u></u></span><=
/p></div><div><p class=3D"MsoNormal"><span>&gt; If the non-local (reflexive=
?) candidates are allocated using STUN on a per-port basis, addresses could=
 be different too.<u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">=C2=A0<u></u><u></u></span><=
/p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-f=
amily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">My assumption, =
and understanding of the text, is that a clone is a=E2=80=A6 eeeeh=E2=80=A6=
 clone =E2=80=93 meaning that the local address is the same :)</span><span =
style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&=
quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0</span><span=
 style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif=
&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Regards,</span><sp=
an><u></u><u></u></span></p></div><div><p><span style=3D"font-size:10.0pt;f=
ont-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">=C2=A0</sp=
an><span style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:red">Christer</span><sp=
an style=3D"font-size:10.0pt;font-family:&quot;Calibri&quot;,&quot;sans-ser=
if&quot;"><u></u><u></u></span></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">=C2=A0<u></u><u></u></span><=
/p></div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-f=
amily:&quot;Calibri&quot;,&quot;sans-serif&quot;">=C2=A0<u></u><u></u></spa=
n></p>

</div><div><p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-fami=
ly:&quot;Calibri&quot;,&quot;sans-serif&quot;">=C2=A0<u></u><u></u></span><=
/p></div></div><p></p></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>

--0016369c8d402db72e04cbd66684--

From harald@alvestrand.no  Fri Oct 12 02:36:55 2012
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Subject: [rtcweb] Fwd: WG poll for consensus MSID draft
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Reminder: This poll in MMUSIC closes on Sunday the 14th, if I read the 
announcement correctly; if you want to make sure your voice is heard, 
speak up today.


-------- Original Message --------
Subject: 	WG poll for consensus MSID draft
Date: 	Tue, 9 Oct 2012 15:24:04 +0200
From: 	Miguel A. Garcia <Miguel.A.Garcia@ericsson.com>
To: 	mmusic <mmusic@ietf.org>
CC: 	Flemming Andreasen <fandreas@cisco.com>, Harald Alvestrand 
<harald@alvestrand.no>



At the last IETF meeting we had a presentation of the following draft:

http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/

At that time, there was interest in solving the problem of cross session
stream identification.

We would like to poll the working group for consensus on adopting the
above mentioned draft as a working group item, once we get an approved
milestone.

If you support this work or have problems with the adoption of this
document as WG item, please indicate it by replying to this e-mail within
the next 5 days.

Flemming and Miguel (co-chairs)

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain





--------------010909010500080404050805
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Reminder: This poll in MMUSIC closes on Sunday the 14th, if I read
    the announcement correctly; if you want to make sure your voice is
    heard, speak up today.<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" cellpadding="0"
        cellspacing="0" border="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>WG poll for consensus MSID draft</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 9 Oct 2012 15:24:04 +0200</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>Miguel A. Garcia <a class="moz-txt-link-rfc2396E" href="mailto:Miguel.A.Garcia@ericsson.com">&lt;Miguel.A.Garcia@ericsson.com&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>mmusic <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td>Flemming Andreasen <a class="moz-txt-link-rfc2396E" href="mailto:fandreas@cisco.com">&lt;fandreas@cisco.com&gt;</a>, Harald
              Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>At the last IETF meeting we had a presentation of the following draft:

<a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/">http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid/</a>

At that time, there was interest in solving the problem of cross session 
stream identification.

We would like to poll the working group for consensus on adopting the 
above mentioned draft as a working group item, once we get an approved 
milestone.

If you support this work or have problems with the adoption of this 
document as WG item, please indicate it by replying to this e-mail within 
the next 5 days.

Flemming and Miguel (co-chairs)

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain
</pre>
      <br>
      <br>
    </div>
    <br>
  </body>
</html>

--------------010909010500080404050805--

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On Oct 9, 2012, at 4:58 PM, Martin Thomson <martin.thomson@gmail.com> wrote=
:

> Quit beating around the bush: it's me you are talking about.

Sorry Martin - I did not mean exactly you. I meant, you, me, Christer, othe=
rs. I think we just need to get all very pragmatic and specific about what =
the code needs to do. And I don't mean to imply that you or Christer are no=
t - you both are very pragmatic. But we all need to get this thread to spec=
ific examples about real world use cases that people can understand. =20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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On Oct 9, 2012, at 4:58 PM, Martin Thomson <martin.thomson@gmail.com> wrote=
:

> On 9 October 2012 13:05, Cullen Jennings (fluffy) <fluffy@cisco.com> wrot=
e:
>> 3264, SDP, 3261, and related documents are dealing with a bunch of thing=
s including what happens at media plane and signaling plane.
>>=20
>> I'll note that though O/A is per dialog, there is only one O shared acro=
ss multiple legs and when you create the O you don't know how many dialogs =
there will be. So from the media point of view (covered more in SDP spec th=
an 3264) there is one O with a bunch of A. From signaling point of view the=
re are a bunch of O/A pairs).
>>=20
>> The dialog is SIP signaling concept not a media plane level concept. We =
moved the signaling part out of the browser and into the JS. But the media =
part is still in the browser. So as 3264 says, after the offer is construct=
ed, we have to be willing to receive media for all the codec type in the of=
fer. When we get the answer 3264 makes it clear that one MAY stop receiving=
 the codecs that were in the offer but not selected in the answer. However,=
 this can not be done until the signaling layer is sure that no more offers=
 will be honored. Since that signalling part of Offer/Answer is in the JS, =
the API need to to have a way to signal what the MAY part should do and tha=
t is the PRANSWER vs ANSWER.
>=20
> That's a consistent and logical view.  It is not what RFC 3264 says.
> It may even be the only reasonable solution.
>=20
>> People keep trying to make this some complex weird argument invoking the=
 SIP deities of the past and quoting incomprehensible phrases from various =
RFC caved in stone [...]
>=20
> Quit beating around the bush: it's me you are talking about.
>=20
> It's not a complex weird argument, it's this:
>  PRANSWER is not described in RFC 3264.
>=20
> draft-*-jsep might say something about it, but I'd have to guess about
> what to do if I were to implement the feature and that doesn't help
> with interoperation.
>=20
> If there was text that described the above, plus the implications in
> some detail (including what resources can be released as they relate
> to the SDP), then we're good.  I don't even think that writing this is
> especially hard.=20

Agree that text has to be a clear. But provision is a term from 3261 which =
is referenced from 3264 and bunch of this also relies on SDP RFC too. Some =
of the key information about it is 3262. This is all intertwined but my key=
 point is that I don't think we are trying to do anything that was not part=
 of 3264 offer / answer. This is more or less a SIP 180.=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
Thread-Index: AQHNqJNERbmAruLOa06QaNt9Wva3zw==
Date: Fri, 12 Oct 2012 16:04:31 +0000
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References: <506B0367.4000103@ericsson.com> <C5E08FE080ACFD4DAE31E4BDBF944EB111867718@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD03A6@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB11187F8F1@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD087E@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1118810C3@xmb-aln-x02.cisco.com>, <CABkgnnV8YbbPi3pZt9mBjBfSgkx4EgL=EY0YogpeRobUMQ506w@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAFAA6B@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1118819FB@xmb-aln-x02.cisco.com> <000b01cda720$bd5097a0$37f1c6e0$@co.in>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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I fully agree this needs to work early media to PSTN via SIP. I don't see a=
ny problem with doing that. I'm happy to work through detail call flows to =
make sure that early media to PSTN works.=20

Update after a 180 is not allowed by the Update RFC as I pointed out in oth=
er email exactly because just in SIP with no RTCP web, the UA that caret th=
e invite would not have enough information to be able to process this as yo=
u point in the thread below.=20


On Oct 10, 2012, at 1:51 PM, Parthasarathi R <partha@parthasarathi.co.in> w=
rote:

> Cullen,
>=20
> As signaling is moved to JS (application layer), Offer-answer in JSEP has=
 to
> be generic enough. The underlying assumption in PRANSWER state is that
> webserver knows whether the received SDP from remote is PRANSWER or final
> ANSWER. Unfortunately, Webserver may not able to tell whether the receive=
 in
> lot of real time deployment which is an open issue now. UPDATE offer afte=
r
> 18x answer from remote is the typical example where PRANSWER state breaks=
.
> The real-time usage of SIP UPDATE offer is to update the existing answer =
in
> 18x in the same dialog. Early dialog UPDATE callflow is well deployment
> because of PSTN remote ringback (18x) from media server and then UPDATE w=
ith
> SDP to update the media information of remote endpoing and then, call
> connect (200 ok) from actual endpoint. The same PSTN callflow shall be
> achieved by serial forking as well. Please note that the final answer or =
not
> is not based on browser or originating SIP UA but it is based on the
> intermediate SIP entities.=20
>=20
> From RTCWeb perspective, The new OFFER is possible to be received in brow=
ser
> as per RFC 3264 after the first answer is received. PRANSWER is the new
> state in JSEP wherein JSEP is the extension of RFC 3264. It will be good =
in
> case you explain how browser in PRANSWER has to handle new OFFER from the
> remote side.=20
>=20
> Please include me in case any phone discussion if you are planning.
>=20
> Thanks
> Partha
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Cullen Jennings (fluffy)
> Sent: Wednesday, October 10, 2012 1:35 AM
> To: Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
>=20
>=20
> On Oct 9, 2012, at 12:04 , Christer Holmberg
> <christer.holmberg@ericsson.com>
> wrote:
>=20
>> Hi,
>>=20
>>>> I have not seen any reason to relax 3264 yet but if something comes up=
,
> agree we should carefully look at the cases. I think we can just do strai=
ght
> up 3264.
>>>=20
>>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>>>=20
>>> The offerer MAY immediately cease listening for media formats that
>>> were listed in the initial offer, but not present in the answer.
>>>=20
>>> "the" answer.
>>=20
>> I agree with Martin. 3264 O/A is always per dialog, and forking is
> supported by generating multiple dialogs. JSEP, OTOH, in order to support
> forking with a single "dialog" (peerConnection local descriptor), now
> defines O/A as offer+any number of pranswers + answer.
>>=20
>> So, we would e.g. have to define what happens if a new offer is received
> from the remote side while the browser is in pranswer-received state (see=
 my
> call flow in another reply).
>=20
>=20
> 3264, SDP, 3261, and related documents are dealing with a bunch of things
> including what happens at media plane and signaling plane.=20
>=20
> I'll note that though O/A is per dialog, there is only one O shared acros=
s
> multiple legs and when you create the O you don't know how many dialogs
> there will be. So from the media point of view (covered more in SDP spec
> than 3264) there is one O with a bunch of A. From signaling point of view
> there are a bunch of O/A pairs).=20
>=20
> The dialog is SIP signaling concept not a media plane level concept. We
> moved the signaling part out of the browser and into the JS. But the medi=
a
> part is still in the browser. So as 3264 says, after the offer is
> constructed, we have to be willing to receive media for all the codec typ=
e
> in the offer. When we get the answer 3264 makes it clear that one MAY sto=
p
> receiving the codecs that were in the offer but not selected in the answe=
r.
> However, this can not be done until the signaling layer is sure that no m=
ore
> offers will be honored. Since that signalling part of Offer/Answer is in =
the
> JS, the API need to to have a way to signal what the MAY part should do a=
nd
> that is the PRANSWER vs ANSWER.=20
>=20
> People keep trying to make this some complex weird argument invoking the =
SIP
> deities of the past and quoting incomprehensible phrases from various RFC
> caved in stone but the bottom line in the code is very simple to understa=
nd.
> When creating the offer you alloced some resources ( like a port to recei=
ve
> video on ). When you get an answer that does not use that resource, you n=
eed
> to tell the media stack if it should free the resources or not. O/A has
> situation where you need to keep the resources available (like there are
> more dialogs coming) and situation where you need to free the resource.
> Since we split the signalling out of the browser and left the media in th=
e
> browser, we need be able to allow the JS that is dealing with signaling t=
o
> tell the browser when to free the resource.=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From christer.holmberg@ericsson.com  Fri Oct 12 10:42:56 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Martin Thomson <martin.thomson@gmail.com>
Date: Fri, 12 Oct 2012 19:41:29 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Cullen,=20

I did send a real world call flow example, which you haven't commented :)

Regards,

Christer

________________________________________
From: Cullen Jennings (fluffy) [fluffy@cisco.com]
Sent: Friday, October 12, 2012 6:52 PM
To: Martin Thomson
Cc: Christer Holmberg; rtcweb@ietf.org
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting

On Oct 9, 2012, at 4:58 PM, Martin Thomson <martin.thomson@gmail.com> wrote=
:

> Quit beating around the bush: it's me you are talking about.

Sorry Martin - I did not mean exactly you. I meant, you, me, Christer, othe=
rs. I think we just need to get all very pragmatic and specific about what =
the code needs to do. And I don't mean to imply that you or Christer are no=
t - you both are very pragmatic. But we all need to get this thread to spec=
ific examples about real world use cases that people can understand.=

From christer.holmberg@ericsson.com  Fri Oct 12 10:45:20 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Parthasarathi R <partha@parthasarathi.co.in>
Date: Fri, 12 Oct 2012 19:43:24 +0200
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Hi,

> I fully agree this needs to work early media to PSTN via SIP. I don't see=
 any problem with doing that. I'm happy to work through detail call flows t=
o make sure that early media to PSTN works.
>
> Update after a 180 is not allowed by the Update RFC as I pointed out in o=
ther email exactly because just in SIP with no RTCP web, the UA that caret =
the invite would not have enough information to be able to process this as =
you point in the thread below.

As far as I know, UPDATE is allowed after a reliable 180. If you don't agre=
e, please show me the text which forbids it :)

Regards,

Christer



On Oct 10, 2012, at 1:51 PM, Parthasarathi R <partha@parthasarathi.co.in> w=
rote:

> Cullen,
>
> As signaling is moved to JS (application layer), Offer-answer in JSEP has=
 to
> be generic enough. The underlying assumption in PRANSWER state is that
> webserver knows whether the received SDP from remote is PRANSWER or final
> ANSWER. Unfortunately, Webserver may not able to tell whether the receive=
 in
> lot of real time deployment which is an open issue now. UPDATE offer afte=
r
> 18x answer from remote is the typical example where PRANSWER state breaks=
.
> The real-time usage of SIP UPDATE offer is to update the existing answer =
in
> 18x in the same dialog. Early dialog UPDATE callflow is well deployment
> because of PSTN remote ringback (18x) from media server and then UPDATE w=
ith
> SDP to update the media information of remote endpoing and then, call
> connect (200 ok) from actual endpoint. The same PSTN callflow shall be
> achieved by serial forking as well. Please note that the final answer or =
not
> is not based on browser or originating SIP UA but it is based on the
> intermediate SIP entities.
>
> From RTCWeb perspective, The new OFFER is possible to be received in brow=
ser
> as per RFC 3264 after the first answer is received. PRANSWER is the new
> state in JSEP wherein JSEP is the extension of RFC 3264. It will be good =
in
> case you explain how browser in PRANSWER has to handle new OFFER from the
> remote side.
>
> Please include me in case any phone discussion if you are planning.
>
> Thanks
> Partha
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Cullen Jennings (fluffy)
> Sent: Wednesday, October 10, 2012 1:35 AM
> To: Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
>
>
> On Oct 9, 2012, at 12:04 , Christer Holmberg
> <christer.holmberg@ericsson.com>
> wrote:
>
>> Hi,
>>
>>>> I have not seen any reason to relax 3264 yet but if something comes up=
,
> agree we should carefully look at the cases. I think we can just do strai=
ght
> up 3264.
>>>
>>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>>>
>>> The offerer MAY immediately cease listening for media formats that
>>> were listed in the initial offer, but not present in the answer.
>>>
>>> "the" answer.
>>
>> I agree with Martin. 3264 O/A is always per dialog, and forking is
> supported by generating multiple dialogs. JSEP, OTOH, in order to support
> forking with a single "dialog" (peerConnection local descriptor), now
> defines O/A as offer+any number of pranswers + answer.
>>
>> So, we would e.g. have to define what happens if a new offer is received
> from the remote side while the browser is in pranswer-received state (see=
 my
> call flow in another reply).
>
>
> 3264, SDP, 3261, and related documents are dealing with a bunch of things
> including what happens at media plane and signaling plane.
>
> I'll note that though O/A is per dialog, there is only one O shared acros=
s
> multiple legs and when you create the O you don't know how many dialogs
> there will be. So from the media point of view (covered more in SDP spec
> than 3264) there is one O with a bunch of A. From signaling point of view
> there are a bunch of O/A pairs).
>
> The dialog is SIP signaling concept not a media plane level concept. We
> moved the signaling part out of the browser and into the JS. But the medi=
a
> part is still in the browser. So as 3264 says, after the offer is
> constructed, we have to be willing to receive media for all the codec typ=
e
> in the offer. When we get the answer 3264 makes it clear that one MAY sto=
p
> receiving the codecs that were in the offer but not selected in the answe=
r.
> However, this can not be done until the signaling layer is sure that no m=
ore
> offers will be honored. Since that signalling part of Offer/Answer is in =
the
> JS, the API need to to have a way to signal what the MAY part should do a=
nd
> that is the PRANSWER vs ANSWER.
>
> People keep trying to make this some complex weird argument invoking the =
SIP
> deities of the past and quoting incomprehensible phrases from various RFC
> caved in stone but the bottom line in the code is very simple to understa=
nd.
> When creating the offer you alloced some resources ( like a port to recei=
ve
> video on ). When you get an answer that does not use that resource, you n=
eed
> to tell the media stack if it should free the resources or not. O/A has
> situation where you need to keep the resources available (like there are
> more dialogs coming) and situation where you need to free the resource.
> Since we split the signalling out of the browser and left the media in th=
e
> browser, we need be able to allow the JS that is dealing with signaling t=
o
> tell the browser when to free the resource.
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=

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Date: Fri, 12 Oct 2012 14:21:14 -0500
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Parthasarathi R <partha@parthasarathi.co.in>
From: James Polk <jmpolk@cisco.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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At 12:43 PM 10/12/2012, Christer Holmberg wrote:
>Hi,
>
> > I fully agree this needs to work early media to PSTN via SIP. I 
> don't see any problem with doing that. I'm happy to work through 
> detail call flows to make sure that early media to PSTN works.
> >
> > Update after a 180 is not allowed by the Update RFC as I pointed 
> out in other email exactly because just in SIP with no RTCP web, 
> the UA that caret the invite would not have enough information to 
> be able to process this as you point in the thread below.
>
>As far as I know, UPDATE is allowed after a reliable 180. If you 
>don't agree, please show me the text which forbids it :)

UPDATE is allowed after a before and after a 180, and allowed after a 
200 OK to the INVITE. RFC 6442 (Geolocation Header) took advantage of 
this function through/with/by/whatever the blessing of Jonathan R.

James


>Regards,
>
>Christer
>
>
>
>On Oct 10, 2012, at 1:51 PM, Parthasarathi R 
><partha@parthasarathi.co.in> wrote:
>
> > Cullen,
> >
> > As signaling is moved to JS (application layer), Offer-answer in 
> JSEP has to
> > be generic enough. The underlying assumption in PRANSWER state is that
> > webserver knows whether the received SDP from remote is PRANSWER or final
> > ANSWER. Unfortunately, Webserver may not able to tell whether the 
> receive in
> > lot of real time deployment which is an open issue now. UPDATE offer after
> > 18x answer from remote is the typical example where PRANSWER state breaks.
> > The real-time usage of SIP UPDATE offer is to update the existing answer in
> > 18x in the same dialog. Early dialog UPDATE callflow is well deployment
> > because of PSTN remote ringback (18x) from media server and then 
> UPDATE with
> > SDP to update the media information of remote endpoing and then, call
> > connect (200 ok) from actual endpoint. The same PSTN callflow shall be
> > achieved by serial forking as well. Please note that the final 
> answer or not
> > is not based on browser or originating SIP UA but it is based on the
> > intermediate SIP entities.
> >
> > From RTCWeb perspective, The new OFFER is possible to be received 
> in browser
> > as per RFC 3264 after the first answer is received. PRANSWER is the new
> > state in JSEP wherein JSEP is the extension of RFC 3264. It will be good in
> > case you explain how browser in PRANSWER has to handle new OFFER from the
> > remote side.
> >
> > Please include me in case any phone discussion if you are planning.
> >
> > Thanks
> > Partha
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> > Cullen Jennings (fluffy)
> > Sent: Wednesday, October 10, 2012 1:35 AM
> > To: Christer Holmberg
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
> >
> >
> > On Oct 9, 2012, at 12:04 , Christer Holmberg
> > <christer.holmberg@ericsson.com>
> > wrote:
> >
> >> Hi,
> >>
> >>>> I have not seen any reason to relax 3264 yet but if something comes up,
> > agree we should carefully look at the cases. I think we can just 
> do straight
> > up 3264.
> >>>
> >>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
> >>>
> >>> The offerer MAY immediately cease listening for media formats that
> >>> were listed in the initial offer, but not present in the answer.
> >>>
> >>> "the" answer.
> >>
> >> I agree with Martin. 3264 O/A is always per dialog, and forking is
> > supported by generating multiple dialogs. JSEP, OTOH, in order to support
> > forking with a single "dialog" (peerConnection local descriptor), now
> > defines O/A as offer+any number of pranswers + answer.
> >>
> >> So, we would e.g. have to define what happens if a new offer is received
> > from the remote side while the browser is in pranswer-received 
> state (see my
> > call flow in another reply).
> >
> >
> > 3264, SDP, 3261, and related documents are dealing with a bunch of things
> > including what happens at media plane and signaling plane.
> >
> > I'll note that though O/A is per dialog, there is only one O shared across
> > multiple legs and when you create the O you don't know how many dialogs
> > there will be. So from the media point of view (covered more in SDP spec
> > than 3264) there is one O with a bunch of A. From signaling point of view
> > there are a bunch of O/A pairs).
> >
> > The dialog is SIP signaling concept not a media plane level concept. We
> > moved the signaling part out of the browser and into the JS. But the media
> > part is still in the browser. So as 3264 says, after the offer is
> > constructed, we have to be willing to receive media for all the codec type
> > in the offer. When we get the answer 3264 makes it clear that one MAY stop
> > receiving the codecs that were in the offer but not selected in the answer.
> > However, this can not be done until the signaling layer is sure 
> that no more
> > offers will be honored. Since that signalling part of 
> Offer/Answer is in the
> > JS, the API need to to have a way to signal what the MAY part should do and
> > that is the PRANSWER vs ANSWER.
> >
> > People keep trying to make this some complex weird argument 
> invoking the SIP
> > deities of the past and quoting incomprehensible phrases from various RFC
> > caved in stone but the bottom line in the code is very simple to 
> understand.
> > When creating the offer you alloced some resources ( like a port to receive
> > video on ). When you get an answer that does not use that 
> resource, you need
> > to tell the media stack if it should free the resources or not. O/A has
> > situation where you need to keep the resources available (like there are
> > more dialogs coming) and situation where you need to free the resource.
> > Since we split the signalling out of the browser and left the media in the
> > browser, we need be able to allow the JS that is dealing with signaling to
> > tell the browser when to free the resource.
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > 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 fluffy@cisco.com  Fri Oct 12 19:21:48 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] Agenda requests for Atlanta meeting
Thread-Index: AQHNqOl+kDeDfA5eqEW9OAbsBiOagg==
Date: Sat, 13 Oct 2012 02:21:46 +0000
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On Oct 12, 2012, at 11:43 AM, Christer Holmberg <christer.holmberg@ericsson=
.com> wrote:

> Hi,
>=20
>> I fully agree this needs to work early media to PSTN via SIP. I don't se=
e any problem with doing that. I'm happy to work through detail call flows =
to make sure that early media to PSTN works.
>>=20
>> Update after a 180 is not allowed by the Update RFC as I pointed out in =
other email exactly because just in SIP with no RTCP web, the UA that caret=
 the invite would not have enough information to be able to process this as=
 you point in the thread below.
>=20
> As far as I know, UPDATE is allowed after a reliable 180. If you don't ag=
ree, please show me the text which forbids it :)
>=20
> Regards,

I agree - but it's not ok after a normal 180


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] PRANSWER is unusable for serial forking
Thread-Index: AQHNqOnc3QfmtiokskqSBC02DG7ABw==
Date: Sat, 13 Oct 2012 02:24:25 +0000
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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On Oct 9, 2012, at 12:40 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
>=20
> I THINK Roman is talking about the following flow:
>=20
>=20
> BROWSER		JS APP			FORKING PROXY			SIP UA 1			SIP UA 2
>=20
> ------- OFFER --------------->
>                                                  ----- INVITE (OFFER 1) -=
-------->
> 							----- INVITE (OFFER 1) --------->
> 							----- INVITE (OFFER 1) -------------------------------------------=
--------->
> 							<----180 (ANSWER 1a) ----------
> 			<----180 (ANSWER 1a) ----------
> <----PRANSWER -----------
> 							<----UPDATE (OFFER 2) --------
> 			<----UPDATE (OFFER 2) ---------
> <----???? --------------------
> 							<----200 (ANSWER 1b) ---------------------------------------------=
---------
> 			<----200 (ANSWER 1b) ----------
> <----ANSWER ---------------
>=20
>=20

That's not a call flow that is legal for use of update. From section 5.1 of=
 of RFC 3311 has=20

     o  If the UPDATE is being sent before completion of the initial
         INVITE transaction, and the initial INVITE contained an offer,
         the UPDATE can contain an offer if the callee generated an
         answer in a reliable provisional response, and the caller has
         received answers to any other offers it sent in either PRACK or
         UPDATE, and has generated answers for any offers it received in
         an UPDATE from the callee.

If the answer from UA1 is going to do updates before the invite transaction=
 complete, it needs to do that with 100rel / PRACK. Note the diagram should=
 also show an the 200 response to the update is going to cary an answer.

So if you made all these changes, lets talk about how you would solve this =
problem in the JS App.=20

Fundamentally what is happening here is the call signaling is setting up tw=
o session - this is parallel media forking and parallel signaling forking. =
One fork to UA1, and one to UA2. The JS App needs to pick which one it is g=
oing to choose to have a media session with. Let's say it choose UA1, then =
it would map the Answer 1a to an Answer, the Offer2 to an offer. If it was =
going to choose UA2, it would not pass the answer down to the browser at al=
l and would instead wait till it Answer 1b and pass that down. If you think=
 it should simultaneously have a media session with both UA1 and UA2, then =
things are more complicated. I realize that at the time it gets ANSWER 1a, =
it does not know it will later get ANSWR 1b so that make it hard to choose =
UA2.=20

Parallel forking with early media and secure media is a hard problem in SIP=
 as we know from the HERFP discussion. I'm unaware of real world problem th=
at are solved well with SIP and require parallel forking and thus my focus =
has been on making sure we solve serial forking. I'm perfectly happy if we =
*also* solve parallel forking but I don't care too much as long a early med=
ia serial forking scenarios work because theses do happen all the time. I'm=
 pretty skeptical that are a lot of existing SIP phones that deal well with=
 simultaneously receiving and sending media to two other UAs in a scenario =
like this (I'm trying to separate this from conferencing - I know lots of S=
IP UA do 3 way calling and conferencing just fine).=20

If you want to do both, one way might be to create a new peer connection fo=
r the 2nd call, and map it to and IVNTE with replaces, or send a SIP refer =
to get the second session mapped to the new PeerConnection object. One coul=
d also try and work on some thing that allows one PeerConenction to clone t=
o two places but that's going to be a bunch of work to figure out - and aga=
in, perfectly happy if someone goes and does that but it's a hard problem w=
hen you have resource allocations.=20






From christer.holmberg@ericsson.com  Sat Oct 13 00:22:46 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Sat, 13 Oct 2012 09:21:30 +0200
Thread-Topic: [rtcweb] PRANSWER is unusable for serial forking
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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Hi Cullen,

I'll read your reply more in detail later, but please note that I assume th=
at the 180 is sent *reliably* (I left the PRACK out).

Regards,

Christer

________________________________________
From: Cullen Jennings (fluffy) [fluffy@cisco.com]
Sent: Saturday, October 13, 2012 5:24 AM
To: Christer Holmberg
Cc: Roman Shpount; Cullen Jennings; rtcweb@ietf.org
Subject: Re: [rtcweb] PRANSWER is unusable for serial forking

On Oct 9, 2012, at 12:40 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
>
> I THINK Roman is talking about the following flow:
>
>
> BROWSER               JS APP                  FORKING PROXY              =
     SIP UA 1                        SIP UA 2
>
> ------- OFFER --------------->
>                                                  ----- INVITE (OFFER 1) -=
-------->
>                                                       ----- INVITE (OFFER=
 1) --------->
>                                                       ----- INVITE (OFFER=
 1) ---------------------------------------------------->
>                                                       <----180 (ANSWER 1a=
) ----------
>                       <----180 (ANSWER 1a) ----------
> <----PRANSWER -----------
>                                                       <----UPDATE (OFFER =
2) --------
>                       <----UPDATE (OFFER 2) ---------
> <----???? --------------------
>                                                       <----200 (ANSWER 1b=
) ------------------------------------------------------
>                       <----200 (ANSWER 1b) ----------
> <----ANSWER ---------------
>
>

That's not a call flow that is legal for use of update. From section 5.1 of=
 of RFC 3311 has

     o  If the UPDATE is being sent before completion of the initial
         INVITE transaction, and the initial INVITE contained an offer,
         the UPDATE can contain an offer if the callee generated an
         answer in a reliable provisional response, and the caller has
         received answers to any other offers it sent in either PRACK or
         UPDATE, and has generated answers for any offers it received in
         an UPDATE from the callee.

If the answer from UA1 is going to do updates before the invite transaction=
 complete, it needs to do that with 100rel / PRACK. Note the diagram should=
 also show an the 200 response to the update is going to cary an answer.

So if you made all these changes, lets talk about how you would solve this =
problem in the JS App.

Fundamentally what is happening here is the call signaling is setting up tw=
o session - this is parallel media forking and parallel signaling forking. =
One fork to UA1, and one to UA2. The JS App needs to pick which one it is g=
oing to choose to have a media session with. Let's say it choose UA1, then =
it would map the Answer 1a to an Answer, the Offer2 to an offer. If it was =
going to choose UA2, it would not pass the answer down to the browser at al=
l and would instead wait till it Answer 1b and pass that down. If you think=
 it should simultaneously have a media session with both UA1 and UA2, then =
things are more complicated. I realize that at the time it gets ANSWER 1a, =
it does not know it will later get ANSWR 1b so that make it hard to choose =
UA2.

Parallel forking with early media and secure media is a hard problem in SIP=
 as we know from the HERFP discussion. I'm unaware of real world problem th=
at are solved well with SIP and require parallel forking and thus my focus =
has been on making sure we solve serial forking. I'm perfectly happy if we =
*also* solve parallel forking but I don't care too much as long a early med=
ia serial forking scenarios work because theses do happen all the time. I'm=
 pretty skeptical that are a lot of existing SIP phones that deal well with=
 simultaneously receiving and sending media to two other UAs in a scenario =
like this (I'm trying to separate this from conferencing - I know lots of S=
IP UA do 3 way calling and conferencing just fine).

If you want to do both, one way might be to create a new peer connection fo=
r the 2nd call, and map it to and IVNTE with replaces, or send a SIP refer =
to get the second session mapped to the new PeerConnection object. One coul=
d also try and work on some thing that allows one PeerConenction to clone t=
o two places but that's going to be a bunch of work to figure out - and aga=
in, perfectly happy if someone goes and does that but it's a hard problem w=
hen you have resource allocations.=

From christer.holmberg@ericsson.com  Sat Oct 13 02:51:07 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Date: Sat, 13 Oct 2012 11:51:00 +0200
Thread-Topic: [rtcweb] PRANSWER is unusable for serial forking
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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Hi Cullen,

>> I THINK Roman is talking about the following flow:
>>
>>
>> BROWSER               JS APP                  FORKING PROXY             =
      SIP UA 1                        SIP UA 2
>>
>> ------- OFFER --------------->
>>                                                  ----- INVITE (OFFER 1) =
--------->
>>                                                       ----- INVITE (OFFE=
R 1) --------->
>>                                                       ----- INVITE (OFFE=
R 1) ---------------------------------------------------->
>>                                                       <----180 (ANSWER 1=
a) ----------
>>                       <----180 (ANSWER 1a) ----------
>> <----PRANSWER -----------
>>                                                       <----UPDATE (OFFER=
 2) --------
>>                       <----UPDATE (OFFER 2) ---------
>> <----???? --------------------
>>                                                       <----200 (ANSWER 1=
b) ------------------------------------------------------
>>                       <----200 (ANSWER 1b) ----------
>> <----ANSWER ---------------
>>
>>
>
> That's not a call flow that is legal for use of update. From section 5.1 =
of of RFC 3311 has
>
>     o  If the UPDATE is being sent before completion of the initial
>         INVITE transaction, and the initial INVITE contained an offer,
>         the UPDATE can contain an offer if the callee generated an
>        answer in a reliable provisional response, and the caller has
>         received answers to any other offers it sent in either PRACK or
>         UPDATE, and has generated answers for any offers it received in
>         an UPDATE from the callee.
>
> If the answer from UA1 is going to do updates before the invite transacti=
on complete, it needs to do that with 100rel / PRACK. Note the diagram shou=
ld also show an the 200 response to the update is going to cary an answer.

I agree. I appologize for making too many assumptions. The 180 IS sent reli=
ably, there IS a PRACK, and there IS a response to the UPDATE.

> So if you made all these changes, lets talk about how you would solve thi=
s problem in the JS App.
>
> Fundamentally what is happening here is the call signaling is setting up =
two session - this is parallel media forking and parallel signaling forking=
.=20

We could even make it more simply by removing the forking part, ie there is=
 a reliable 180, follwed by an UPDATE. But, for now, let's continue with th=
e flow above...

(I have added some numbering to your text below)

>1. One fork to UA1, and one to UA2.

Yes.

>2. The JS App needs to pick which one it is going to choose to have a medi=
a session with.

Yes.

>3. Let's say it choose UA1, then it would map the Answer 1a to an Answer, =
the Offer2 to an offer.

Here comes the issue:

According to jsep-02, when Offer2 comes the JSEP O/A state machine is in th=
e "pranswer" state, where a new offer is not accepted.

And, yes, we could fix that by accepting a new offer at that point. But, to=
 which state would be then go?

>4. If it was going to choose UA2, it would not pass the answer down to the=
 browser at all and would instead wait till it Answer 1b and pass that down=
.

Sure.

>5. If you think it should simultaneously have a media session with both UA=
1 and UA2, then things are more complicated. I realize that at the time it =
gets ANSWER 1a, it does not know it will later get ANSWR 1b so that make it=
 hard to choose UA2.

I am assuming serial forking. Once the answer from UA2 has been provided to=
 the browser, I don't care if the media session with UA1 goes away.

But, again, my main issue at this point is not about media - it's about how=
 the offers and answers between the JS app and the UAs are mapped to JSEP o=
ffer/pranswer/answer.

> Parallel forking with early media and secure media is a hard problem in S=
IP as we know from the HERFP discussion. I'm unaware of real world problem =
that are solved well with
> SIP and require parallel forking and thus my focus has been on making sur=
e we solve serial forking.

No disagreement from my side there :)=20

> I'm perfectly happy if we *also* solve parallel forking but I don't care =
too much as long a early media serial forking scenarios work because theses=
 do happen all the time.

At the moment, JSEP-02 does talk about cloning. I don't know whether you su=
ggest that to be removed, but if it stays it would work also for serial for=
king.

Regards,

Christer=

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] PRANSWER is unusable for serial forking
Thread-Index: AQHNqOnc3QfmtiokskqSBC02DG7AB5e21Ob3gACuEoA=
Date: Sat, 13 Oct 2012 12:44:31 +0000
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Subject: Re: [rtcweb] PRANSWER is unusable for serial forking
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On Oct 13, 2012, at 1:21 AM, Christer Holmberg <christer.holmberg@ericsson.=
com>
 wrote:

> Hi Cullen,
>=20
> I'll read your reply more in detail later, but please note that I assume =
that the 180 is sent *reliably* (I left the PRACK out).

ok - that was probably part of the confusing was we were reading that diffe=
rntly=

From pkyzivat@alum.mit.edu  Sun Oct 14 12:26:10 2012
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On 10/12/12 11:59 AM, Cullen Jennings (fluffy) wrote:
>
> On Oct 9, 2012, at 4:58 PM, Martin Thomson <martin.thomson@gmail.com> wrote:
>
>> On 9 October 2012 13:05, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>>> 3264, SDP, 3261, and related documents are dealing with a bunch of things including what happens at media plane and signaling plane.
>>>
>>> I'll note that though O/A is per dialog, there is only one O shared across multiple legs and when you create the O you don't know how many dialogs there will be. So from the media point of view (covered more in SDP spec than 3264) there is one O with a bunch of A. From signaling point of view there are a bunch of O/A pairs).
>>>
>>> The dialog is SIP signaling concept not a media plane level concept. We moved the signaling part out of the browser and into the JS. But the media part is still in the browser. So as 3264 says, after the offer is constructed, we have to be willing to receive media for all the codec type in the offer. When we get the answer 3264 makes it clear that one MAY stop receiving the codecs that were in the offer but not selected in the answer. However, this can not be done until the signaling layer is sure that no more offers will be honored. Since that signalling part of Offer/Answer is in the JS, the API need to to have a way to signal what the MAY part should do and that is the PRANSWER vs ANSWER.
>>
>> That's a consistent and logical view.  It is not what RFC 3264 says.
>> It may even be the only reasonable solution.
>>
>>> People keep trying to make this some complex weird argument invoking the SIP deities of the past and quoting incomprehensible phrases from various RFC caved in stone [...]
>>
>> Quit beating around the bush: it's me you are talking about.
>>
>> It's not a complex weird argument, it's this:
>>   PRANSWER is not described in RFC 3264.
>>
>> draft-*-jsep might say something about it, but I'd have to guess about
>> what to do if I were to implement the feature and that doesn't help
>> with interoperation.
>>
>> If there was text that described the above, plus the implications in
>> some detail (including what resources can be released as they relate
>> to the SDP), then we're good.  I don't even think that writing this is
>> especially hard.
>
> Agree that text has to be a clear. But provision is a term from 3261 which is referenced from 3264 and bunch of this also relies on SDP RFC too. Some of the key information about it is 3262. This is all intertwined but my key point is that I don't think we are trying to do anything that was not part of 3264 offer / answer. This is more or less a SIP 180.

I am trying to follow this argument, but am having difficulty, even 
though I agree with much of what has been said by those who seem to be 
on different sides of the "argument".

There is much about what needs to be done in presence of forking that is 
not written, but which is a corollary to what is specified. Most of what 
Christer described falls into this category. Implementations that don't 
do this will fail on some cases.

The problem I have with PRANSWER is that I don't if it ever applies when 
sip is at one end, and if so, how it applies. IMO the simple answer is 
that it is an extension that should not be used when gatewaying to sip. 
But now I think I see people saying it is only there for gatewaying to 
sip. If so, then I want to see a specification for how this maps.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Sun Oct 14 12:43:33 2012
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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On 10/10/12 4:03 AM, Christer Holmberg wrote:
> Hi,
>
>>>>> I have not seen any reason to relax 3264 yet but if something comes up, agree we should carefully look at the cases. I think we can just do straight up 3264.
>>>>
>>>> RFC 3264 doesn't describe PRANSWER.  The concept is entirely absent.
>>>>
>>>>   The offerer MAY immediately cease listening for media formats that
>>>> were listed in the initial offer, but not present in the answer.
>>>>
>>>> "the" answer.
>>>
>>> I agree with Martin. 3264 O/A is always per dialog, and forking is supported by generating multiple dialogs. JSEP, OTOH, in order to support forking with a
>>> single "dialog" (peerConnection local descriptor), now defines O/A as offer+any number of pranswers + answer.
>>>
>>> So, we would e.g. have to define what happens if a new offer is received from the remote side while the browser is in pranswer-received state (see my call flow in another reply).
>>
>> 3264, SDP, 3261, and related documents are dealing with a bunch of things including what happens at media plane and signaling plane.
>>
>> I'll note that though O/A is per dialog, there is only one O shared across multiple legs and when you create the O you don't know how many dialogs there will be.
>
> The *initial* O is shared across multiple legs. Once a dialog has been created, O/A transactions on that leg will not affect other legs.

+1

I'll also observe that you can have early dialogs without reliable 
provisionals. E.g.:

- A sends INIVTE offering codecs C1 and C2
- proxy P forks first to B.
- B sends unreliable 180, tag BBB, with answer selecting codec C1,
   and starts sending media.
- P cancels the invite to B and forwards the invite to C
- C sends unreliable 180, tag CCC, with answer selecting codec C2,
   and starts sending media
- C sends 200 with same answer as before.

Note that A must not invalidate the use of C2 when it receives the 180 
from B, unless it first clones the invite. It must leave it eligible for 
use on other dialogs. And of course while it can manage separate O/A 
state machines per dialog, down at the RTP stack it may not be able to 
correlate the incoming media with a particular one of those O/A state 
machines.

(Consider the example above, but the 180 from B is lost - never received 
by A. Since it isn't reliable, nobody knows this. But the media still 
comes.)

Of course things are a bit different when you layer ICE on top of this. 
I won't try to discuss that.

	Thanks,
	Paul



From harald@alvestrand.no  Mon Oct 15 09:38:43 2012
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Subject: [rtcweb] Fwd: New Version Notification for draft-alvestrand-rtcweb-vp8-00.txt
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This is a multi-part message in MIME format.
--------------090502000200000308030305
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As the chairs asked for - here is some written input for the video codec 
MTI discussion.

    Harald


-------- Original Message --------
Subject: 	New Version Notification for draft-alvestrand-rtcweb-vp8-00.txt
Date: 	Mon, 15 Oct 2012 09:31:31 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no
CC: 	agrange@google.com



A new version of I-D, draft-alvestrand-rtcweb-vp8-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-alvestrand-rtcweb-vp8
Revision:	 00
Title:		 VP8 as RTCWEB Mandatory to Implement
Creation date:	 2012-10-15
WG ID:		 Individual Submission
Number of pages: 7
URL:             http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-vp8-00.txt
Status:          http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8
Htmlized:        http://tools.ietf.org/html/draft-alvestrand-rtcweb-vp8-00


Abstract:
    This document recommends that the RTCWEB working group choose the VP8
    specification as a mandatory to implement video codec for RTCWEB
    implementations.

    This document is not intended for publication as an RFC.


                                                                                   


The IETF Secretariat





--------------090502000200000308030305
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    As the chairs asked for - here is some written input for the video
    codec MTI discussion.<br>
    <br>
       Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-alvestrand-rtcweb-vp8-00.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Mon, 15 Oct 2012 09:31:31 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:agrange@google.com">agrange@google.com</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-alvestrand-rtcweb-vp8-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-alvestrand-rtcweb-vp8
Revision:	 00
Title:		 VP8 as RTCWEB Mandatory to Implement
Creation date:	 2012-10-15
WG ID:		 Individual Submission
Number of pages: 7
URL:             <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-vp8-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-vp8-00.txt</a>
Status:          <a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8">http://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-alvestrand-rtcweb-vp8-00">http://tools.ietf.org/html/draft-alvestrand-rtcweb-vp8-00</a>


Abstract:
   This document recommends that the RTCWEB working group choose the VP8
   specification as a mandatory to implement video codec for RTCWEB
   implementations.

   This document is not intended for publication as an RFC.


                                                                                  


The IETF Secretariat

</pre>
      <br>
      <br>
    </div>
    <br>
  </body>
</html>

--------------090502000200000308030305--

From partha@parthasarathi.co.in  Mon Oct 15 10:55:28 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
References: <506B0367.4000103@ericsson.com> <C5E08FE080ACFD4DAE31E4BDBF944EB111867718@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD03A6@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB11187F8F1@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAD087E@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1118810C3@xmb-aln-x02.cisco.com>, <CABkgnnV8YbbPi3pZt9mBjBfSgkx4EgL=EY0YogpeRobUMQ506w@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAFAA6B@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB1118819FB@xmb-aln-x02.cisco.com> <000b01cda720$bd5097a0$37f1c6e0$@co.in>, <C5E08FE080ACFD4DAE31E4BDBF944EB11188661A@xmb-aln-x02.cisco.com> <7F2072F1E0DE894DA4B517B93C6A0585340BAFAA86@ESESSCMS0356.eemea.ericsson.se> <C5E08FE080ACFD4DAE31E4BDBF944EB111888AC9@xmb-aln-x02.cisco.com>
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Date: Mon, 15 Oct 2012 23:25:12 +0530
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Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
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Cullen,

As you mentioned in another mail thread, the discussion is about mapping
UPDATE with SDP just after reliable 18x with SDP in JSEP PRANSWER state.
Hope you agree that PRANSWER does not address UPDATE with SDP (OFFER from
remote) in the early dialog scenario.

Thanks
Partha

-----Original Message-----
From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com] 
Sent: Saturday, October 13, 2012 7:52 AM
To: Christer Holmberg
Cc: Parthasarathi R; <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda requests for Atlanta meeting


On Oct 12, 2012, at 11:43 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:

> Hi,
> 
>> I fully agree this needs to work early media to PSTN via SIP. I don't see
any problem with doing that. I'm happy to work through detail call flows to
make sure that early media to PSTN works.
>> 
>> Update after a 180 is not allowed by the Update RFC as I pointed out in
other email exactly because just in SIP with no RTCP web, the UA that caret
the invite would not have enough information to be able to process this as
you point in the thread below.
> 
> As far as I know, UPDATE is allowed after a reliable 180. If you don't
agree, please show me the text which forbids it :)
> 
> Regards,

I agree - but it's not ok after a normal 180


From internet-drafts@ietf.org  Mon Oct 15 16:48:27 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-qos-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : DSCP and other packet markings for RTCWeb QoS
	Author(s)       : Subha Dhesikan
                          Dan Druta
                          Paul Jones
                          James Polk
	Filename        : draft-ietf-rtcweb-qos-00.txt
	Pages           : 10
	Date            : 2012-10-15

Abstract:
   Many networks, such as Service Provider and Enterprise networks, can
   provide per packet treatments based on Differentiated Services Code
   Points (DSCP) on a per hop basis.  This document defines the
   recommended DSCP values for browsers to use for various classes of
   traffic.

   This draft is a very early and far from done.  It is meant to provide
   the structure for the idea of how to do this but much discussion is
   needed about the details.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-qos-00


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


From dbenham@cisco.com  Mon Oct 15 17:17:47 2012
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From magnus.westerlund@ericsson.com  Tue Oct 16 04:20:38 2012
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Subject: [rtcweb] Summary of Video Codec contributions
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WG,

The Video Codec selection internet draft to RTCWEB WG I have seen are
the following four:

http://www.ietf.org/internet-drafts/draft-dbenham-webrtc-videomti-00.txt

https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8/

https://datatracker.ietf.org/doc/draft-burman-rtcweb-h264-proposal/

https://datatracker.ietf.org/doc/draft-marjou-rtcweb-video-codec/

Please review information and proposals and lets start discussing them
on the mailing list prior to the WG meeting in Atlanta.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From neils@vipadia.com  Tue Oct 16 06:35:40 2012
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On Tue, Oct 16, 2012 at 12:20 PM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> WG,
>
> The Video Codec selection internet draft to RTCWEB WG I have seen are
> the following four:
>
> http://www.ietf.org/internet-drafts/draft-dbenham-webrtc-videomti-00.txt
>
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8/
>
> https://datatracker.ietf.org/doc/draft-burman-rtcweb-h264-proposal/
>
> https://datatracker.ietf.org/doc/draft-marjou-rtcweb-video-codec/
>
> Please review information and proposals and lets start discussing them
> on the mailing list prior to the WG meeting in Atlanta.

Something that concerns me about video codecs is the availability of a
scalable coding profile. Many plugins that we are trying to replace
with WebRTC appear to be starting to make use of scalable coding
techniques to enable efficient multi-party calls. To encourage the
replacement of such plugins the codec specified in WebRTC should offer
the ability to match what is best practice today.

I understand that H.264 has Annex G, and that VP8 has something
similar. Are there any issues around making this MTI for either codec
that would help drive a decision?

Neil

From harald@alvestrand.no  Tue Oct 16 07:49:14 2012
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On 10/16/2012 03:35 PM, Neil Stratford wrote:
> On Tue, Oct 16, 2012 at 12:20 PM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> WG,
>>
>> The Video Codec selection internet draft to RTCWEB WG I have seen are
>> the following four:
>>
>> http://www.ietf.org/internet-drafts/draft-dbenham-webrtc-videomti-00.txt
>>
>> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8/
>>
>> https://datatracker.ietf.org/doc/draft-burman-rtcweb-h264-proposal/
>>
>> https://datatracker.ietf.org/doc/draft-marjou-rtcweb-video-codec/
>>
>> Please review information and proposals and lets start discussing them
>> on the mailing list prior to the WG meeting in Atlanta.
> Something that concerns me about video codecs is the availability of a
> scalable coding profile. Many plugins that we are trying to replace
> with WebRTC appear to be starting to make use of scalable coding
> techniques to enable efficient multi-party calls. To encourage the
> replacement of such plugins the codec specified in WebRTC should offer
> the ability to match what is best practice today.
>
> I understand that H.264 has Annex G, and that VP8 has something
> similar. Are there any issues around making this MTI for either codec
> that would help drive a decision?
For one thing, I believe the hardware support for SVC is minimal to 
nonexistent.
People have also been raising doubts about the cost/benefit tradeoff of 
H.264 SVC.

At the moment, we have no proposals to adopt SVC as MTI, so I believe 
the point is moot.

Of course, any implementation that supports codecs beyond the MTI will 
be free to offer them as part of the negotiation phase. But that doesn't 
make them MTI.

                   Harald




From bernard_aboba@hotmail.com  Tue Oct 16 10:35:39 2012
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--485b390f7df4d78e2604cc3d7388
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Thought it important to point out the fact that FREE-P8 (VP8) would likely
be the preferred codec implemented by independent developers. Royalties and
admin headaches create a barrier to entry. Independent dev shops equate to
a great deal of innovation and some of these have been responsible for
dictating the standard in market with hundreds of millions of users. Yes,
there is plenty of interoperability with 264 today but that doesn't mean it
will be that way tomorrow.

Can't we do both, as we have done for Audio MTI? Cost for implementation of
both is not really that high and if VP8 is included it will likely be the
choice for rtc indie devs globally, spurring innovation.

If unknown IPR is the only real reason not to include VP8 then I maybe we
are not trying hard enough. RTCweb / WebRTC is supposed to be the next big
thing, why hamper innovation like this?

I understand this will likely not count as a vote, just wanted to voice
some concerns. My apologies in advance if this has been answered in another
thread or offline. I searched the threads and found no reference and my
double isn't always available.

/Erik

*Erik Lagerway <http://ca.linkedin.com/in/lagerway> |
*Hookflash<http://hookflash.com>
* | m 1.604.562.8647*
****


On Tue, Oct 16, 2012 at 4:20 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> WG,
>
> The Video Codec selection internet draft to RTCWEB WG I have seen are
> the following four:
>
> http://www.ietf.org/internet-drafts/draft-dbenham-webrtc-videomti-00.txt
>
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8/
>
> https://datatracker.ietf.org/doc/draft-burman-rtcweb-h264-proposal/
>
> https://datatracker.ietf.org/doc/draft-marjou-rtcweb-video-codec/
>
> Please review information and proposals and lets start discussing them
> on the mailing list prior to the WG meeting in Atlanta.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--485b390f7df4d78e2604cc3d7388
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thought it important to point out the fact that FREE-P8 (VP8) would likely =
be the preferred codec implemented by independent developers. Royalties and=
 admin headaches create a barrier to entry. Independent dev shops equate to=
 a great deal of innovation and some of these have been responsible for dic=
tating the standard in market with hundreds of millions of users. Yes, ther=
e is plenty of interoperability with 264 today but that doesn&#39;t mean it=
 will be that way tomorrow.<br>
<br>Can&#39;t we do both, as we have done for Audio MTI? Cost for implement=
ation of both is not really that high and if VP8 is included it will likely=
 be the choice for rtc indie devs globally, spurring innovation.<br><br>
If unknown IPR is the only real reason not to include VP8 then I maybe we a=
re not trying hard enough. RTCweb / WebRTC is supposed to be the next big t=
hing, why hamper innovation like this?<br><br>I understand this will likely=
 not count as a vote, just wanted to voice some concerns. My apologies in a=
dvance if this has been answered in another thread or offline. I searched t=
he threads and found no reference and my double isn&#39;t always available.=
 <br>
<br>/Erik<br><br clear=3D"all"><font color=3D"#943634" face=3D"arial, sans-=
serif"><span style=3D"border-collapse:collapse;line-height:14px"><span styl=
e=3D"border-collapse:separate;color:rgb(0,0,0);font-family:arial;line-heigh=
t:normal"><span style=3D"font-family:arial,sans-serif;border-collapse:colla=
pse;color:rgb(148,54,52);line-height:14px"><b><span style=3D"color:rgb(0,0,=
0);font-weight:normal;font-size:13px"><font color=3D"#943634"><span style=
=3D"font-size:small"><b><span style=3D"color:rgb(0,0,0);font-weight:normal;=
font-size:13px"><span style=3D"font-size:8pt;line-height:12px;color:gray"><=
a href=3D"http://ca.linkedin.com/in/lagerway" target=3D"_blank"><span style=
=3D"color:rgb(204,0,0)">Erik Lagerway</span></a> | </span></span></b></span=
></font></span></b><a href=3D"http://hookflash.com" target=3D"_blank"><span=
 style=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><font color=
=3D"#943634"><span style=3D"font-size:small"><span style=3D"color:rgb(0,0,0=
);font-weight:normal;font-size:13px"><span style=3D"font-size:8pt;line-heig=
ht:12px;color:gray"></span></span></span></font></span><span style=3D"color=
:rgb(0,0,0);font-weight:normal;font-size:13px"><font color=3D"#943634"><spa=
n style=3D"font-size:small"><span style=3D"color:rgb(0,0,0);font-weight:nor=
mal;font-size:13px"><span style=3D"font-size:8pt;line-height:12px;color:gra=
y"><span style=3D"color:rgb(51,51,51)">Hookflash</span></span></span></span=
></font></span></a><span style=3D"color:rgb(0,0,0);font-weight:normal;font-=
size:13px"><font color=3D"#943634"><span style=3D"font-size:small"><span st=
yle=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=3D"f=
ont-size:8pt;line-height:12px;color:gray"></span></span></span></font></spa=
n><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><fo=
nt color=3D"#943634"><span style=3D"font-size:small"><b><span style=3D"colo=
r:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=3D"font-size:8p=
t;line-height:12px;color:gray"> | m 1.604.562.8647</span></span></b></span>=
</font></span></b></span></span></span></font><br>
<font color=3D"#943634" face=3D"arial, sans-serif"><span style=3D"border-co=
llapse:collapse;line-height:14px"><span style=3D"border-collapse:separate;c=
olor:rgb(0,0,0);font-family:arial;line-height:normal"><span style=3D"font-f=
amily:arial,sans-serif;border-collapse:collapse;color:rgb(148,54,52);line-h=
eight:14px"><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-size=
:13px"><font color=3D"#943634"><span style=3D"font-size:small"><b><span sty=
le=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=3D"fo=
nt-size:10pt;line-height:14px;color:rgb(148,54,52)"></span><span style=3D"f=
ont-size:8pt;line-height:12px"></span></span></b></span></font></span></b><=
/span></span></span></font><font color=3D"#943634" face=3D"arial, sans-seri=
f"><span style=3D"border-collapse:collapse;line-height:14px"><span style=3D=
"border-collapse:separate;color:rgb(0,0,0);font-family:arial;line-height:no=
rmal"></span></span></font><font color=3D"#943634" face=3D"arial, sans-seri=
f"><span style=3D"border-collapse:collapse;line-height:14px"><span style=3D=
"border-collapse:separate;color:rgb(0,0,0);font-family:arial;line-height:no=
rmal"><span style=3D"font-family:arial,sans-serif;border-collapse:collapse;=
color:rgb(148,54,52);line-height:14px"><b><span style=3D"color:rgb(0,0,0);f=
ont-weight:normal;font-size:13px"><font color=3D"#943634"><span style=3D"fo=
nt-size:small"><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-s=
ize:13px"><span style=3D"font-size:10pt;line-height:14px;color:rgb(148,54,5=
2)"></span><span style=3D"font-size:8pt;line-height:12px"></span></span></b=
></span></font></span></b></span></span></span></font><font color=3D"#94363=
4" face=3D"arial, sans-serif"><span style=3D"border-collapse:collapse;line-=
height:14px"><span style=3D"border-collapse:separate;color:rgb(0,0,0);font-=
family:arial;line-height:normal"></span></span></font><br>

<br><br><div class=3D"gmail_quote">On Tue, Oct 16, 2012 at 4:20 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</span> wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">WG,<br>
<br>
The Video Codec selection internet draft to RTCWEB WG I have seen are<br>
the following four:<br>
<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-dbenham-webrtc-videomt=
i-00.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-dbenh=
am-webrtc-videomti-00.txt</a><br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-vp8/" t=
arget=3D"_blank">https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-v=
p8/</a><br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-burman-rtcweb-h264-propos=
al/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-burman-rtcweb=
-h264-proposal/</a><br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-marjou-rtcweb-video-codec=
/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-marjou-rtcweb-v=
ideo-codec/</a><br>
<br>
Please review information and proposals and lets start discussing them<br>
on the mailing list prior to the WG meeting in Atlanta.<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br>

--485b390f7df4d78e2604cc3d7388--

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Hi Erik,

On Wed, Oct 17, 2012 at 01:56:55AM -0700, Erik Lagerway wrote:
> Thought it important to point out the fact that FREE-P8 (VP8) would likely
> be the preferred codec implemented by independent developers. Royalties and
> admin headaches create a barrier to entry. Independent dev shops equate to
> a great deal of innovation and some of these have been responsible for
> dictating the standard in market with hundreds of millions of users. Yes,
> there is plenty of interoperability with 264 today but that doesn't mean it
> will be that way tomorrow.

Very true.  Though for developers who don't click-track and count users
of their software it is more than just preferred.  An RF solution is
actually the only viable way for this standard to be accessible to them.

Any standard that immediately excludes a large portion of potential
developers and adopters can only ever be of niche value and is certain
to be replaced before long by an alternative that doesn't have that
constraint as a fundamental flaw preventing its broadest use.

If the whole point of this group was to avoid further fracturing of
the developer efforts in this space, then solving this problem now
would be much preferable to forcing another group to develop another
competing standard that does address that issue.  It would be a sad
waste of the invested time of the people here if we didn't recognise
this necessity.

I think the points you make above are very relevant here.


> Can't we do both, as we have done for Audio MTI?

Well, the M in MTI stands for Mandatory.  So mandating both would be
mandating the barriers to entry that you so clearly identified above?

Mandating an RF video codec, and permitting any other to be negotiated
as an extension would be more like what we decided was best for audio.


> If unknown IPR is the only real reason not to include VP8 then I maybe we
> are not trying hard enough.

If that's the only reason people have for opposing VP8 then maybe they're
trying too hard.  Resorting to imaginary reasons to make the issue seem
more complex than it really is doesn't seem very enlightening.

Every day that goes by leaves this argument dangling from a thinner, more
strained thread of tenuousness.  It's been well over a year since the very
public call to form an offensive pool was made or even since the last very
vague mumbling of "yeah, some people responded.  honest." - meanwhile VP8
continues to serve millions of users every day through things like youtube
et al. with no concrete claim being made against them of infringement.

It's not like this is a secret, so any submarine patent holder must know
that laches is now daily eroding their 'investment'.  The extensive patent
search made in the attempt to form a pool surely only weakens their case
further in the absence of any real disclosure or claim being made - if not
outright coming close in itself to 'proving' that no such IPR exists.

If VP8 was mandated here, I presume that would additionally place an
obligation of disclosure on any of the participants of this group, which
would strengthen that protection even more.

If we get to the end of this process and there are still no actual claims
being presented for substantiation, then it seems reasonable to conclude
that we may be as safe as it is ever possible to be from "unknown IPR"
(until the laws are changed to end that kind of madness once and for all).


By contrast the present day known burdens of H.264 are very real, and if
we're going to project our future concerns, then I think we've got more
to worry about from the royalties for H.264 being dialled up greatly as
the patents on it begin to expire and their holders want to 'encourage'
people to switch to H.265 while its patents are still nice and fresh.

Locking people into that seems like a far greater danger.  Especially
since unlike VP8, H.264 really has had submarine patents surface on it
and really is an exposed pawn in the present theatre of patent warfare
that everyone else seems to be ducking stray bullets from these days.

So spruiking there's somehow a spookier risk to VP8 in the face of all
available evidence to the contrary seems like some sport where you have
to squint pretty hard and tip your head 90 degrees to the left in a
dark room, before that line of plot extrapolation might perhaps appear
to be spinning in an even vaguely-real kind of direction, sort of ...

And even then, only if the people wanting you to buy that proposition
don't properly label their axes ...


> RTCweb / WebRTC is supposed to be the next big thing, why hamper
> innovation like this?

Indeed we should not, and I for one certainly hope consensus is
that we will not.


 Stay sharp!
 Ron



From worley@shell01.TheWorld.com  Wed Oct 17 13:05:11 2012
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Subject: [rtcweb] WebRTC offer/answer design and corresponding SIP gateway design
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> From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
> 
> But we all need to get this thread to specific examples about real
> world use cases that people can understand.

This is an analysis of how SIP/WebRTC gateway design depends upon and
reflects the offer/answer architecture that is chosen for WebRTC.  The
unexpected result is that a SIP/WebRTC gateway can work well with a
*simpler* WebRTC offer/answer architecture than is being discussed,
indeed, an O/A architecture resembling a single caller/callee pair.

* Background

The overall task that must be performed by a SIP/WebRTC-server gateway
combined with the WebRTC-client is the same as what a SIP telephone
does.  A detailed discussion is in my previous message
http://www.ietf.org/mail-archive/web/rtcweb/current/msg05435.html.
The important point for this discussion is:  Quality handing of
outgoing SIP calls requires integrating (mixing) information from both
the signaling and media for all early dialogs to generate the media
stream to be presented to the user.

Let me now outline three architectures and their characteristics.  I
still haven't checked the consequences of requiring ICE, but I believe
that it does not fundamentally change the analysis.

* cloning

In the cloning architecture, the WebRTC protocol reveals the multiple
dialogs of the SIP call.  When the client Javascript initiates the
call, a "call object" is created.  When the SIP signaling reveals to
the gateway the existence of a new early dialog, the gateway uses
WebRTC to tell the client of the new dialog, and a child "dialog
object" is created.  The dialog objects are continually updated by
information from the WebRTC server to tell the Javascript the status
of the early dialogs.

The WebRTC client is the source/target of the media streams.  (If it
is possible for an early dialog's media stream to arrive at the client
before corresponding SIP signaling, the client stack creates
additional dialog objects for those early dialogs.)

The client stack or Javascript must integrate the information in the
signaling and media to produce the audio stream to be presented to the
user.

This architecture is the most complicated form of WebRTC.  It also
requires that the client stack and/or Javascript implement the full
processing required by a SIP phone in all cases, because in general,
the client-side code never knows whether a communication session is
directed to a SIP gateway or not.

* PRANSWER

In the PRANSWER architecture, the WebRTC protocol carries SDP in a
manner that is similar to SDP offer/answer between two endpoints.  The
exception is that the offer provided with call initiation may receive
a sequence of answers, which inform the client of the origins of
successive early dialogs that it should listen to.

The WebRTC client is the source/target of the media streams.

This architecture presupposes that there will be only one early dialog
at a time that is producing media that the user needs to hear, and
that the gateway can determine from the SIP signaling what this
sequence of dialogs is.

The difficulty is that neither of these presuppositions is correct:
Given parallel forking, the call may have reached destinations that
are both producing important feedback at the same time.  Also, the SIP
signaling does not completely inform the SIP caller which destinations
are producing media streams that contain information -- that decision
requires determining if RTP is arriving from each destination and
whether the RTP is non-silent.  In addition, if the signaling shows
that a destination is ringing but is not providing media, synthetic
ringback media should be provided.  This architecture requires that
the gateway make these decisions despite not being the endpoint of the
media streams.

* simple

In the "simple" architecture, the WebRTC protocol models a *single*
offer/answer dialog, and all of the media processing required for a
SIP early dialog is placed in the gateway:  Initially, the gateway
provides an answer giving *itself* as the server end of the media
streams.  During the early phase of the call, the gateway decodes,
synthesizes, and mixes all of the early media according to the
principles for SIP telephones, and sends a single RTP stream to the
WebRTC client that is the audio the user should hear.  This is
relatively easy because the gateway has direct access to both the
signaling and the media from the SIP destinations.

At some point (probably when one dialog becomes established), the
gateway decides to remove itself from the media processing, and does a
"Z operation" to pass a new offer/answer pair between the now-unique
UAS and the WebRTC client.

(A "Z operation" is named for its appearance on protocol diagrams:  An
intermediate device sends a request to the endpoint in one direction
demanding that it produce an offer; the intermediate device passes the
offer through to the other endpoint, which produces an answer; the
intermediate device passes the answer through back to the first
endpoint.)

This architecture has several advantages:

(1) The WebRTC protocol implements only a single offer/answer
sequence.

(2) The gateway is responsible for combining the media and signaling
information of the early dialogs; the multiple early dialogs are not
revealed in WebRTC.  This processing is relatively easy to implement,
as it can be implemented in about the same way that SIP phones now
implement it.

(3) Improving the handling of multiple early dialogs can be done
entirely by updating the gateway; the WebRTC protocol does not
commit the processing to any particular model.

(4) WebRTC client stacks and application Javascript provide no special
support for SIP forking; all connections appear to them to be "a direct
connection to a media server".

Dale

From matthew.kaufman@skype.net  Wed Oct 17 13:25:37 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: Relaxing SDP O/A (was RE: [rtcweb] Agenda requests for Atlanta meeting)
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From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings (fluffy)
Sent: Tuesday, October 9, 2012 7:17 AM

> I have not seen any reason to relax 3264 yet but if something comes up, a=
gree we should carefully look at the cases. I think we can just do straight=
 up > 3264. Arguments that SIP early media in a 180 is not compliant with 3=
264 are just wrong.

We've *already* "relaxed" 3264's requirements in JSEP (comments apply to dr=
aft-ietf-rtcweb-jsep-01)
=09
Straight from 3264: "At any time, either agent MAY generate a new offer tha=
t updates the session. However, it MUST NOT generate a new offer if it has =
received an offer which it has not yet answered or rejected. Furthermore, i=
t MUST NOT generate a new offer if it has generated a prior offer for which=
 it has not yet received an answer or a rejection. If an agent receives an =
offer after having sent one, but before receiving an answer to it, this is =
considered a "glare" condition."

The JSEP API enforces no such rules. If you want to change the API to bring=
 it into compliance with *just this section* of 3264, it would need all sor=
ts of changes...

- 5.1.1 createOffer ... "Calling this method does not change state; its use=
 is not required". Ok, so how then is JSEP to enforce the two "MUST NOT gen=
erate a new offer" is the quote from 3264 above? I would think that createO=
ffer must return an error whenever the "MUST NOT" cases apply. (Never mind =
that if the "use is not required" then somehow I'm to guess what SDP is leg=
al for this browser to pass in to setLocalDescription as an offer?)

- 5.1.4 setLocalDescription... has no language indicating that it is prohib=
ited to call "setLocalDescription" with an offer and then call it again wit=
h a different offer without having supplied an answer. There also appears t=
o be no way to pass a "reject" in. (These two issues appear in the W3C API =
state description too, so appears to really be what we mean when we say tha=
t we're going to "use JSEP")

- 5.1.2 createAnswer... "Calling this method does not change state; its use=
 is not required". So then how can setLocalDescription know when it is forb=
idden to accept an "offer" passed to it (as createOffer is not required) wh=
en setRemoteDescription has been passed an "offer" but "createAnswer" has n=
ot yet been called?

And so on.

This isn't the only section of 3264 that's gone out the window with JSEP, b=
ut saying "I have not seen any reason to relax 3264 yet" doesn't make any s=
ense... that ship sailed when JSEP was proposed.

Matthew Kaufman

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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Justin Uberti <juberti@google.com>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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From: =?iso-8859-1?Q?G=F6ran_Eriksson_AP?= <goran.ap.eriksson@ericsson.com>
To: "sdeshika@cisco.com" <sdeshika@cisco.com>, "DRUTA, DAN" <dd5826@att.com>,  "paulej@packetizer.com" <paluej@packetizer.com>
Thread-Topic: Comments on draft-ietf-rtcweb-qos-00.txt
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-qos-00.txt
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Hi!

I have some questions/comments about the draft (which I think is a nice sta=
rt BTW, :-)!).

Chapter 1
=3D=3D=3D=3D=3D=3D=3D=3D=3D

Paragraph 2: "This draft proposes how browser and other VOIP applications..=
.

Question: Addressing also non-browser "apps" widens the scope and this has=
=20
implications on the amount to work that need to be covered to secure=20
completeness and correctness. Differences in security model for browser app=
s
vs other execution environments would need to be addressed. Would it hurt m=
uch to
limit the scope to browser *only*?

Paragraph 6: "...defines some input..."

Question: Are the references included in the "some", that is the use of tra=
fficclass
attribute in SDP [6] part of what webRTC should support, e.g. the solution =
for
the case with site (network) specific DSCP values?

Question: "Some" implies other inputs as well- should this not be an exhaus=
tive list if what governs the browser decision to ensure interoperability=20
between different implementations? Or am I missing something?

Chapter 4
=3D=3D=3D=3D=3D=3D=3D=3D=3D
Question: I assume the "relative treament" input the Low, Medium and High v=
alues in Table 1?

Question: With "session" I understand one PeerConnection instantiation- cor=
rect?

Question: The priority indication used by the web app is coming from somewh=
ere- is
this an SDP issue?


Chapter 6
=3D=3D=3D=3D=3D=3D=3D=3D=3D
Question: What is the intent of having the LTE QCI Mapping table in the dra=
ft? Which
QCI that actually in the end will be associated to the trafficclass attribu=
te is
decided in the cellular network and it is I guess far from for sure one wou=
ld get that QCI? Also, there are only a limited set of standardised QCI's- =
there are and will be custom made QCI's out there. Isn't there a risk that =
by putting it the readers will be lead to believe the web app will get that=
 treatment? Perhaps an "example LTE QCI mapping" is more appropriate?

Comment: The reference to the GBR bearer in LTE leads us into the discussio=
n about
number and value of bw parameters in SDP for Max and Min bw. Perhaps just h=
ighlighting that SDP issue/API requirement is in place referring to the app=
ropriate rfc/draft where this will be covered?

Chapter 8
=3D=3D=3D=3D=3D=3D=3D=3D=3D
Question: Should not the trafficclass attribute [6] be mentioned here?

Question: Should not the statistics API also include this information?

Chapter 9:
=3D=3D=3D=3D=3D=3D=3D=3D=3D
Comment: The need to figure out IP address and port used makes sense but pe=
rhaps the background to that requirement could be explained more in the cha=
pters before since it covers sofar only on the use of priority and DSCP?

Comment: I guess one potential risk is a maliceous/poortly written web app
can mess with the trafficclass attribute/priority labeling/bw attribute and=
=20
if the marking in the IP header cannot be trusted, then it will I guess be
discarded in the network, e.g. in the LTE modem receiving the IP packet fro=
m the browser when it decides which "QCI" to use- no trust and it ends up o=
n the default bearer, :-).

Many regards
GE


> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of internet-drafts@ietf.org
> Sent: den 16 oktober 2012 01:48
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-qos-00.txt
>=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in=20
> WEB-browsers Working Group of the IETF.
>=20
> 	Title           : DSCP and other packet markings for RTCWeb QoS
> 	Author(s)       : Subha Dhesikan
>                           Dan Druta
>                           Paul Jones
>                           James Polk
> 	Filename        : draft-ietf-rtcweb-qos-00.txt
> 	Pages           : 10
> 	Date            : 2012-10-15
>=20
> Abstract:
>    Many networks, such as Service Provider and Enterprise=20
> networks, can
>    provide per packet treatments based on Differentiated Services Code
>    Points (DSCP) on a per hop basis.  This document defines the
>    recommended DSCP values for browsers to use for various classes of
>    traffic.
>=20
>    This draft is a very early and far from done.  It is meant=20
> to provide
>    the structure for the idea of how to do this but much discussion is
>    needed about the details.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-qos
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-qos-00
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From pkyzivat@alum.mit.edu  Wed Oct 17 15:56:46 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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Matthew,

IIUC, JSEP is what can euphemistically called an "extended subset" of 
3264. (Or less euphemistically, an incompatible dialect.)

If so, how is one to interwork this with SIP components that are 3264 
conformant?

I have been assuming that this is to be accomplished by the server 
pushing javascript that is aware it is/may be interworking with sip, and 
which then constrains itself to the subset of JSEP that compatible with 
3264. And then it is of course limited to those behaviors on the sip 
side that fall within the subset of 3264 that JSEP supports.

Is that the right way to view it?

	Thanks,
	Paul

On 10/17/12 4:25 PM, Matthew Kaufman wrote:
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings (fluffy)
> Sent: Tuesday, October 9, 2012 7:17 AM
>
>> I have not seen any reason to relax 3264 yet but if something comes up, agree we should carefully look at the cases. I think we can just do straight up > 3264. Arguments that SIP early media in a 180 is not compliant with 3264 are just wrong.
>
> We've *already* "relaxed" 3264's requirements in JSEP (comments apply to draft-ietf-rtcweb-jsep-01)
> 	
> Straight from 3264: "At any time, either agent MAY generate a new offer that updates the session. However, it MUST NOT generate a new offer if it has received an offer which it has not yet answered or rejected. Furthermore, it MUST NOT generate a new offer if it has generated a prior offer for which it has not yet received an answer or a rejection. If an agent receives an offer after having sent one, but before receiving an answer to it, this is considered a "glare" condition."
>
> The JSEP API enforces no such rules. If you want to change the API to bring it into compliance with *just this section* of 3264, it would need all sorts of changes...
>
> - 5.1.1 createOffer ... "Calling this method does not change state; its use is not required". Ok, so how then is JSEP to enforce the two "MUST NOT generate a new offer" is the quote from 3264 above? I would think that createOffer must return an error whenever the "MUST NOT" cases apply. (Never mind that if the "use is not required" then somehow I'm to guess what SDP is legal for this browser to pass in to setLocalDescription as an offer?)
>
> - 5.1.4 setLocalDescription... has no language indicating that it is prohibited to call "setLocalDescription" with an offer and then call it again with a different offer without having supplied an answer. There also appears to be no way to pass a "reject" in. (These two issues appear in the W3C API state description too, so appears to really be what we mean when we say that we're going to "use JSEP")
>
> - 5.1.2 createAnswer... "Calling this method does not change state; its use is not required". So then how can setLocalDescription know when it is forbidden to accept an "offer" passed to it (as createOffer is not required) when setRemoteDescription has been passed an "offer" but "createAnswer" has not yet been called?
>
> And so on.
>
> This isn't the only section of 3264 that's gone out the window with JSEP, but saying "I have not seen any reason to relax 3264 yet" doesn't make any sense... that ship sailed when JSEP was proposed.
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From christer.holmberg@ericsson.com  Wed Oct 17 23:30:40 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>, Justin Uberti <juberti@google.com>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Hey folks,

On 18.10.12, 08:30, Christer Holmberg wrote:
> On 17.10.12, 22:40, Matthew Kaufman wrote:
>>=20
>> 4) How does trickle ICE work without =E2=80=9Crelaxing=E2=80=9D RFC326=
4 O/A? It
>> seems like you really want to be able to trickle via updated offers
>> that may be generated prior to the corresponding answer or reject?
>=20
> One of my comments on the trickle draft was about that. The draft
> says that a new offer can be sent "at any time", but my comment was
> that it should be according to 3264.

That's not exactly what the draft says. What it does say is:

  At any point of ICE processing, a trickle ICE agent may receive new
  candidates from the remote agent.

> IF we are going to relax 3264 (I really hope we are NOT), it needs to
> be clearly described somewhere. We cannot have a number of I-Ds doing
> it "on the run"...

I don't see how trickle ICE would require any changes to the O/A model.
Candidate trickling semantics are completely separate from those in 3264.=


Yes, the 3264 offer may, in some cases, contain a first batch of
candidates and the the 3264 may have to be delayed until ICE processing
yields valid pairs for every component but that's about it.

Am I missing something?

Emil


--=20
https://jitsi.org


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Hi,

I agree with Matthew.

In my opinion, the default should be that JSEP O/A is aligned with RFC 3264=
.

Then, IF some relaxation is needed, we should look at it case by case. At t=
he moment, one has to try to figure out him/herself what the relaxations ar=
e.

 And, as the authors don't seem to think there are any relaxations in the f=
irst place, I REALLY think we need to clarify this :)

And, it's not only about PRANSWER - even without that there are relaxations=
 (as Matthew's e-mail describes).

Regards,

Christer





-----Original Message-----
From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]=20
Sent: 17. lokakuuta 2012 23:25
To: Cullen Jennings (fluffy); Christer Holmberg
Cc: rtcweb@ietf.org
Subject: Relaxing SDP O/A (was RE: [rtcweb] Agenda requests for Atlanta mee=
ting)

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings (fluffy)
Sent: Tuesday, October 9, 2012 7:17 AM

> I have not seen any reason to relax 3264 yet but if something comes up, a=
gree we should carefully look at the cases. I think we can just do straight=
 up > 3264. Arguments that SIP early media in a 180 is not compliant with 3=
264 are just wrong.

We've *already* "relaxed" 3264's requirements in JSEP (comments apply to dr=
aft-ietf-rtcweb-jsep-01)
=09
Straight from 3264: "At any time, either agent MAY generate a new offer tha=
t updates the session. However, it MUST NOT generate a new offer if it has =
received an offer which it has not yet answered or rejected. Furthermore, i=
t MUST NOT generate a new offer if it has generated a prior offer for which=
 it has not yet received an answer or a rejection. If an agent receives an =
offer after having sent one, but before receiving an answer to it, this is =
considered a "glare" condition."

The JSEP API enforces no such rules. If you want to change the API to bring=
 it into compliance with *just this section* of 3264, it would need all sor=
ts of changes...

- 5.1.1 createOffer ... "Calling this method does not change state; its use=
 is not required". Ok, so how then is JSEP to enforce the two "MUST NOT gen=
erate a new offer" is the quote from 3264 above? I would think that createO=
ffer must return an error whenever the "MUST NOT" cases apply. (Never mind =
that if the "use is not required" then somehow I'm to guess what SDP is leg=
al for this browser to pass in to setLocalDescription as an offer?)

- 5.1.4 setLocalDescription... has no language indicating that it is prohib=
ited to call "setLocalDescription" with an offer and then call it again wit=
h a different offer without having supplied an answer. There also appears t=
o be no way to pass a "reject" in. (These two issues appear in the W3C API =
state description too, so appears to really be what we mean when we say tha=
t we're going to "use JSEP")

- 5.1.2 createAnswer... "Calling this method does not change state; its use=
 is not required". So then how can setLocalDescription know when it is forb=
idden to accept an "offer" passed to it (as createOffer is not required) wh=
en setRemoteDescription has been passed an "offer" but "createAnswer" has n=
ot yet been called?

And so on.

This isn't the only section of 3264 that's gone out the window with JSEP, b=
ut saying "I have not seen any reason to relax 3264 yet" doesn't make any s=
ense... that ship sailed when JSEP was proposed.

Matthew Kaufman

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Date: Thu, 18 Oct 2012 14:21:02 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Hey Christer,

On 18.10.12, 13:17, Christer Holmberg wrote:
> Hi,
>=20
>>>> 4) How does trickle ICE work without =E2=80=9Crelaxing=E2=80=9D RFC3=
264 O/A? It
>>>> seems like you really want to be able to trickle via updated
>>>> offers that may be generated prior to the corresponding answer
>>>> or reject?
>>>=20
>>> One of my comments on the trickle draft was about that. The draft
>>> says that a new offer can be sent "at any time", but my comment
>>> was that it should be according to 3264.
>>=20
>> That's not exactly what the draft says. What it does say is:
>>=20
>> At any point of ICE processing, a trickle ICE agent may receive
>> new candidates from the remote agent.
>=20
> Correct.
>=20
> And, it is of course true in one sense, because STUN requests
> creating peer reflexive candidates can of course be received at any
> time :)

True but in the "Trickle ICE" case it would be more common to learn new
candidates through signalling (rather than PR candidates during conn
checks). This is what the text refers to. I agree we should make it
clearer that we do not really mean PR candidates here.

>>> IF we are going to relax 3264 (I really hope we are NOT), it
>>> needs to be clearly described somewhere. We cannot have a number
>>> of I-Ds doing it "on the run"...
>>=20
>> I don't see how trickle ICE would require any changes to the O/A
>> model. Candidate trickling semantics are completely separate from
>> those in 3264.
>>=20
>> Yes, the 3264 offer may, in some cases, contain a first batch of
>> candidates and the the 3264 may have to be delayed until ICE
>> processing yields valid pairs for every component but that's about
>> it.
>>=20
>> Am I missing something?
>=20
> I guess the question was whether one, after the first batch of
> candidates have been sent in an offer, should be allowed to send the
> second batch in a new offer - before an answer to the previous offer
> has been received. That would be against 3264.

It would indeed but I am not sure why we would think of additional
candidate drops as offers at all. They are just independent signalling
and are only loosely related to the 3264 semantics.

Of course with SIP we would have a problem caused by the fact that
additional in-dialog signalling is blocked by the 3264 answer. However,
that's specific to SIP and will probably be best served with a SIP
specific solution (e.g. UPDATEs or forcing early answers, or something
else).

Does this make sense?
Emil

> Regards,
>=20
> Christer
>=20

--=20
https://jitsi.org


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 18.10.12, 14:26, Christer Holmberg wrote:
> Hi,
>=20
>>>>>> 4) How does trickle ICE work without =E2=80=9Crelaxing=E2=80=9D RF=
C3264
>>>>>> O/A? It seems like you really want to be able to trickle
>>>>>> via updated offers that may be generated prior to the
>>>>>> corresponding answer or reject?
>>>>>=20
>>>>> One of my comments on the trickle draft was about that. The
>>>>> draft says that a new offer can be sent "at any time", but my
>>>>> comment was that it should be according to 3264.
>>>>=20
>>>> That's not exactly what the draft says. What it does say is:
>>>>=20
>>>> At any point of ICE processing, a trickle ICE agent may receive
>>>> new candidates from the remote agent.
>>>=20
>>> Correct.
>>>=20
>>> And, it is of course true in one sense, because STUN requests
>>> creating peer reflexive candidates can of course be received at
>>> any time :)
>>=20
>> True but in the "Trickle ICE" case it would be more common to learn
>> new candidates through signalling (rather than PR candidates during
>> conn checks). This is what the text refers to. I agree we should
>> make it clearer that we do not really mean PR candidates here.
>=20
> Sure.
>=20
> (Remember our previous discussion, though, about PR candidates being
> enough if the remote peer is ICE lite ;)

I do :).

>>>>> IF we are going to relax 3264 (I really hope we are NOT), it
>>>>> needs to be clearly described somewhere. We cannot have a
>>>>> number of I-Ds doing it "on the run"...
>>>>=20
>>>> I don't see how trickle ICE would require any changes to the
>>>> O/A model. Candidate trickling semantics are completely
>>>> separate from those in 3264.
>>>>=20
>>>> Yes, the 3264 offer may, in some cases, contain a first batch
>>>> of candidates and the the 3264 may have to be delayed until ICE
>>>>  processing yields valid pairs for every component but that's
>>>> about it.
>>>>=20
>>>> Am I missing something?
>>>=20
>>> I guess the question was whether one, after the first batch of=20
>>> candidates have been sent in an offer, should be allowed to send
>>> the second batch in a new offer - before an answer to the
>>> previous offer has been received. That would be against 3264.
>>=20
>> It would indeed but I am not sure why we would think of additional
>> candidate drops as offers at all. They are just independent
>> signalling and are only loosely related to the 3264 semantics.
>>=20
>> Of course with SIP we would have a problem caused by the fact that
>> additional in-dialog signalling is blocked by the 3264 answer.
>> However, that's specific to SIP and will probably be best served
>> with a SIP specific solution (e.g. UPDATEs or forcing early
>> answers, or something else).
>=20
> It is sure that SIP may add its own limitations, but the general O/A
> is generic.

Sure, and we agree that the general O/A need not be used for trickle
ICE, right?

Emil

--=20
https://jitsi.org


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Date: Thu, 18 Oct 2012 14:53:09 +0200
From: Emil Ivov <emcho@jitsi.org>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
References: <CABcZeBMzgAs=hK38hCjS7t6yLjkTydS2TQUb8R3rBbRKGakVdQ@mail.gmail.com> <50508ED7.9080805@ericsson.com> <505B6A7E.6010309@jitsi.org> <CAOJ7v-00BAp8M0_+FJGuXqAdXQ=e=MRN_L_6_CmkQWX-Gy5JAg@mail.gmail.com> <AE1A6B5FD507DC4FB3C5166F3A05A484160ED5CD@tk5ex14mbxc272.redmond.corp.microsoft.com> <7594FB04B1934943A5C02806D1A2204B018842@ESESSMB209.ericsson.se> <507FD1C8.8000100@jitsi.org> <7594FB04B1934943A5C02806D1A2204B018C70@ESESSMB209.ericsson.se> <507FF42E.5070106@jitsi.org> <7594FB04B1934943A5C02806D1A2204B018D1F@ESESSMB209.ericsson.se> <507FF688.2010704@jitsi.org> <7594FB04B1934943A5C02806D1A2204B018D4B@ESESSMB209.ericsson.se>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Hey Christer,

On 18.10.12, 14:34, Christer Holmberg wrote:
> Hi,
> 
>>>>>>> IF we are going to relax 3264 (I really hope we are NOT), it needs 
>>>>>>> to be clearly described somewhere. We cannot have a number of I-Ds 
>>>>>>> doing it "on the run"...
>>>>>>
>>>>>> I don't see how trickle ICE would require any changes to the O/A 
>>>>>> model. Candidate trickling semantics are completely separate from 
>>>>>> those in 3264.
>>>>>>
>>>>>> Yes, the 3264 offer may, in some cases, contain a first batch of 
>>>>>> candidates and the the 3264 may have to be delayed until ICE  
>>>>>> processing yields valid pairs for every component but that's about 
>>>>>> it.
>>>>>>
>>>>>> Am I missing something?
>>>>>
>>>>> I guess the question was whether one, after the first batch of 
>>>>> candidates have been sent in an offer, should be allowed to send the 
>>>>> second batch in a new offer - before an answer to the previous offer 
>>>>> has been received. That would be against 3264.
>>>>
>>>> It would indeed but I am not sure why we would think of additional 
>>>> candidate drops as offers at all. They are just independent 
>>>> signalling and are only loosely related to the 3264 semantics.
>>>>
>>>> Of course with SIP we would have a problem caused by the fact that 
>>>> additional in-dialog signalling is blocked by the 3264 answer.
>>>> However, that's specific to SIP and will probably be best served with 
>>>> a SIP specific solution (e.g. UPDATEs or forcing early answers, or 
>>>> something else).
>>>
>>> It is sure that SIP may add its own limitations, but the general O/A 
>>> is generic.
>>
>> Sure, and we agree that the general O/A need not be used for trickle ICE, right?
> 
> Well, I think general O/A SHALL be used - not only for trickle ICE,

But why? What do we get from trickling via offers and answers other than
problems?

Or did you mean that

> but also for JSEP :)
> 
> (Vanilla ICE is also using general O/A)

Not really. At least not always. ICE is quite nicely separated from the
media O/A in XMPP.

Both are indeed stuffed together for SIP by 5245 but that's just a
design choice that we don't need to stick with. At least I don't see why
we would.


Emil

-- 
https://jitsi.org

From christer.holmberg@ericsson.com  Thu Oct 18 06:03:31 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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On 10/16/2012 07:35 PM, Bernard Aboba wrote:
> Some chipsets do support temporal scaling (search 'soc support for H.264/SVC'), and there are sw implementations that support temporal, quality and spatial scaling ( http://www.polycom.com/company/news/press-releases/2012/20121004.html).
I did search for what you suggested, but what came up seemed to be 
software running on System-on-a-chip processors (eInfochips' C library 
from 2009, an IEEE Explore article detailing an implementation running 
on a Linux system with a DSP). Your search results may be different.

These may have satisfactory performance - using DSPs for video codec 
tasks kind of blurs the distinction between "hardware implementation" 
and "software implementation". But I guess that the more the processors 
are general purpose, the less it matters which particular codec you 
choose to implement on them - so they become irrelevant to whether a 
particular codec has "hardware support".

                  Harald


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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Hey Christer,

On 18.10.12, 15:03, Christer Holmberg wrote:
> Hi,
> 
>>>>>>>>> IF we are going to relax 3264 (I really hope we are NOT), it 
>>>>>>>>> needs to be clearly described somewhere. We cannot have a number 
>>>>>>>>> of I-Ds doing it "on the run"...
>>>>>>>>
>>>>>>>> I don't see how trickle ICE would require any changes to the O/A 
>>>>>>>> model. Candidate trickling semantics are completely separate from 
>>>>>>>> those in 3264.
>>>>>>>>
>>>>>>>> Yes, the 3264 offer may, in some cases, contain a first batch of 
>>>>>>>> candidates and the the 3264 may have to be delayed until ICE 
>>>>>>>> processing yields valid pairs for every component but that's 
>>>>>>>> about it.
>>>>>>>>
>>>>>>>> Am I missing something?
>>>>>>>
>>>>>>> I guess the question was whether one, after the first batch of 
>>>>>>> candidates have been sent in an offer, should be allowed to send 
>>>>>>> the second batch in a new offer - before an answer to the previous 
>>>>>>> offer has been received. That would be against 3264.
>>>>>>
>>>>>> It would indeed but I am not sure why we would think of additional 
>>>>>> candidate drops as offers at all. They are just independent 
>>>>>> signalling and are only loosely related to the 3264 semantics.
>>>>>>
>>>>>> Of course with SIP we would have a problem caused by the fact that 
>>>>>> additional in-dialog signalling is blocked by the 3264 answer.
>>>>>> However, that's specific to SIP and will probably be best served 
>>>>>> with a SIP specific solution (e.g. UPDATEs or forcing early 
>>>>>> answers, or something else).
>>>>>
>>>>> It is sure that SIP may add its own limitations, but the general O/A 
>>>>> is generic.
>>>>
>>>> Sure, and we agree that the general O/A need not be used for trickle ICE, right?
>>>
>>> Well, I think general O/A SHALL be used - not only for trickle ICE,
>>
>> But why? What do we get from trickling via offers and answers other than problems?
>>
>> Or did you mean that
>>
>>> but also for JSEP :)
>>>
>>> (Vanilla ICE is also using general O/A)
>>
>> Not really. At least not always. ICE is quite nicely separated from the media O/A in XMPP.
>>
>> Both are indeed stuffed together for SIP by 5245 but that's just a design choice that we don't need to stick with. At least I don't see why we would.
> 
> I was thinking about 5245, and JSEP, both with use 3264.
> 
> You are right that other protocols (e.g. XMPP) also can use ICE. However, you do use "Offer" and "Answer" terminology in your draft, and at least I associate that with 3264...

Yes, but I already agreed that the term was somewhat abused and we need
to review that. We may probably just switch to calling them ICE
initiation request/response ... or something along those lines.

We will do this in the 01 version which should be coming later this week.

Cheers,
Emil

-- 
https://jitsi.org

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On Wed, Oct 17, 2012 at 1:40 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:

> 2)      Are we seriously considering defining the allowed SDP for
> RTCWEB/WEBRTC by referencing an I-D and not an RFC, or are we going to delay
> defining the allowed SDP until trickle ICE is in an RFC, or are we going to
> drop trickle ICE from RTCWEB 1.0?
>

It's fairly normal for an I-D to reference other I-Ds spawned by the
work.  There's an obvious race condition, and a common set of
responses to that.  Typically, if the I-D making the reference is
ready to advance before the one it refers to is published, there's
discussion at that point just prior to IETF Last Call of whether the
reference is stable enough for a down reference.  If it is not,
there's a reference hold at the RFC publication stage (Bill Fenner has
produced some lovely/scary graphs of what can occur with these holds).

Basically, we don't need to decide which path to take now, but it's
clear that paying attention to the drafts in other working groups  is
a good thing for the advancement of our work.

regards,

Ted

From andrew.hutton@siemens-enterprise.com  Thu Oct 18 08:55:18 2012
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Thread-Topic: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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Also agree in general we need more clarity on SDP usage and it would be goo=
d if the default use of JSEP was to maintain compatibility with 3264 offer/=
answer but I assume it is really the application that has to ensure that.

I am assuming that jsep-02 will start to clarify some of this and align wit=
h the W3C spec which for example does require createOffer and createAnswer =
to be used.

Also wondering if jsep-02 will clarify the current thinking on how rehydrat=
ion is going to work I am thinking that in this case it would be has to be =
down to the application to do what it can to stick to O/A rules but there w=
ill be limitation.

I look forward to reviewing jsep-02 in detail when it is available.

Regards
Andy





> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: 18 October 2012 12:56
> To: Matthew Kaufman; Cullen Jennings (fluffy)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for
> Atlanta meeting)
>=20
> Hi,
>=20
> I agree with Matthew.
>=20
> In my opinion, the default should be that JSEP O/A is aligned with RFC
> 3264.
>=20
> Then, IF some relaxation is needed, we should look at it case by case.
> At the moment, one has to try to figure out him/herself what the
> relaxations are.
>=20
>  And, as the authors don't seem to think there are any relaxations in
> the first place, I REALLY think we need to clarify this :)
>=20
> And, it's not only about PRANSWER - even without that there are
> relaxations (as Matthew's e-mail describes).
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]
> Sent: 17. lokakuuta 2012 23:25
> To: Cullen Jennings (fluffy); Christer Holmberg
> Cc: rtcweb@ietf.org
> Subject: Relaxing SDP O/A (was RE: [rtcweb] Agenda requests for Atlanta
> meeting)
>=20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Cullen Jennings (fluffy)
> Sent: Tuesday, October 9, 2012 7:17 AM
>=20
> > I have not seen any reason to relax 3264 yet but if something comes
> up, agree we should carefully look at the cases. I think we can just do
> straight up > 3264. Arguments that SIP early media in a 180 is not
> compliant with 3264 are just wrong.
>=20
> We've *already* "relaxed" 3264's requirements in JSEP (comments apply
> to draft-ietf-rtcweb-jsep-01)
>=20
> Straight from 3264: "At any time, either agent MAY generate a new offer
> that updates the session. However, it MUST NOT generate a new offer if
> it has received an offer which it has not yet answered or rejected.
> Furthermore, it MUST NOT generate a new offer if it has generated a
> prior offer for which it has not yet received an answer or a rejection.
> If an agent receives an offer after having sent one, but before
> receiving an answer to it, this is considered a "glare" condition."
>=20
> The JSEP API enforces no such rules. If you want to change the API to
> bring it into compliance with *just this section* of 3264, it would
> need all sorts of changes...
>=20
> - 5.1.1 createOffer ... "Calling this method does not change state; its
> use is not required". Ok, so how then is JSEP to enforce the two "MUST
> NOT generate a new offer" is the quote from 3264 above? I would think
> that createOffer must return an error whenever the "MUST NOT" cases
> apply. (Never mind that if the "use is not required" then somehow I'm
> to guess what SDP is legal for this browser to pass in to
> setLocalDescription as an offer?)
>=20
> - 5.1.4 setLocalDescription... has no language indicating that it is
> prohibited to call "setLocalDescription" with an offer and then call it
> again with a different offer without having supplied an answer. There
> also appears to be no way to pass a "reject" in. (These two issues
> appear in the W3C API state description too, so appears to really be
> what we mean when we say that we're going to "use JSEP")
>=20
> - 5.1.2 createAnswer... "Calling this method does not change state; its
> use is not required". So then how can setLocalDescription know when it
> is forbidden to accept an "offer" passed to it (as createOffer is not
> required) when setRemoteDescription has been passed an "offer" but
> "createAnswer" has not yet been called?
>=20
> And so on.
>=20
> This isn't the only section of 3264 that's gone out the window with
> JSEP, but saying "I have not seen any reason to relax 3264 yet" doesn't
> make any sense... that ship sailed when JSEP was proposed.
>=20
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ibc@aliax.net  Thu Oct 18 09:19:55 2012
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Date: Thu, 18 Oct 2012 18:19:32 +0200
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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2012/10/18 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> I have been assuming that this is to be accomplished by the server pushin=
g
> javascript that is aware it is/may be interworking with sip, and which th=
en
> constrains itself to the subset of JSEP that compatible with 3264. And th=
en
> it is of course limited to those behaviors on the sip side that fall with=
in
> the subset of 3264 that JSEP supports.
>
> Is that the right way to view it?

Hi Paul, what about is the JavaScript app directly speaks SIP? please
let's also consider this case. Current web apps based on HTML5 and
modern JavaScript are powerful like any other desktop application. No
need to consider them limited and rely all the logic in the web server
(like 5 years ago).

Regards.


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

From andrew.hutton@siemens-enterprise.com  Thu Oct 18 09:26:17 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: draft-nandakumar-rtcweb-sdp-00 - Legacy Interop
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Subject: [rtcweb] draft-nandakumar-rtcweb-sdp-00 - Legacy Interop
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Hi,

Just looking at the legacy interop examples in this draft at http://tools.i=
etf.org/html/draft-nandakumar-rtcweb-sdp-00#section-6.=20

I know it states "The ideas included in here are not fully baked into the s=
tandards and might be controversial in nature" but don't they actually conf=
lict directly with existing decisions made?

For example section 6.1 implies that the browser could be made to generate =
an m line for RTP/AVP which it of course will not do. I know that actually =
this is trick because DTLS-SRTP is still used but does look very messy. Or =
do I misunderstand something?

I am wondering whether the authors envisage that this SDP is generated by t=
he browser or somehow handcrafted by the application but I assume the brows=
er would have to do this, correct?

Regards
Andy

From martin.thomson@gmail.com  Thu Oct 18 09:36:02 2012
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On 18 October 2012 09:19, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:
> 2012/10/18 Paul Kyzivat <pkyzivat@alum.mit.edu>:
>> I have been assuming that this is to be accomplished by the server pushi=
ng
>> javascript that is aware it is/may be interworking with sip, and which t=
hen
>> constrains itself to the subset of JSEP that compatible with 3264. And t=
hen
>> it is of course limited to those behaviors on the sip side that fall wit=
hin
>> the subset of 3264 that JSEP supports.
>>
>> Is that the right way to view it?
>
> Hi Paul, what about is the JavaScript app directly speaks SIP? please
> let's also consider this case. Current web apps based on HTML5 and
> modern JavaScript are powerful like any other desktop application. No
> need to consider them limited and rely all the logic in the web server
> (like 5 years ago).
I=C3=B1aki,

I think that is *exactly* what Paul said.  After all, the Javascript
has to come from somewhere.  That's usually a server.  The alternative
is that there is a server somewhere that does this work.

I find the distinction uninteresting.  Server or Javascript are both
"application" and where this code is executed matters very little.

--Martin

From martin.thomson@gmail.com  Thu Oct 18 09:47:45 2012
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Subject: [rtcweb] Comments on draft-miniero-rtcweb-http-fallback
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This seems grossly inefficient.  Even if you assume that we have to
tolerate the head-of-line blocking properties of a stream transport,
the overhead of HTTP headers is immense.

What is wrong with WebSockets for this use case?

Why is there so much discussion on topology?  It seems that the
topology used for TURN is perfectly good and would require the least
disruption.

From ibc@aliax.net  Thu Oct 18 09:49:46 2012
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2012/10/18 Martin Thomson <martin.thomson@gmail.com>:
> I=C3=B1aki,
>
> I think that is *exactly* what Paul said.  After all, the Javascript
> has to come from somewhere.  That's usually a server.  The alternative
> is that there is a server somewhere that does this work.

Sorry then, I understood something more similar to a "B2BUA" , my fault :)

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

From lorenzo@meetecho.com  Thu Oct 18 10:30:32 2012
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From martin.thomson@gmail.com  Thu Oct 18 10:47:43 2012
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On 18 October 2012 10:30, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg05035.html

I remember reading that, but considered that starting a new thread was
appropriate, that's all.

My view on the subject is that firewall circumvention is not and
should not be the goal of any such work.  A proper approach would be
to use a TURN server that manages access control in a transparent
fashion.

That said, firewalls often block UDP for reasons other than a deep and
abiding hatred of VoIP.  Those firewalls are probably not doing
anything that would prevent something like WebSockets from working.
In those environments, a standard method for using the available
communication channels could be helpful.  WebSockets is also far more
efficient with its use of bits.  Hence, WebSockets.

The main objection to something like WebSockets is that some
environments actively block WebSockets.  These firewalls often block
HTTPS in the same way.  In that case, I see no reason to continue to
build ever more elaborate circumvention methods.

I've said enough.  I just realized that this is not really an rtcweb
topic.  (behave?)

From worley@shell01.TheWorld.com  Thu Oct 18 11:18:04 2012
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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> From: Matthew Kaufman <matthew.kaufman@skype.net>
>
> The JSEP API enforces no such rules. If you want to change the API
> to bring it into compliance with *just this section* of 3264, it
> would need all sorts of changes...

I'm not familiar with the details of JSEP, but on general
architectural principles, my suspicion is that the key regarding
interworking with SIP is not whether a JSEP-using application can send
offers at times that the 3264 offer/answer model forbids it, but
rather whether JSEP/WebRTC can *reject* an offer.

Certainly in SIP, a new offer presented in a re-INVITE or UPDATE
request can be rejected by the other UA by returning a failure
response.  Any WebRTC protocol that can interwork smoothly with SIP
will require this capability.

Given that, if the JSEP model permits an application to send offers at
times when SIP does not, the WebRTC/SIP gateway/server can immediately
send rejection of such offers back to the client, without sending a
SIP request containing the offer.

This approach tolerates considerable difference between the rules
enforced by JSEP and the rules enforced by SIP, which may be
convenient.

Dale

From christer.holmberg@ericsson.com  Thu Oct 18 11:28:02 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "'Dale R. Worley'" <worley@ariadne.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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Hi,=20

>> The JSEP API enforces no such rules. If you want to change the API to=20
>> bring it into compliance with *just this section* of 3264, it would=20
>> need all sorts of changes...
>
>I'm not familiar with the details of JSEP, but on general architectural pr=
inciples, my suspicion is that the key regarding interworking with SIP is n=
ot whether a JSEP-using application can send offers at times that the 3264=
=20
>offer/answer model forbids it, but rather whether JSEP/WebRTC can *reject*=
 an offer.
>
>Certainly in SIP, a new offer presented in a re-INVITE or UPDATE request c=
an be rejected by the other UA by returning a failure response.  Any WebRTC=
 protocol that can interwork smoothly with SIP will require this capability=
.
>
>Given that, if the JSEP model permits an application to send offers at tim=
es when SIP does not, the WebRTC/SIP gateway/server can immediately send re=
jection of such offers back to the client, without sending a SIP request co=
ntaining=20
>the offer.

For sure it must be possible to reject things which are not allowed by the =
state machnine.

The issue, however, is that the JSUP SDP O/A state machine currently does a=
llow things which are not alligned with 3264.

Regards,

Christer


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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/18/2012 08:17 PM, Dale R. Worley wrote:
>> From: Matthew Kaufman <matthew.kaufman@skype.net>
>>
>> The JSEP API enforces no such rules. If you want to change the API
>> to bring it into compliance with *just this section* of 3264, it
>> would need all sorts of changes...
> I'm not familiar with the details of JSEP, but on general
> architectural principles, my suspicion is that the key regarding
> interworking with SIP is not whether a JSEP-using application can send
> offers at times that the 3264 offer/answer model forbids it, but
> rather whether JSEP/WebRTC can *reject* an offer.
>
> Certainly in SIP, a new offer presented in a re-INVITE or UPDATE
> request can be rejected by the other UA by returning a failure
> response.  Any WebRTC protocol that can interwork smoothly with SIP
> will require this capability.
>
> Given that, if the JSEP model permits an application to send offers at
> times when SIP does not, the WebRTC/SIP gateway/server can immediately
> send rejection of such offers back to the client, without sending a
> SIP request containing the offer.
>
> This approach tolerates considerable difference between the rules
> enforced by JSEP and the rules enforced by SIP, which may be
> convenient.
Note that the JSEP API never sends anything. It permits the application 
to generate an offer (using CreateOffer), and it permits the application 
to tell the browser that it should set the local configuration according 
to an offer (using setLocal).

The assumption is that the normal way of using the API when interworking 
with SIP is to generate the offer, set it locally with SetLocal(), and 
send it as a SIP OFFER.

The API also offers signals (negotiationneeded) to tell the application 
that it would be good to change the offer it's sending; if the 
application intends to obey SIP rules, it will schedule that 
renegotiation after the current negotiation round finishes.

But the API never forces the application to send an offer; sending or 
not sending an offer is strictly the application's responsibility.


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 18 Oct 2012 20:58:23 +0200
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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2012/10/18 Harald Alvestrand <harald@alvestrand.no>:
> But the API never forces the application to send an offer; sending or not
> sending an offer is strictly the application's responsibility.

In other words, the API allows implementing SIP's SDP usage, or not.
That's up to the application.

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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Matthew Kaufman' <matthew.kaufman@skype.net>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Hi,=20

>Oh, I agree wholeheartedly here... in order to support things like trickle=
 ICE it would be enormously helpful to separate the ICE negotiation=20
>from the SDP O/A state machine, so that candidates may be added without ma=
king them part of an "Offer".
>
>The "Microsoft API proposal" neatly handles the ICE negotiation in a way t=
hat is separated from the media negotiation... I suggest we use that=20
>method instead of trying to bend SDP O/A all out of shape in order to do t=
rickle ICE.

Sure, between the browser and the JS App we might choose to separate ICE fr=
om SDP and O/A completely.

However, the way the trickle-ICE draft is written, I assume it focuses on t=
he interactions between two entities, and what is sent on the wire.

Regards,

Christer




________________________________________
From: Emil Ivov [emcho@jitsi.org]
Sent: Thursday, October 18, 2012 5:53 AM
To: Christer Holmberg
Cc: Matthew Kaufman; Justin Uberti; rtcweb@ietf.org
Subject: Re: [rtcweb] Filling in details on "trickle ICE"

Hey Christer,

On 18.10.12, 14:34, Christer Holmberg wrote:
> Hi,
>
>>>>>>> IF we are going to relax 3264 (I really hope we are NOT), it=20
>>>>>>> needs to be clearly described somewhere. We cannot have a number=20
>>>>>>> of I-Ds doing it "on the run"...
>>>>>>
>>>>>> I don't see how trickle ICE would require any changes to the O/A=20
>>>>>> model. Candidate trickling semantics are completely separate from=20
>>>>>> those in 3264.
>>>>>>
>>>>>> Yes, the 3264 offer may, in some cases, contain a first batch of=20
>>>>>> candidates and the the 3264 may have to be delayed until ICE=20
>>>>>> processing yields valid pairs for every component but that's=20
>>>>>> about it.
>>>>>>
>>>>>> Am I missing something?
>>>>>
>>>>> I guess the question was whether one, after the first batch of=20
>>>>> candidates have been sent in an offer, should be allowed to send=20
>>>>> the second batch in a new offer - before an answer to the previous=20
>>>>> offer has been received. That would be against 3264.
>>>>
>>>> It would indeed but I am not sure why we would think of additional=20
>>>> candidate drops as offers at all. They are just independent=20
>>>> signalling and are only loosely related to the 3264 semantics.
>>>>
>>>> Of course with SIP we would have a problem caused by the fact that=20
>>>> additional in-dialog signalling is blocked by the 3264 answer.
>>>> However, that's specific to SIP and will probably be best served=20
>>>> with a SIP specific solution (e.g. UPDATEs or forcing early=20
>>>> answers, or something else).
>>>
>>> It is sure that SIP may add its own limitations, but the general O/A=20
>>> is generic.
>>
>> Sure, and we agree that the general O/A need not be used for trickle ICE=
, right?
>
> Well, I think general O/A SHALL be used - not only for trickle ICE,

But why? What do we get from trickling via offers and answers other than pr=
oblems?

Or did you mean that

> but also for JSEP :)
>
> (Vanilla ICE is also using general O/A)

Not really. At least not always. ICE is quite nicely separated from the med=
ia O/A in XMPP.

Both are indeed stuffed together for SIP by 5245 but that's just a design c=
hoice that we don't need to stick with. At least I don't see why we would.


Emil

--
https://jitsi.org


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From: worley@ariadne.com (Dale R. Worley)
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> From: Harald Alvestrand <harald@alvestrand.no>
> 
> But the API never forces the application to send an offer; sending or 
> not sending an offer is strictly the application's responsibility.

Is the API able to report that the far end has rejected the offer?

Dale

From ekr@rtfm.com  Thu Oct 18 14:22:02 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On Thu, Oct 18, 2012 at 2:10 PM, Dale R. Worley <worley@ariadne.com> wrote:
>> From: Harald Alvestrand <harald@alvestrand.no>
>>
>> But the API never forces the application to send an offer; sending or
>> not sending an offer is strictly the application's responsibility.
>
> Is the API able to report that the far end has rejected the offer?

No. It's able to report that the local end couldn't negotiate but
learning about the far end depends on messaging...

-Ekr

From martin.thomson@gmail.com  Thu Oct 18 14:28:38 2012
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To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 18 October 2012 14:21, Eric Rescorla <ekr@rtfm.com> wrote:
> On Thu, Oct 18, 2012 at 2:10 PM, Dale R. Worley <worley@ariadne.com> wrote:
>> Is the API able to report that the far end has rejected the offer?
>
> No. It's able to report that the local end couldn't negotiate but
> learning about the far end depends on messaging...

That doesn't quite answer the question.

I think that the question is, if I setLocalDescription(offer), but the
remote peer rejects this offer, what do I do to (as 3264 dictates)
revert the session to the state it was in prior to "making the offer".

From ekr@rtfm.com  Thu Oct 18 14:56:16 2012
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On Thu, Oct 18, 2012 at 2:28 PM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 18 October 2012 14:21, Eric Rescorla <ekr@rtfm.com> wrote:
>> On Thu, Oct 18, 2012 at 2:10 PM, Dale R. Worley <worley@ariadne.com> wrote:
>>> Is the API able to report that the far end has rejected the offer?
>>
>> No. It's able to report that the local end couldn't negotiate but
>> learning about the far end depends on messaging...
>
> That doesn't quite answer the question.
>
> I think that the question is, if I setLocalDescription(offer), but the
> remote peer rejects this offer, what do I do to (as 3264 dictates)
> revert the session to the state it was in prior to "making the offer".

I assume you mean in an update scenario, since in the initial offer
scenario, you do pc.close();

In the update scenario it seems to me that with the current
API, the natural gesture is setRemoteDescription() with the previous
answer. But obviously the world would be better if this were written
down.

-Ekr

From martin.thomson@gmail.com  Thu Oct 18 15:11:00 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 18 October 2012 14:55, Eric Rescorla <ekr@rtfm.com> wrote:
> I assume you mean in an update scenario, since in the initial offer
> scenario, you do pc.close();
>
> In the update scenario it seems to me that with the current
> API, the natural gesture is setRemoteDescription() with the previous
> answer. But obviously the world would be better if this were written
> down.

Or maybe setLocalDescription + setRemoteDescription with the entirety
of the previous session state.

From roman@telurix.com  Thu Oct 18 15:22:29 2012
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From: Roman Shpount <roman@telurix.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On Thu, Oct 18, 2012 at 5:55 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> > I think that the question is, if I setLocalDescription(offer), but the
> > remote peer rejects this offer, what do I do to (as 3264 dictates)
> > revert the session to the state it was in prior to "making the offer".
>
> I assume you mean in an update scenario, since in the initial offer
> scenario, you do pc.close();
>
> In the update scenario it seems to me that with the current
> API, the natural gesture is setRemoteDescription() with the previous
> answer. But obviously the world would be better if this were written
> down.
>
>
In case of the session update scenario where update was refused (ie
something similar to re-INVITE failure), would not giving the previous
answer cause ICE and DTLS negotiations to run again or do we assume that
browser O/A logic is smart enough to know that since the SDP version is the
same it should be ignored?

Also, description that we passed in setLocalDescription might be very
different from the description that was passed in the
previous setLocalDescription call (for instance we added or removed video).
Would sending the previous answer in setRemoteDescription still work?

I would feel better if we had some sort of cancel offer command, which can
be just setRemoteDescription with an empty string instead of SDP.
_____________
Roman Shpount

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<br clear=3D"all"><br><div class=3D"gmail_quote">On Thu, Oct 18, 2012 at 5:=
55 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" =
target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
&gt; I think that the question is, if I setLocalDescription(offer), but the=
<br>
&gt; remote peer rejects this offer, what do I do to (as 3264 dictates)<br>
&gt; revert the session to the state it was in prior to &quot;making the of=
fer&quot;.<br>
<br>
I assume you mean in an update scenario, since in the initial offer<br>
scenario, you do pc.close();<br>
<br>
In the update scenario it seems to me that with the current<br>
API, the natural gesture is setRemoteDescription() with the previous<br>
answer. But obviously the world would be better if this were written<br>
down.<br>
<br></blockquote><div><br></div><div>In case of the session update scenario=
 where update was refused (ie something similar to re-INVITE failure), woul=
d not giving the previous answer cause ICE and DTLS negotiations to run aga=
in or do we assume that browser O/A logic is smart enough to know that sinc=
e the SDP version is the same it should be ignored?</div>
<div><br></div><div>Also, description that we passed in setLocalDescription=
 might be very different from the description that was passed in the previo=
us=A0setLocalDescription=A0call (for instance we added or removed video). W=
ould sending the previous answer in setRemoteDescription still work?</div>
<div><br></div><div>I would feel better if we had some sort of cancel offer=
 command, which can be just setRemoteDescription with an empty string inste=
ad of SDP.</div>_____________<br>Roman Shpount<br><br><div>=A0</div></div>

--047d7b2e79886b7fb904cc5cd232--

From martin.thomson@gmail.com  Thu Oct 18 17:23:54 2012
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [rtcweb] Change ice ufrag or password
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-jsep-01 currently includes the following option:
    - change ICE ufrag/password (a=ice-ufrag/pwd)

That's a no-go [-security-arch, S5.3].  Will this be removed from the
next draft?

I'm interested though, was there a specific case that motivated the
inclusion of this option?

From bernard_aboba@hotmail.com  Thu Oct 18 19:48:08 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta	meeting)
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It seems to me that some scenarios would definitely require both APIs=2C fo=
r example if hold or key changes or codec additions are involved.

Martin Thomson <martin.thomson@gmail.com> wrote:

On 18 October 2012 14:55=2C Eric Rescorla <ekr@rtfm.com> wrote:
> I assume you mean in an update scenario=2C since in the initial offer
> scenario=2C you do pc.close()=3B
>
> In the update scenario it seems to me that with the current
> API=2C the natural gesture is setRemoteDescription() with the previous
> answer. But obviously the world would be better if this were written
> down.

Or maybe setLocalDescription + setRemoteDescription with the entirety
of the previous session state.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From harald@alvestrand.no  Thu Oct 18 23:23:42 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/18/2012 11:10 PM, Dale R. Worley wrote:
>> From: Harald Alvestrand <harald@alvestrand.no>
>>
>> But the API never forces the application to send an offer; sending or
>> not sending an offer is strictly the application's responsibility.
> Is the API able to report that the far end has rejected the offer?
The concept of "the far end" (application far end, not RTP far end) and 
"reject" are application-level concepts, not API-level concepts.
So in the context of the API, I can't parse the question.

The API can report that it was handed an SDP blob it couldn't use.
The API can also report that the ICE state ended up in "failed" because 
communication didn't work.

But figuring out that a 500 error code in the SDP session (or whatever 
other method the signalling protocol uses to say "no, I don't want to 
talk to you") means "offer rejectd" has to be done at the application.

The application then has to tell the API not to bother with this 
PeerConnection any more - probably by calling close() or by deleting its 
references to the PeerConnection, so that it's garbage collected.


From stefan.lk.hakansson@ericsson.com  Thu Oct 18 23:37:01 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/19/2012 08:23 AM, Harald Alvestrand wrote:
> On 10/18/2012 11:10 PM, Dale R. Worley wrote:
>>> From: Harald Alvestrand <harald@alvestrand.no>
>>>
>>> But the API never forces the application to send an offer; sending or
>>> not sending an offer is strictly the application's responsibility.
>> Is the API able to report that the far end has rejected the offer?
> The concept of "the far end" (application far end, not RTP far end) and
> "reject" are application-level concepts, not API-level concepts.
> So in the context of the API, I can't parse the question.
>
> The API can report that it was handed an SDP blob it couldn't use.

One discussion I think we've not had yet is the granularity the API 
could report "can't use this SDP blob" on. Say that a new offer (in on 
ongoing sessions) was received indicating the addition of two audio and 
two video tracks (from Alice to Bob). What if Bob's browser could decode 
both audio tracks but has resources to handle only one of the video 
tracks, should it be able to tell so (perhaps with the result that two 
audio and one video track was added)? Or should the update of the 
session just fail?




From salvatore.loreto@ericsson.com  Fri Oct 19 00:17:56 2012
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] Comments on draft-miniero-rtcweb-http-fallback
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On 10/18/12 8:47 PM, Martin Thomson wrote:
> On 18 October 2012 10:30, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg05035.html
> I remember reading that, but considered that starting a new thread was
> appropriate, that's all.
>
> My view on the subject is that firewall circumvention is not and
> should not be the goal of any such work.  A proper approach would be
> to use a TURN server that manages access control in a transparent
> fashion.
>
> That said, firewalls often block UDP for reasons other than a deep and
> abiding hatred of VoIP.  Those firewalls are probably not doing
> anything that would prevent something like WebSockets from working.
> In those environments, a standard method for using the available
> communication channels could be helpful.  WebSockets is also far more
> efficient with its use of bits.  Hence, WebSockets.
>
> The main objection to something like WebSockets is that some
> environments actively block WebSockets.  These firewalls often block
> HTTPS in the same way.
that is interesting,
I have heard a lot people saying that usually the firewalls
(well some time they talk of the proxy really more then firewall)
are more likely to block WebSocket then HTTP

and they use that statement to promote an HTTP fallback
(here I am talking in general not specifically for webrtc)

However if we are talking of secure WebSocket and https fallback
I agree that if a firewall block secure Websocket then it will most 
likely block
HTTP as well !

>   In that case, I see no reason to continue to
> build ever more elaborate circumvention methods.
then I tend to agree in not trying to define a fallback of the fallback
>
> I've said enough.  I just realized that this is not really an rtcweb
> topic.  (behave?)
I would say that is rtcweb until we do not define what we want to do

ciao
Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 19 Oct 2012 10:19:27 +0200
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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2012/10/19 Harald Alvestrand <harald@alvestrand.no>:
> But figuring out that a 500 error code in the SDP session (or whatever ot=
her
> method the signalling protocol uses to say "no, I don't want to talk to
> you") means "offer rejectd" has to be done at the application.

Right. SDP must be signaled by the application making use of it. SDP
is not a "end-to-end" protocol, it must be carried somehow (via a
signaling protocol in which the remote endpoint can reject the SDP
offer with some signaling code that the local endpoint must understand
for closing/aborting its SDP offer).

SIP and XMPP are signaling protocols using SDP and fit very well into
the state machine defined in RFC 3264. In WebRTC we DON'T have a
(network) signaling protocol and, thus, it's the JavaScript
responsability to respect or not RFC 3264 rules, but those rules
cannot be mandated at the media stack level.

So honestly I don't fully understand what we are discussing in this
thread. Regards.


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

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Subject: [rtcweb] Handling a partially-useful offer (Re: Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting))
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Forking thread, since this is a new topic....

On 10/19/2012 08:36 AM, Stefan Hakansson LK wrote:
> On 10/19/2012 08:23 AM, Harald Alvestrand wrote:
>> On 10/18/2012 11:10 PM, Dale R. Worley wrote:
>>>> From: Harald Alvestrand <harald@alvestrand.no>
>>>>
>>>> But the API never forces the application to send an offer; sending or
>>>> not sending an offer is strictly the application's responsibility.
>>> Is the API able to report that the far end has rejected the offer?
>> The concept of "the far end" (application far end, not RTP far end) and
>> "reject" are application-level concepts, not API-level concepts.
>> So in the context of the API, I can't parse the question.
>>
>> The API can report that it was handed an SDP blob it couldn't use.
>
> One discussion I think we've not had yet is the granularity the API 
> could report "can't use this SDP blob" on. Say that a new offer (in on 
> ongoing sessions) was received indicating the addition of two audio 
> and two video tracks (from Alice to Bob). What if Bob's browser could 
> decode both audio tracks but has resources to handle only one of the 
> video tracks, should it be able to tell so (perhaps with the result 
> that two audio and one video track was added)? Or should the update of 
> the session just fail?

This seems to be something that should be happening to SIP endpoints too.

I can imagine either answering with capabilities reduced to reflect what 
could be handled, accepting the offer followed by generating a new offer 
of reduced capabilities, or rejecting the offer.

What is existing practice in this area? (I would not be surprised if it 
was "it depends"....)


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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/18/2012 09:08 PM, Matthew Kaufman wrote:
> I think this interpretation is correct... which of course leads to the same question I asked on one of the last W3C calls on the subject: *IF* we are going to be producing an API which does not in fact comply with RFC 3264, but in fact significantly relaxes the rules of SDP O/A (which this email thread has confirmed is the case, as I suspected it would), then why are we so hung up on producing an API which looks anything like it (as opposed to, say, the proposal I jointly made with others here at Microsoft)?
Because making an interface that makes following an existing practice 
easy (but does not enforce all the constraints of that practice) is 
better than making an interface that makes following that practice very 
complex?


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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2012/10/19 Hutton, Andrew <andrew.hutton@siemens-enterprise.com>:
> RFC3624 requires that the session reverts to the state prior to the offer=
 if the offer/answer exchange fails.
>
> If an applications attempts to modify a session but the updated offer is =
rejected by the far end there should be no interruption to the existing ses=
sion. Like Roman suggested I am also thinking it would be nice if the API p=
rovided some way of cancelling an offer and reverting to the original state=
 but if this is putting too much state back in the browser we would need a =
clearly documented mechanism.

Ok, I understand. Then I agree that the API should provide some
mechanism for respecting that rule in RFC 3264.

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

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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 19 October 2012 07:37 Stefan Hakansson LK wrote:
>=20
> One discussion I think we've not had yet is the granularity the API
> could report "can't use this SDP blob" on. Say that a new offer (in on
> ongoing sessions) was received indicating the addition of two audio and
> two video tracks (from Alice to Bob). What if Bob's browser could
> decode
> both audio tracks but has resources to handle only one of the video
> tracks, should it be able to tell so (perhaps with the result that two
> audio and one video track was added)? Or should the update of the
> session just fail?
>=20

The update to the session should not just fail. Normally in this case the a=
nswer would simple reject the one video stream and not the whole offer. =20

Andy

From andrew.hutton@siemens-enterprise.com  Fri Oct 19 06:58:51 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Video Codec - Possible to use.
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Subject: [rtcweb] Video Codec - Possible to use.
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Hi,

The question of whether it will be possible for a webrtc application to mak=
e us of video codec's (or audio codec) which are not implemented in the bro=
wser itself has been raised before but I don't think there has been a clear=
 answer.

Some clarity on this might help some of us to come to a conclusion on where=
 we stand in the MTI debate.

I looked through the archives and found some statements that indicate this =
should be possible but I not sure we have a definitive statement on this. F=
or example:

On 06 September 2012 00:29 Randall Gellens
>=20
> One of the goals of rtcweb is to encourage greater use of native
> codecs, and greater access to such codecs by applications such as
> those running in browsers.  These are worthy goals, since native
> codecs often have better performance within the environment.
> (Examples include AMR and EVRC on handsets.)  These codecs also can
> be supported out-of-the-box, no separate downloads, no signed forms.
>

Is this really a goal we have agreement on? There does not seem to specific=
 requirements in the requirements draft.

Also I think there are SDP implications relating to this. How would the bro=
wser and/or application be able to build the correct SDP to offer additiona=
l codecs?

Regards
Andy=20

From stefan.lk.hakansson@ericsson.com  Fri Oct 19 07:22:36 2012
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Subject: Re: [rtcweb] Video Codec - Possible to use.
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On 10/19/2012 03:58 PM, Hutton, Andrew wrote:

>
> Also I think there are SDP implications relating to this. How would
> the browser and/or application be able to build the correct SDP to
> offer additional codecs?

To me this is a "must". Regardless of what codecs are made MTI now, they 
are likely to not be state-of-art in a few years time.

I had the impression this was one of the main advantages with using SDP: 
it offers a well established way to negotiate what codec(s) to use 
between end-points.

Br,
Stefan


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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Video Codec - Possible to use.
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I don't really understand the question.

Obviously, it should be possible for WebRTC implementations (e.g., browsers)
to derive codecs from the underlying platform (just as some browsers get
their libc or their crypto libraries from the underlying platform). Similarly,
it should be possible for browsers to allow pluggable additional codecs.
In both cases these seem like features browser implementors might
choose to provide at their discretion and which wouldn't be visible to
Web content.

Did you have something else in mind?

-Ekr


On Fri, Oct 19, 2012 at 6:58 AM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> Hi,
>
> The question of whether it will be possible for a webrtc application to make us of video codec's (or audio codec) which are not implemented in the browser itself has been raised before but I don't think there has been a clear answer.
>
> Some clarity on this might help some of us to come to a conclusion on where we stand in the MTI debate.
>
> I looked through the archives and found some statements that indicate this should be possible but I not sure we have a definitive statement on this. For example:
>
> On 06 September 2012 00:29 Randall Gellens
>>
>> One of the goals of rtcweb is to encourage greater use of native
>> codecs, and greater access to such codecs by applications such as
>> those running in browsers.  These are worthy goals, since native
>> codecs often have better performance within the environment.
>> (Examples include AMR and EVRC on handsets.)  These codecs also can
>> be supported out-of-the-box, no separate downloads, no signed forms.
>>
>
> Is this really a goal we have agreement on? There does not seem to specific requirements in the requirements draft.
>
> Also I think there are SDP implications relating to this. How would the browser and/or application be able to build the correct SDP to offer additional codecs?
>
> Regards
> Andy
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From pkyzivat@alum.mit.edu  Fri Oct 19 08:02:23 2012
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Subject: Re: [rtcweb] Handling a partially-useful offer (Re: Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting))
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On 10/19/12 5:23 AM, Harald Alvestrand wrote:
> Forking thread, since this is a new topic....
>
> On 10/19/2012 08:36 AM, Stefan Hakansson LK wrote:
>> On 10/19/2012 08:23 AM, Harald Alvestrand wrote:
>>> On 10/18/2012 11:10 PM, Dale R. Worley wrote:
>>>>> From: Harald Alvestrand <harald@alvestrand.no>
>>>>>
>>>>> But the API never forces the application to send an offer; sending or
>>>>> not sending an offer is strictly the application's responsibility.
>>>> Is the API able to report that the far end has rejected the offer?
>>> The concept of "the far end" (application far end, not RTP far end) and
>>> "reject" are application-level concepts, not API-level concepts.
>>> So in the context of the API, I can't parse the question.
>>>
>>> The API can report that it was handed an SDP blob it couldn't use.
>>
>> One discussion I think we've not had yet is the granularity the API
>> could report "can't use this SDP blob" on. Say that a new offer (in on
>> ongoing sessions) was received indicating the addition of two audio
>> and two video tracks (from Alice to Bob). What if Bob's browser could
>> decode both audio tracks but has resources to handle only one of the
>> video tracks, should it be able to tell so (perhaps with the result
>> that two audio and one video track was added)? Or should the update of
>> the session just fail?
>
> This seems to be something that should be happening to SIP endpoints too.
>
> I can imagine either answering with capabilities reduced to reflect what
> could be handled, accepting the offer followed by generating a new offer
> of reduced capabilities, or rejecting the offer.
>
> What is existing practice in this area? (I would not be surprised if it
> was "it depends"....)

Andy just talked about this.

The existing practice in sip is that the answerer will accept that part 
of the offer that it understands and is willing to use, and refuses the 
rest - by setting the port number to zero in the m-lines it can't/won't 
support.

	Thanks,
	Paul

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On: 19 October 2012 15:23 Stefan Hakansson LK wrote:
> On 10/19/2012 03:58 PM, Hutton, Andrew wrote:
>=20
> >
> > Also I think there are SDP implications relating to this. How would
> > the browser and/or application be able to build the correct SDP to
> > offer additional codecs?
>=20
> To me this is a "must". Regardless of what codecs are made MTI now,
> they
> are likely to not be state-of-art in a few years time.

That is not really the issue. Of course the browser can add support for new=
 codec's in years to come the question is whether the browser implementatio=
n will be able to make use of codec's provided by the underlying platform o=
r plugin.

>=20
> I had the impression this was one of the main advantages with using
> SDP:
> it offers a well established way to negotiate what codec(s) to use
> between end-points.
>=20

SDP of course is flexible on this but will the browsers SDP implementation =
be able to handle this?


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

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On 19 October 2012 15:58 Eric Rescorla wrote:
>=20
> I don't really understand the question.
>=20
> Obviously, it should be possible for WebRTC implementations (e.g.,
> browsers)
> to derive codecs from the underlying platform (just as some browsers
> get
> their libc or their crypto libraries from the underlying platform).
> Similarly,
> it should be possible for browsers to allow pluggable additional
> codecs.
> In both cases these seem like features browser implementors might
> choose to provide at their discretion and which wouldn't be visible to
> Web content.
>=20
> Did you have something else in mind?

Sorry if it is a silly question I am not so familiar with the mechanism her=
e.

What concerns me is that I hear it stated that using additional codec's fro=
m the underlying platform is possible but can someone tell me how the corre=
ct SDP would be generated? Would the browser have the information to build =
the SDP for codec's it did not natively support or would the application ne=
ed to craft the SDP and then if the application did it would the browser be=
 able to do the right thing when the app calls setLocalDescription() etc. =
=20

Andy


>=20
> -Ekr
>=20
>=20
> On Fri, Oct 19, 2012 at 6:58 AM, Hutton, Andrew
> <andrew.hutton@siemens-enterprise.com> wrote:
> > Hi,
> >
> > The question of whether it will be possible for a webrtc application
> to make us of video codec's (or audio codec) which are not implemented
> in the browser itself has been raised before but I don't think there
> has been a clear answer.
> >
> > Some clarity on this might help some of us to come to a conclusion on
> where we stand in the MTI debate.
> >
> > I looked through the archives and found some statements that indicate
> this should be possible but I not sure we have a definitive statement
> on this. For example:
> >
> > On 06 September 2012 00:29 Randall Gellens
> >>
> >> One of the goals of rtcweb is to encourage greater use of native
> >> codecs, and greater access to such codecs by applications such as
> >> those running in browsers.  These are worthy goals, since native
> >> codecs often have better performance within the environment.
> >> (Examples include AMR and EVRC on handsets.)  These codecs also can
> >> be supported out-of-the-box, no separate downloads, no signed forms.
> >>
> >
> > Is this really a goal we have agreement on? There does not seem to
> specific requirements in the requirements draft.
> >
> > Also I think there are SDP implications relating to this. How would
> the browser and/or application be able to build the correct SDP to
> offer additional codecs?
> >
> > Regards
> > Andy
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb

From ekr@rtfm.com  Fri Oct 19 08:29:40 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 19 Oct 2012 08:28:58 -0700
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Subject: Re: [rtcweb] Video Codec - Possible to use.
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On Fri, Oct 19, 2012 at 8:20 AM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> On 19 October 2012 15:58 Eric Rescorla wrote:
>>
>> I don't really understand the question.
>>
>> Obviously, it should be possible for WebRTC implementations (e.g.,
>> browsers)
>> to derive codecs from the underlying platform (just as some browsers
>> get
>> their libc or their crypto libraries from the underlying platform).
>> Similarly,
>> it should be possible for browsers to allow pluggable additional
>> codecs.
>> In both cases these seem like features browser implementors might
>> choose to provide at their discretion and which wouldn't be visible to
>> Web content.
>>
>> Did you have something else in mind?
>
> Sorry if it is a silly question I am not so familiar with the mechanism h=
ere.
>
> What concerns me is that I hear it stated that using additional codec's f=
rom the underlying platform is possible but can someone tell me how the cor=
rect SDP would be generated? Would the browser have the information to buil=
d the SDP for codec's it did not natively support or would the application =
need to craft the SDP and then if the application did it would the browser =
be able to do the right thing when the app calls setLocalDescription() etc.

So, obviously it depends on *exactly* what one means by the codec being
available on the underlying platform. There seem to be at least two cases:

1. This is some feature provided by the OS or platform vendor. In this
case presumably it comes with documentation and the browser implementor
reads the documentation and the associated RFCS and generates the
right SDP. In this case, the situation is not much different from that when
the codec implementation is part of the browser. (Note that in many cases
these codecs come from third party libraries in any case).

2. The browser supports some kind of plugin scheme which allows
people to download their own codecs and install them. This would require
the browser vendor to implement some interface for each codec to
get a crack at the SDP and is therefore a lot more work.

Which case are you concerned about?
-Ekr

From andrew.hutton@siemens-enterprise.com  Fri Oct 19 08:49:41 2012
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To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] Video Codec - Possible to use.
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On 19 October 2012 16:29 Eric Rescorla wrote:
> To: Hutton, Andrew
> So, obviously it depends on *exactly* what one means by the codec being
> available on the underlying platform. There seem to be at least two
> cases:
>=20
> 1. This is some feature provided by the OS or platform vendor. In this
> case presumably it comes with documentation and the browser implementor
> reads the documentation and the associated RFCS and generates the
> right SDP. In this case, the situation is not much different from that
> when
> the codec implementation is part of the browser. (Note that in many
> cases
> these codecs come from third party libraries in any case).
>=20
> 2. The browser supports some kind of plugin scheme which allows
> people to download their own codecs and install them. This would
> require
> the browser vendor to implement some interface for each codec to
> get a crack at the SDP and is therefore a lot more work.
>=20
> Which case are you concerned about?

Both and thanks your answer really helps my understanding and it seems it i=
s really up to the browser vendors and nothing the IETF or W3C can do to in=
fluence this.

So would I be correct in thinking that at least for the immediate future we=
brtc applications will be limited to the codec(s) that the browsers nativel=
y support or do browser vendors intend to at least implement option 1 above=
 for at least some commonly used video codec's?

Andy



From martin.thomson@gmail.com  Fri Oct 19 09:26:29 2012
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On 19 October 2012 04:55, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> On 19 October 2012 07:37 Stefan Hakansson LK wrote:
>> [...] What if Bob's browser could
>> decode
>> both  audio tracks but has resources to handle only one of the video
>> tracks, should it be able to tell so (perhaps with the result that two
>> audio and one video track was added)? Or should the update of the
>> session just fail?
>
> The update to the session should not just fail. Normally in this case the answer would simple reject the one video stream and not the whole offer.

A more interesting case is where Alice decides to (attempt to) switch
codecs.  That decision might even be based on some prior knowledge of
Bob's ability to understand codecs.  Maybe Bob originally offers A and
B, but when Alice decides to upgrade from A to B, that is not a mode
that Bob supports any more.  The session should continue with A.

An answer that rejected the stream would cause an interruption to the session.

> [...] I am also thinking it would be nice if the API provided some way of cancelling an offer and reverting to the original state but if this is putting too much state back in the browser we would need a clearly documented mechanism.

Nice or necessary?  There is already a great deal of state in the browser.

I am actually interested in a small modification to the API for this
case, for outstanding offers there are actually three pieces of SDP in
play: original local description (could be offer or answer), original
remote description (answer or offer) and the new outstanding offer
(which is only ever a local description, the action can be immediate
on the answering side).  We currently have a means to get two of
these: the remote description and (I assume) the local description
that was most recently set.  We do not have a way to get the original
local description.

From harald@alvestrand.no  Fri Oct 19 09:43:01 2012
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On 10/19/2012 06:06 PM, Matthew Kaufman wrote:
> I see nothing in this draft which has changed the landscape as it 
> existed when these blog postings were made:
>
> http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx
May 3, 2010
>
> http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx
Likely May 19, 2010, although the article does not say so.

Since WebM was open-sourced around May 20, 2010, which postdates these 
articles, I guess there isn't much that you would consider to have 
changed the landscape.

>
> I think that we should only seriously consider MTI codecs which have 
> been developed and improved within the context of an existing 
> standards body, for the multitude of technical and non-technical 
> benefits that process provides for those who ship the codecs in their 
> products.
>
> I will note, for instance, that the Opus audio codec is a 
> substantially better general-purpose Internet codec than SILK as a 
> result of following this process.
If the video-codec BOF is a success, such a codec, which is also 
possible to practice without onerous, secret and/or expensive permission 
agreements,  may exist 2 years from now.

At the moment, if we want open, good and developed in cooperation with a 
standards body, it seems that we only get 2 out of 3. At most.
>
> Matthew Kaufman


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    <div class="moz-cite-prefix">On 10/19/2012 06:06 PM, Matthew Kaufman
      wrote:<br>
    </div>
    <blockquote
cite="mid:AE1A6B5FD507DC4FB3C5166F3A05A484160F0374@tk5ex14mbxc272.redmond.corp.microsoft.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <style type="text/css" id="owaParaStyle"></style>
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">I see nothing in this draft which has
        changed the landscape as it existed when these blog postings
        were made:<br>
        <br>
        <a moz-do-not-send="true"
href="http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx"
          target="_blank">http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx</a></div>
    </blockquote>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <span style="color: rgb(102, 102, 102); font-family: 'Segoe UI',
      'Lucida Grande', Verdana, Arial, Helvetica, sans-serif; font-size:
      13px; font-style: normal; font-variant: normal; font-weight:
      normal; letter-spacing: normal; line-height: normal; orphans: 2;
      text-align: left; text-indent: 0px; text-transform: none;
      white-space: normal; widows: 2; word-spacing: 0px;
      -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;
      background-color: rgb(255, 255, 255); display: inline !important;
      float: none;">May 3, 2010</span>
    <blockquote
cite="mid:AE1A6B5FD507DC4FB3C5166F3A05A484160F0374@tk5ex14mbxc272.redmond.corp.microsoft.com"
      type="cite">
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">
        <div><br>
        </div>
        <div><a moz-do-not-send="true"
href="http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx"
            target="_blank">http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx</a></div>
      </div>
    </blockquote>
    Likely May 19, 2010, although the article does not say so.<br>
    <br>
    Since WebM was open-sourced around May 20, 2010, which postdates
    these articles, I guess there isn't much that you would consider to
    have changed the landscape.<br>
    <br>
    <blockquote
cite="mid:AE1A6B5FD507DC4FB3C5166F3A05A484160F0374@tk5ex14mbxc272.redmond.corp.microsoft.com"
      type="cite">
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">
        <div><br>
        </div>
        <div>I think that we should only seriously consider MTI codecs
          which have been developed and improved within the context of
          an existing standards body, for the multitude of technical and
          non-technical benefits that process provides for those who
          ship the codecs in their products.</div>
        <div><br>
        </div>
        <div>I will note, for instance, that the Opus audio codec is a
          substantially better general-purpose Internet codec than SILK
          as a result of following this process.</div>
      </div>
    </blockquote>
    If the video-codec BOF is a success, such a codec, which is also
    possible to practice without onerous, secret and/or expensive
    permission agreements,&nbsp; may exist 2 years from now.<br>
    <br>
    At the moment, if we want open, good and developed in cooperation
    with a standards body, it seems that we only get 2 out of 3. At
    most.<br>
    <blockquote
cite="mid:AE1A6B5FD507DC4FB3C5166F3A05A484160F0374@tk5ex14mbxc272.redmond.corp.microsoft.com"
      type="cite">
      <div style="direction: ltr;font-family: Tahoma;color:
        #000000;font-size: 10pt;">
        <div><br>
        </div>
        <div>Matthew Kaufman</div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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--f46d042f9318101b4404cc6c52b2
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One other common case for offer roll back are O/A collisions. Both sides
are trying to make changes to SDP at the same time, offers cross over on
the wire, so, at least one side needs to rollback the offer and process the
offer received. The only way I can see it implemented is to generate a hold
response SDP based on the generated offer, send it to the
setRemoteDescription, and then process the offer from the remote side. Mind
you, this does not map well to SIP collusion handling where both offers
needs to be reverted, and this will most likely generate an audio
disruption even in case of only one offer being rolled back by putting on
hold.
_____________
Roman Shpount

--f46d042f9318101b4404cc6c52b2
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One other common case for offer roll back are O/A=A0collisions. Both sides =
are trying to make changes to SDP at the same time, offers cross over on th=
e wire, so, at least one side needs to rollback the offer and process the o=
ffer received. The only way I can see it implemented is to generate a hold =
response SDP based on the generated offer, send it to the setRemoteDescript=
ion, and then process the offer from the remote side. Mind you, this does n=
ot map well to SIP=A0collusion handling where both offers needs to be rever=
ted, and this will most likely generate an audio disruption even in case of=
 only one offer being rolled back by putting on hold.=A0<br clear=3D"all">
_____________<br>Roman Shpount<br><br>

--f46d042f9318101b4404cc6c52b2--

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[re-sent with the correct From address]

> From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
> 
> One discussion I think we've not had yet is the granularity the API 
> could report "can't use this SDP blob" on. Say that a new offer (in on 
> ongoing sessions) was received indicating the addition of two audio and 
> two video tracks (from Alice to Bob). What if Bob's browser could decode 
> both audio tracks but has resources to handle only one of the video 
> tracks, should it be able to tell so (perhaps with the result that two 
> audio and one video track was added)? Or should the update of the 
> session just fail?

Within the use case of a WebRTC server/SIP gateway, whenever the
WebRTC client requests an update of SDP, the SIP far end UA can reject
the update.  (In regard to current practice, most of the time a UA
will accept the part of the SDP that it can and make the rest
inactive, but some transcoding SIP B2BUAs will fix a codec early in
the dialog and refuse any SDP changes after that.  In addition, some
UAs do not support UPDATE at all.)

The SIP gateway needs to be able to report this failure to the WebRTC
client API in a usable way, and the reporting has to be asynchronous
to the SDP update request.  By the rules of SIP, the session state
reverts to the previous session state, but it would be a reasonable
architecture if the gateway was required to execute additional WebRTC
actions to achieve that effect.  (However, the WebRTC client must be
forbidden from rejecting those actions!)

In theory, the reestablishment of the previous SDP state could require
renegotiating ICE (and SRTP keys?), but in practice, that would
require excessive overhead, I think.

Ideally, this does not require the code in the JSEP application to
know specifically that the remote end is a SIP gateway, because we
don't want the WebRTC client applications to have to be specially
coded to support SIP gateways (or any other particular type of WebRTC
server).

Given that these things need to be supported, it is easy for the SIP
gateway to interwork with a WebRTC architecture that allows SDP
updates in states where the SIP signaling does not:  If the SIP
gateway receives a WebRTC SDP update in a state where an update is not
permitted in SIP, it can instead send an update rejection to the
WebRTC client.  This allows the WebRTC offer/answer structure to be
more flexible than the SIP offer/answer structure without hindering
interworking (other than that sometimes WebRTC SDP updates fail).

Dale

From worley@shell01.TheWorld.com  Fri Oct 19 10:26:11 2012
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Subject: Re: [rtcweb] Video Codec - Possible to use.
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Clearly, the API that the application uses must have some feature that
allows the application to determine the codecs that are provided by
the platform on which the application executes.  This information must
be provided in a format that plugs into the SDP, that is, the
<encoding name> field value in the a=rtpmap line.

Exactly how the API determines this information is unspecified, but of
course, it is coordinated with the mechanism by which the incoming RTP
is sent to the codec and ultimately displayed.

And there should be a "mandatory to implement" video codec to ensure
interoperation.  The tricky question is which one should it be?  It
should be believed to be universally royalty-free.  Certainly it
should not require a negotiated licensing agreement, as that would
suppress development by non-equipment-vendors.  It does *not* have to
be state of the art.  It can't be too difficult to find *one* codec
with these properties.

Dale

From martin.thomson@gmail.com  Fri Oct 19 10:36:31 2012
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On 19 October 2012 10:24, Dale R. Worley <worley@ariadne.com> wrote:
> It can't be too difficult to find *one* codec with these properties.

RFC 2435 was not considered adequate for the described use cases.

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Draft agenda for RTCWeb session 2 at IETF85
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We have two WebRTC session at the next IETF. We have not worked out the age=
nda for the first session yet but our current plan is to dedicate the secon=
d session to the MTI video codec question.=20

The draft agenda for this season is:

Admin issues (5 min)

IPR at the IETF ( 10 min )
- this is a reminder of IETF rules and guidelines around IPR

Presentations ( 80 min )
- a 20 minute presentation from each of the following drafts=20
     draft-alvestrand-rtcweb-vp8-00
     draft-burman-rtcweb-h264-proposal-00
     draft-dbenham-webrtc-videomti-00
     draft-marjou-rtcweb-video-codec-00
- If the authors of drafts recommending the same codec can consult with
each other to avoid overlap, or even merge presentations, the
recovered time would be given to the general discussion.


General discussion (  30 min )
- general time for everyone to speak at the mic to express information impo=
rtant for the decision


Call the question of which mandatory to implement video codec to select  ( =
5 min )


Next steps ( 20 min )


Cullen, Magnus, and Ted <RTCWeb Chairs>





From pkyzivat@alum.mit.edu  Fri Oct 19 17:06:34 2012
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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/19/12 12:26 PM, Martin Thomson wrote:
> On 19 October 2012 04:55, Hutton, Andrew
> <andrew.hutton@siemens-enterprise.com> wrote:
>> On 19 October 2012 07:37 Stefan Hakansson LK wrote:
>>> [...] What if Bob's browser could
>>> decode
>>> both  audio tracks but has resources to handle only one of the video
>>> tracks, should it be able to tell so (perhaps with the result that two
>>> audio and one video track was added)? Or should the update of the
>>> session just fail?
>>
>> The update to the session should not just fail. Normally in this case the answer would simple reject the one video stream and not the whole offer.
>
> A more interesting case is where Alice decides to (attempt to) switch
> codecs.  That decision might even be based on some prior knowledge of
> Bob's ability to understand codecs.  Maybe Bob originally offers A and
> B, but when Alice decides to upgrade from A to B, that is not a mode
> that Bob supports any more.  The session should continue with A.
>
> An answer that rejected the stream would cause an interruption to the session.

Well, actually it would interrupt the media stream (rtp session).
I agree there are cases where rejecting the offer would make more sense.
That is certainly an option in sip. Deciding whether it is better to 
reject the media stream, or reject the offer can be tricky.

A more important aspect of this scenario is how it is managed on the 
offering side. When Alice makes the attempt, does she assume success and 
immediately put the offered SDP into operation? (And then undo it if the 
offer fails.) Or does the old local/remote SDP remain in effect until 
the answer is received?

SIP says that you must be prepared to receive according to the new offer 
as soon as it is sent. And yet the stuff you are receiving will still be 
in accord to the old o/a until the new offer is received by the 
answerer. And the offerer should still be sending according to the old 
stuff until it receives the answer. Hence there is a time period when 
you need to support *both*. If you switch too soon you may receive old 
stuff you can't deal with. And if you switch too soon and the offer is 
rejected then you will have to switch back, and there will be more 
disruption.

This is pretty ugly, and unless you can actually support both old and 
new concurrently (which many cannot) then you are forced to choose among 
bad choices for how to implement.

But webrtc can't ignore this. It must acknowledge this and provide some 
mechanisms to deal with it.

	Thanks,
	Paul

>> [...] I am also thinking it would be nice if the API provided some way of cancelling an offer and reverting to the original state but if this is putting too much state back in the browser we would need a clearly documented mechanism.
>
> Nice or necessary?  There is already a great deal of state in the browser.
>
> I am actually interested in a small modification to the API for this
> case, for outstanding offers there are actually three pieces of SDP in
> play: original local description (could be offer or answer), original
> remote description (answer or offer) and the new outstanding offer
> (which is only ever a local description, the action can be immediate
> on the answering side).  We currently have a means to get two of
> these: the remote description and (I assume) the local description
> that was most recently set.  We do not have a way to get the original
> local description.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From bernard_aboba@hotmail.com  Sat Oct 20 01:12:42 2012
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Subject: [rtcweb] RTCWEB Session 2:  A plea for brevity
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--_b4d23681-3a5b-481e-ad88-fae6a78171e3_
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I respectfully submit that the allocating the entire 150 minute session to =
the video codec MTI isssue is not the best use of the WG's time.  =20

Do we really  have to allocate 20 minutes to *each* draft?  Since 3 of =0A=
the presentations are favoring H.264 and only one is pushing VP8=2C the =0A=
fairness of this approach is questionable=2C and since brevity is the soul=
=0A=
 of wit (and argument)=2C I believe that each side could probably do the =
=0A=
job in 20 minutes max=2C including example videos.  =20

If we spend=2C 40 minutes on presentations=2C 30 minutes on discussion and =
20 minutes on the combination of consensus and next steps=2C then that woul=
d leave an hour for other important issues=2C such as the way forward on SD=
P.=20


> From: fluffy@cisco.com
> To: rtcweb@ietf.org
> Date: Fri=2C 19 Oct 2012 18:59:50 +0000
> Subject: [rtcweb] Draft agenda for RTCWeb session 2 at IETF85
>=20
>=20
> We have two WebRTC session at the next IETF. We have not worked out the a=
genda for the first session yet but our current plan is to dedicate the sec=
ond session to the MTI video codec question.=20
>=20
> The draft agenda for this season is:
>=20
> Admin issues (5 min)
>=20
> IPR at the IETF ( 10 min )
> - this is a reminder of IETF rules and guidelines around IPR
>=20
> Presentations ( 80 min )
> - a 20 minute presentation from each of the following drafts=20
>      draft-alvestrand-rtcweb-vp8-00
>      draft-burman-rtcweb-h264-proposal-00
>      draft-dbenham-webrtc-videomti-00
>      draft-marjou-rtcweb-video-codec-00
> - If the authors of drafts recommending the same codec can consult with
> each other to avoid overlap=2C or even merge presentations=2C the
> recovered time would be given to the general discussion.
>=20
>=20
> General discussion (  30 min )
> - general time for everyone to speak at the mic to express information im=
portant for the decision
>=20
>=20
> Call the question of which mandatory to implement video codec to select  =
( 5 min )
>=20
>=20
> Next steps ( 20 min )
>=20
>=20
> Cullen=2C Magnus=2C and Ted <RTCWeb Chairs>
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

--_b4d23681-3a5b-481e-ad88-fae6a78171e3_
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<html>
<head>
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<body class=3D'hmmessage'><div dir=3D'ltr'>I respectfully submit that the a=
llocating the entire 150 minute session to the video codec MTI isssue is no=
t the best use of the WG's time.&nbsp=3B&nbsp=3B <br><br>Do we really&nbsp=
=3B have to allocate 20 minutes to *each* draft?&nbsp=3B Since 3 of =0A=
the presentations are favoring H.264 and only one is pushing VP8=2C the =0A=
fairness of this approach is questionable=2C and since brevity is the soul=
=0A=
 of wit (and argument)=2C I believe that each side could probably do the =
=0A=
job in 20 minutes max=2C including example videos.&nbsp=3B&nbsp=3B <br><br>=
If we spend=2C 40 minutes on presentations=2C 30 minutes on discussion and =
20 minutes on the combination of consensus and next steps=2C then that woul=
d leave an hour for other important issues=2C such as the way forward on SD=
P. <br><br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B From: fluf=
fy@cisco.com<br>&gt=3B To: rtcweb@ietf.org<br>&gt=3B Date: Fri=2C 19 Oct 20=
12 18:59:50 +0000<br>&gt=3B Subject: [rtcweb] Draft agenda for RTCWeb sessi=
on 2 at IETF85<br>&gt=3B <br>&gt=3B <br>&gt=3B We have two WebRTC session a=
t the next IETF. We have not worked out the agenda for the first session ye=
t but our current plan is to dedicate the second session to the MTI video c=
odec question. <br>&gt=3B <br>&gt=3B The draft agenda for this season is:<b=
r>&gt=3B <br>&gt=3B Admin issues (5 min)<br>&gt=3B <br>&gt=3B IPR at the IE=
TF ( 10 min )<br>&gt=3B - this is a reminder of IETF rules and guidelines a=
round IPR<br>&gt=3B <br>&gt=3B Presentations ( 80 min )<br>&gt=3B - a 20 mi=
nute presentation from each of the following drafts <br>&gt=3B      draft-a=
lvestrand-rtcweb-vp8-00<br>&gt=3B      draft-burman-rtcweb-h264-proposal-00=
<br>&gt=3B      draft-dbenham-webrtc-videomti-00<br>&gt=3B      draft-marjo=
u-rtcweb-video-codec-00<br>&gt=3B - If the authors of drafts recommending t=
he same codec can consult with<br>&gt=3B each other to avoid overlap=2C or =
even merge presentations=2C the<br>&gt=3B recovered time would be given to =
the general discussion.<br>&gt=3B <br>&gt=3B <br>&gt=3B General discussion =
(  30 min )<br>&gt=3B - general time for everyone to speak at the mic to ex=
press information important for the decision<br>&gt=3B <br>&gt=3B <br>&gt=
=3B Call the question of which mandatory to implement video codec to select=
  ( 5 min )<br>&gt=3B <br>&gt=3B <br>&gt=3B Next steps ( 20 min )<br>&gt=3B=
 <br>&gt=3B <br>&gt=3B Cullen=2C Magnus=2C and Ted &lt=3BRTCWeb Chairs&gt=
=3B<br>&gt=3B <br>&gt=3B <br>&gt=3B <br>&gt=3B <br>&gt=3B _________________=
______________________________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcw=
eb@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div=
> 		 	   		  </div></body>
</html>=

--_b4d23681-3a5b-481e-ad88-fae6a78171e3_--

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Subject: Re: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On 10/19/2012 04:26 PM, Matthew Kaufman wrote:
> Iaki Baz Castillo [ibc@aliax.net]:
>> SIP and XMPP are signaling protocols using SDP and fit very well into
>> the state machine defined in RFC 3264. In WebRTC we DON'T have a
>> (network) signaling protocol and, thus, it's the JavaScript
>> responsability to respect or not RFC 3264 rules, but those rules
>> cannot be mandated at the media stack level.
>>
>> So honestly I don't fully understand what we are discussing in this
>> thread. Regards.
Inaki is not alone.
The text in the JSEP draft could definitely bear improving. But I have 
problems parsing the definition of "compliant" here:
> What I am hoping to discuss is whether the JSEP API (which repeatedly claims to be exposing an RFC3264 SDP O/A compliant API surface*) has or has not "relaxed" RFC3264. I think it is pretty clear that it has (in that it does not in fact enforce any of "MUST NOT" called out in RFC3264 at the API level),
in that I would expect a "compliant" API to be one that permits an RFC 
3264 protocol machine to be built on top of it, which doesn't make the 
sentence above logical
>   and so now that we don't have an API that complies with SDP O/A
.... a premise I don't believe.
>   *any* API that doesn't comply with SDP O/A should be equally acceptable.
.... but this conclusion is still completely bizarre to me.

RFC 3264 compliance (for whatever definition of compliance) is just one 
aspect of the considerations of what main direction of API to choose. 
Anyone who wasn't part of the WEBRTC discussions is free to consult the 
archives of the recent discussion for the other reasons put forward.

"Equally acceptable" is a very strange phrase to use in this discussion 
(which, anyway, is in the wrong forum).


From john@jlc.net  Sat Oct 20 06:48:35 2012
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Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>...
> Do we really  have to allocate 20 minutes to *each* draft?...

   +1

   (Disclaimer: I won't be there, but I will be listening.)

--
John Leslie <john@jlc.net>

From fluffy@iii.ca  Sat Oct 20 07:16:03 2012
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Subject: Re: [rtcweb] RTCWEB Session 2:  A plea for brevity
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On Oct 20, 2012, at 7:48 AM, John Leslie <john@jlc.net> wrote:

> Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>> ...
>> Do we really  have to allocate 20 minutes to *each* draft?...
>=20
>   +1


The agenda is draft and I'm sure I and the other chairs will take the =
feedback we get into account as we refine this closer to the meeting. =
The MTI video codec is a pretty big questions we are asking and my =
observation about getting consensus is that people need to have been =
given the time to make their key argument or it is very hard to get to =
consensus. I realize that some people believe, or even hope, that we =
will not get to consensus on this topic so it is all a waste of time but =
the WG has made it clear it does want to get to have a MTI video codec.=20=


So consider the situation we are in. One of the drafts says when making =
measurements on quality they found A was better than B while the another =
draft says B was better than A. Several people have asked me how this =
could happen two different groups ended up with opposite measurements =
and they want to understand more about how this could have happened and =
which is correct.=20

Let me ask you guys a question, how many people do you think will be in =
the microphone line up when theses two are discussed? You have both seen =
plenty of controversial topics discussed at other ietf meetings. What =
would you recommend to people making slides for a slot like this - how =
many slides do you think they can reasonably get through in a time slot =
like this? What would your best guess be of how long we need to make =
these slots such that the chairs don't have to cut off the mic lineup?=20=


I know you could take these as vaguely cynical rhetorical questions but =
I don't mean them that way. I'm straight up asking you how much time you =
think we need such that authors of the drafts can make their key =
argument and people that have questions or disagreements that they think =
are important for the WG to understand can make ask their question.

Cullen <with my co-chair hat on>







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Subject: Re: [rtcweb] RTCWEB Session 2:  A plea for brevity
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On 10/20/2012 10:12 AM, Bernard Aboba wrote:
> I respectfully submit that the allocating the entire 150 minute 
> session to the video codec MTI isssue is not the best use of the WG's 
> time.
>
> Do we really  have to allocate 20 minutes to *each* draft? Since 3 of 
> the presentations are favoring H.264 and only one is pushing VP8, the 
> fairness of this approach is questionable, and since brevity is the 
> soul of wit (and argument), I believe that each side could probably do 
> the job in 20 minutes max, including example videos.
>
> If we spend, 40 minutes on presentations, 30 minutes on discussion and 
> 20 minutes on the combination of consensus and next steps, then that 
> would leave an hour for other important issues, such as the way 
> forward on SDP.

To my mind, the discussion so far has matured to the point where the 
presentations' salient points could be made in two sentences:

- Technical quality: Close enough that it doesn't matter.
- IPR conditions: Different enough that it matters.

Different people have made different qualitative evaluation of "which 
one is best". I have not yet heard anyone say "if X is N% better on 
measurement Y, I will support that one".

Given what lawyers say when I ask them if I can say anything about legal 
topics (including opinions about other people's licensing terms), it'll 
be hard to have a meaningful discussion on the point that matters.

This is a problem. I don't know how to resolve it. But I'd rather be 
discussing SDP than having 150 minutes of discussion on the stuff that 
doesn't matter because we can't come to grips with the stuff that matters.


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    <div class="moz-cite-prefix">On 10/20/2012 10:12 AM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W62D02E8AAD031475EEE21A93740@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">I respectfully submit that the allocating the
        entire 150 minute session to the video codec MTI isssue is not
        the best use of the WG's time.&nbsp;&nbsp; <br>
        <br>
        Do we really&nbsp; have to allocate 20 minutes to *each* draft?&nbsp;
        Since 3 of the presentations are favoring H.264 and only one is
        pushing VP8, the fairness of this approach is questionable, and
        since brevity is the soul of wit (and argument), I believe that
        each side could probably do the job in 20 minutes max, including
        example videos.&nbsp;&nbsp; <br>
        <br>
        If we spend, 40 minutes on presentations, 30 minutes on
        discussion and 20 minutes on the combination of consensus and
        next steps, then that would leave an hour for other important
        issues, such as the way forward on SDP. <br>
      </div>
    </blockquote>
    <br>
    To my mind, the discussion so far has matured to the point where the
    presentations' salient points could be made in two sentences:<br>
    <br>
    - Technical quality: Close enough that it doesn't matter.<br>
    - IPR conditions: Different enough that it matters.<br>
    <br>
    Different people have made different qualitative evaluation of
    "which one is best". I have not yet heard anyone say "if X is N%
    better on measurement Y, I will support that one".<br>
    <br>
    Given what lawyers say when I ask them if I can say anything about
    legal topics (including opinions about other people's licensing
    terms), it'll be hard to have a meaningful discussion on the point
    that matters.<br>
    <br>
    This is a problem. I don't know how to resolve it. But I'd rather be
    discussing SDP than having 150 minutes of discussion on the stuff
    that doesn't matter because we can't come to grips with the stuff
    that matters.<br>
    <br>
  </body>
</html>

--------------070807050708020209030707--

From bernard_aboba@hotmail.com  Sat Oct 20 09:12:47 2012
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Subject: Re: [rtcweb] RTCWEB Session 2:  A plea for brevity
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Cullen Jennings said:

"One of the drafts says when making measurements on quality they found A was
better than B while the another draft says B was better than A.  Several
people have asked me how this could happen two different groups ended up
with opposite measurements and they want to understand more about how this
could have happened and which is correct.... Let me ask you guys a question,
how many people do you think will be in the microphone line up when these
two are discussed?""
 
[BA] Assuming those at the mic would be asking clarifying questions, why do
we need to wait until IETF 85 to begin that discussion?   If each side were
to start a thread on the list, at least some of the clarifying questions
could be gotten out of the way beforehand.  Who knows, the result could be
akin to letting a child stay up late on Saturday night to watch TV - the
little devil might be too tired to cause trouble at church on Sunday :)


From bernard_aboba@hotmail.com  Sat Oct 20 09:42:46 2012
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Subject: [rtcweb] Mobile tests in draft-dbenham-webrtc-videomti
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--_7faf1d31-2729-4812-af8e-ac8c695663eb_
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Looking at the videos from Section 4=2C using the same software on both sid=
es might help tease out implementation effects.  My understanding is that B=
ria (at least on the iPad) supports both H.264 and VP8=2C and it would appe=
ar that Linphone also supports both.  Would it be possible to compare a Bri=
a/Linphone conversation with H.264 to one using VP8?

>From Section 4:

Captures of Mobile Tests=3B=0A=
=0A=
   FaceTime using H.264 on an iPhone to linphone on a fast desktop PC.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/facetime.mov=0A=
=0A=
   Bria using VP8 on iPhone to linphone on a fast desktop PC. http://=0A=
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov=0A=
=0A=
   Bria using VP8 on Galaxy Nexus to linphone on a fast desktop PC.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/galaxy-nexus.mov=0A=
=0A=
   Shown side-by-side=2C Bria using VP8 on iPhone (left) and FaceTime with=
=0A=
   H.264/AVC on iPhone (right)=2C going to linphone and FaceTime. http://=
=0A=
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov=0A=
=0A=
   Captures of Desktop-only Tests=3B=0A=
=0A=
   FaceTime H.264/AVC from a fast notebook computer to fast desktop. htt=0A=
   p://dl.dropbox.com/u/17089001/video-samples/facetime-264-desktop.mov=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   facetime-264-desktop.webm=0A=
=0A=
   Chrome with VP8 from same fast notebook computer to same fast=0A=
   desktop.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/chrome-vp8-desktop.mov=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome-vp8-desktop.webm=0A=
=0A=
   Two fast notebook computers - one running Chrome and the other=0A=
   running Fictive - both sending to the same desktop computer shown=0A=
   side by side. http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome_and_factime_desktop.mov>=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome_and_factime_desktop.webm
 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Looking at the videos from Secti=
on 4=2C using the same software on both sides might help tease out implemen=
tation effects.&nbsp=3B My understanding is that Bria (at least on the iPad=
) supports both H.264 and VP8=2C and it would appear that Linphone also sup=
ports both.&nbsp=3B Would it be possible to compare a Bria/Linphone convers=
ation with H.264 to one using VP8?<br><br>From Section 4:<br><br><pre>Captu=
res of Mobile Tests=3B=0A=
=0A=
   FaceTime using H.264 on an iPhone to linphone on a fast desktop PC.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/facetime.mov=0A=
=0A=
   Bria using VP8 on iPhone to linphone on a fast desktop PC. http://=0A=
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov=0A=
=0A=
   Bria using VP8 on Galaxy Nexus to linphone on a fast desktop PC.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/galaxy-nexus.mov=0A=
=0A=
   Shown side-by-side=2C Bria using VP8 on iPhone (left) and FaceTime with=
=0A=
   H.264/AVC on iPhone (right)=2C going to linphone and FaceTime. http://=
=0A=
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov=0A=
=0A=
   Captures of Desktop-only Tests=3B=0A=
=0A=
   FaceTime H.264/AVC from a fast notebook computer to fast desktop. htt=0A=
   p://dl.dropbox.com/u/17089001/video-samples/facetime-264-desktop.mov=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   facetime-264-desktop.webm=0A=
=0A=
   Chrome with VP8 from same fast notebook computer to same fast=0A=
   desktop.=0A=
   http://dl.dropbox.com/u/17089001/video-samples/chrome-vp8-desktop.mov=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome-vp8-desktop.webm=0A=
=0A=
   Two fast notebook computers - one running Chrome and the other=0A=
   running Fictive - both sending to the same desktop computer shown=0A=
   side by side. http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome_and_factime_desktop.mov&gt=3B=0A=
=0A=
   -or- http://dl.dropbox.com/u/17089001/video-samples/=0A=
   chrome_and_factime_desktop.webm</pre><br> 		 	   		  </div></body>
</html>=

--_7faf1d31-2729-4812-af8e-ac8c695663eb_--

From matthew@matthew.at  Sat Oct 20 10:10:10 2012
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Please excuse the test message, but several emails I sent to the list 
from my work account over the last couple of days appear to have been 
dropped and not delivered to the list.

Matthew Kaufman

From matthew@matthew.at  Sat Oct 20 10:21:54 2012
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Subject: [rtcweb] (resend) RE: Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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(Resending from my personal account as the list server dropped a bunch 
of messages and I will be resending from my personal account a few that 
I think are still particularly relevant)

Harald Alvestrand [harald@alvestrand.no]:

 >

 > Because making an interface that makes following an existing practice

 > easy (but does not enforce all the constraints of that practice) is

 > better than making an interface that makes following that practice

 > very complex?

If this myth were correct, it would be a good argument. In practice, the 
JSEP interface (which claims to loosely follow an existing practice) is 
*more difficult* to use when interoperating with endpoints that actually 
follow that practice than the API we proposed does.

In our testing, the amount of Javascript required to unpack the pile of 
SDP that is emitted by an existing implementation of JSEP (the Google 
Chrome beta), pick it apart into what you actually need to interop with 
a real gateway endpoint, exhaustively test what SDP can then be put back 
in to setLocalDescription and setRemoteDescription (as there is no 
specification saying what SDP is and is not allowed here, and no error 
reporting to speak of), and then the signaling on top of that all adds 
up to more than twice as much Javascript as it takes to use our proposed 
direct-manipulation API, create SDP that is compatible with the far end 
you are interoperating with, and start media flowing.

If the existing API actually had a description of what SDP is and is not 
acceptable, and specified an error reporting mechanism that was actually 
helpful for developers, then it would *still* take 2x the Javascript to 
call any endpoint other than one of your own because of the need to 
manipulate the SDP and the need to implement a state machine in 
Javascript while the browser is also keeping a bunch of state internally.

I encourage anyone who cares about interoperability to replicate our 
experiment... build an application using the Chrome beta that calls a 
gateway speaking SIP for signaling and ICE-lite for consent verification 
and see how "easy" it is or isn't.

Matthew Kaufman



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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    (Resending from my personal account as the list server dropped a
    bunch of messages and I will be resending from my personal account a
    few that I think are still particularly relevant)<br>
    <br>
    <meta http-equiv="Content-Type" content="text/html;
      charset=ISO-8859-1">
    <p class="MsoPlainText">Harald Alvestrand [<a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>]:<o:p></o:p></p>
    <p class="MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">&gt; Because making an interface that makes
      following an
      existing practice <o:p></o:p></p>
    <p class="MsoPlainText">&gt; easy (but does not enforce all the
      constraints of
      that practice) is <o:p></o:p></p>
    <p class="MsoPlainText">&gt; better than making an interface that
      makes following
      that practice <o:p></o:p></p>
    <p class="MsoPlainText">&gt; very complex?<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">If this myth were correct, it would be a
      good argument.
      In practice, the JSEP interface (which claims to loosely follow an
      existing
      practice) is *more difficult* to use when interoperating with
      endpoints that
      actually follow that practice than the API we proposed does.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">In our testing, the amount of Javascript
      required to
      unpack the pile of SDP that is emitted by an existing
      implementation of JSEP
      (the Google Chrome beta), pick it apart into what you actually
      need to interop
      with a real gateway endpoint, exhaustively test what SDP can then
      be put back
      in to setLocalDescription and setRemoteDescription (as there is no
      specification saying what SDP is and is not allowed here, and no
      error
      reporting to speak of), and then the signaling on top of that all
      adds up to
      more than twice as much Javascript as it takes to use our proposed
      direct-manipulation API, create SDP that is compatible with the
      far end you are
      interoperating with, and start media flowing.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">If the existing API actually had a
      description of what
      SDP is and is not acceptable, and specified an error reporting
      mechanism that
      was actually helpful for developers, then it would *still* take 2x
      the
      Javascript to call any endpoint other than one of your own because
      of the need
      to manipulate the SDP and the need to implement a state machine in
      Javascript
      while the browser is also keeping a bunch of state internally.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">I encourage anyone who cares about
      interoperability to
      replicate our experiment... build an application using the Chrome
      beta that
      calls a gateway speaking SIP for signaling and ICE-lite for
      consent
      verification and see how "easy" it is or isn't.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Matthew Kaufman<o:p></o:p></p>
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From matthew@matthew.at  Sat Oct 20 10:23:21 2012
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Subject: [rtcweb] (resend) RE: Draft agenda for RTCWeb session 2 at IETF85
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Can we just save a bunch of time and have a show of hands at the 
beginning for:

A) People who WILL NOT implement H.264 in their product and cannot be 
swayed by any presentation

B) People who WILL NOT implement VP8 in their product and cannot be 
swayed by any presentation

C) People who are still undecided enough that they want to hear the 
presentations

It is quite possible that once we get that out of the way, the rest of 
the presentations will be irrelevant and we can get back to discussing 
important things like who is going to write down the particulars for the 
"SDP profile" we're using as an API surface.

Matthew Kaufman

________________________________________

From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org> 
[rtcweb-bounces@ietf.org] on behalf of Cullen Jennings (fluffy) 
[fluffy@cisco.com]

Sent: Friday, October 19, 2012 11:59 AM

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

Subject: [rtcweb] Draft agenda for RTCWeb session 2 at IETF85

We have two WebRTC session at the next IETF. We have not worked out the 
agenda for the first session yet but our current plan is to dedicate the 
second session to the MTI video codec question.

The draft agenda for this season is:

Admin issues (5 min)

IPR at the IETF ( 10 min )

- this is a reminder of IETF rules and guidelines around IPR

Presentations ( 80 min )

- a 20 minute presentation from each of the following drafts

draft-alvestrand-rtcweb-vp8-00

draft-burman-rtcweb-h264-proposal-00

draft-dbenham-webrtc-videomti-00

draft-marjou-rtcweb-video-codec-00

- If the authors of drafts recommending the same codec can consult with 
each other to avoid overlap, or even merge presentations, the recovered 
time would be given to the general discussion.

General discussion (30 min )

- general time for everyone to speak at the mic to express information 
important for the decision

Call the question of which mandatory to implement video codec to select( 
5 min )

Next steps ( 20 min )

Cullen, Magnus, and Ted <RTCWeb Chairs>

_______________________________________________

rtcweb mailing list

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

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


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    <p class="MsoPlainText">Can we just save a bunch of time and have a
      show of hands
      at the beginning for:<o:p></o:p></p>
    <p class="MsoPlainText">A) People who WILL NOT implement H.264 in
      their product
      and cannot be swayed by any presentation<o:p></o:p></p>
    <p class="MsoPlainText">B) People who WILL NOT implement VP8 in
      their product and
      cannot be swayed by any presentation<o:p></o:p></p>
    <p class="MsoPlainText">C) People who are still undecided enough
      that they want
      to hear the presentations<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">It is quite possible that once we get that
      out of the
      way, the rest of the presentations will be irrelevant and we can
      get back to
      discussing important things like who is going to write down the
      particulars for
      the "SDP profile" we're using as an API surface.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Matthew Kaufman<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">________________________________________<o:p></o:p></p>
    <p class="MsoPlainText" style="mso-outline-level:1">From: <a
        href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
      [<a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] on behalf of Cullen Jennings (fluffy)
      [<a class="moz-txt-link-abbreviated" href="mailto:fluffy@cisco.com">fluffy@cisco.com</a>]<o:p></o:p></p>
    <p class="MsoPlainText">Sent: Friday, October 19, 2012 11:59 AM<o:p></o:p></p>
    <p class="MsoPlainText">To: <a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p>
    <p class="MsoPlainText">Subject: [rtcweb] Draft agenda for RTCWeb
      session 2 at
      IETF85<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">We have two WebRTC session at the next IETF.
      We have not
      worked out the agenda for the first session yet but our current
      plan is to
      dedicate the second session to the MTI video codec question.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">The draft agenda for this season is:<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Admin issues (5 min)<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">IPR at the IETF ( 10 min )<o:p></o:p></p>
    <p class="MsoPlainText">- this is a reminder of IETF rules and
      guidelines around
      IPR<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Presentations ( 80 min )<o:p></o:p></p>
    <p class="MsoPlainText">- a 20 minute presentation from each of the
      following
      drafts<o:p></o:p></p>
    <p class="MsoPlainText"><span style="mso-spacerun:yes">&nbsp;&nbsp;&nbsp;&nbsp;
      </span>draft-alvestrand-rtcweb-vp8-00<o:p></o:p></p>
    <p class="MsoPlainText"><span style="mso-spacerun:yes">&nbsp;&nbsp;&nbsp;&nbsp;
      </span>draft-burman-rtcweb-h264-proposal-00<o:p></o:p></p>
    <p class="MsoPlainText"><span style="mso-spacerun:yes">&nbsp;&nbsp;&nbsp;&nbsp;
      </span>draft-dbenham-webrtc-videomti-00<o:p></o:p></p>
    <p class="MsoPlainText"><span style="mso-spacerun:yes">&nbsp;&nbsp;&nbsp;&nbsp;
      </span>draft-marjou-rtcweb-video-codec-00<o:p></o:p></p>
    <p class="MsoPlainText">- If the authors of drafts recommending the
      same codec
      can consult with each other to avoid overlap, or even merge
      presentations, the
      recovered time would be given to the general discussion.<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">General discussion (<span
        style="mso-spacerun:yes">&nbsp;
      </span>30 min )<o:p></o:p></p>
    <p class="MsoPlainText">- general time for everyone to speak at the
      mic to
      express information important for the decision<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Call the question of which mandatory to
      implement video
      codec to select<span style="mso-spacerun:yes">&nbsp; </span>( 5 min )<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Next steps ( 20 min )<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">Cullen, Magnus, and Ted &lt;RTCWeb
      Chairs&gt;<o:p></o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
    <p class="MsoPlainText">_______________________________________________<o:p></o:p></p>
    <p class="MsoPlainText">rtcweb mailing list<o:p></o:p></p>
    <p class="MsoPlainText"><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></p>
    <p class="MsoPlainText"><a
        href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
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[Note: Harald's reply to this message which quoted part of it did make 
it to the list, but my original did not]


I see nothing in this draft which has changed the landscape as it 
existed when these blog postings were made:

http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx 


http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx

I think that we should only seriously consider MTI codecs which have 
been developed and improved within the context of an existing standards 
body, for the multitude of technical and non-technical benefits that 
process provides for those who ship the codecs in their products.

I will note, for instance, that the Opus audio codec is a substantially 
better general-purpose Internet codec than SILK as a result of following 
this process.

Matthew Kaufman


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        style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;
mso-fareast-font-family:&quot;Times
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        message which quoted part of it did make it to the list, but my
        original did not]</span></p>
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        style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;
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        I see nothing in this
        draft which has changed the landscape as it existed when these
        blog postings
        were made:<br>
        <br>
        <a
href="http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx"
          target="_blank">http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx</a>
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href="http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx"
          target="_blank">http://blogs.windows.com/windows/b/bloggingwindows/archive/2010/05/19/another-follow-up-on-html5-video-in-ie9.aspx</a><o:p></o:p></span></p>
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        style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;
mso-fareast-font-family:&quot;Times
        New Roman&quot;;color:black">I think that we should
        only seriously consider MTI codecs which have been developed and
        improved
        within the context of an existing standards body, for the
        multitude of
        technical and non-technical benefits that process provides for
        those who ship
        the codecs in their products.<o:p></o:p></span></p>
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        instance, that the Opus audio codec is a substantially better
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        Internet codec than SILK as a result of following this process.<o:p></o:p></span></p>
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Harald Alvestrand [harald@alvestrand.no]*:*

>At the moment, if we want open, good and  developed in cooperation with a standards body, it seems that we only 
get 2 out of 3. At most.

For video, H.264 is available in open source implementations and the 
specification is available to read by all, the quality is more than good 
enough, and it was developed in cooperation with a recognized standards 
body.

I think you're looking for a word other than "open" to explain what you 
might not like about it... but the alternative you've proposed meets 
even fewer of these, and I would claim that the last is more important 
than the missing one you're thinking of.

Matthew Kaufman


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        style="font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;mso-fareast-font-family:
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        New Roman&quot;;color:black">Harald Alvestrand
        [<a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>]<b>:</b></span><span
        style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;mso-fareast-font-family:
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        New Roman&quot;;color:black"><o:p></o:p></span></p>
    <p class="MsoNormal"><span
        style="mso-fareast-font-family:&quot;Times New Roman&quot;;
        color:black">&gt;At the moment, if we want open, good and
        developed in
        cooperation with a standards body, it seems that we only get 2
        out of 3. At
        most.<o:p></o:p></span></p>
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    <p class="MsoNormal"><span
        style="mso-fareast-font-family:&quot;Times New Roman&quot;;
        color:black">For video, H.264 is available in open source
        implementations and
        the specification is available to read by all, the quality is
        more than good
        enough, and it was developed in cooperation with a recognized
        standards body.<o:p></o:p></span></p>
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    <p class="MsoNormal"><span
        style="mso-fareast-font-family:&quot;Times New Roman&quot;;
        color:black">I think you're looking for a word other than "open"
        to
        explain what you might not like about it... but the alternative
        you've proposed
        meets even fewer of these, and I would claim that the last is
        more important
        than the missing one you're thinking of.<o:p></o:p></span></p>
    <p class="MsoNormal"><span
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--------------070503060208060506010102--

From harald@alvestrand.no  Sat Oct 20 15:30:26 2012
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Subject: [rtcweb] SDP testing (Re: (resend) RE: Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting))
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(Chrome developer hat on)
Thanks for the info - can you point to the bugs you filed on these 
issues with our implementation? I'll try to see that they're followed up.

It would be helpful if you specified which endpoint you interoperated 
with, but the most important thing is to show what the SDP was and where 
it was wrong for interoperability.

Frankly, the code in Chrome is a completely new SDP parser/generator, 
and definitely needs testing.

           Harald

On 10/20/2012 07:21 PM, Matthew Kaufman wrote:
> (Resending from my personal account as the list server dropped a bunch 
> of messages and I will be resending from my personal account a few 
> that I think are still particularly relevant)
>
> Harald Alvestrand [harald@alvestrand.no]:
>
> >
>
> > Because making an interface that makes following an existing practice
>
> > easy (but does not enforce all the constraints of that practice) is
>
> > better than making an interface that makes following that practice
>
> > very complex?
>
> If this myth were correct, it would be a good argument. In practice, 
> the JSEP interface (which claims to loosely follow an existing 
> practice) is *more difficult* to use when interoperating with 
> endpoints that actually follow that practice than the API we proposed 
> does.
>
> In our testing, the amount of Javascript required to unpack the pile 
> of SDP that is emitted by an existing implementation of JSEP (the 
> Google Chrome beta), pick it apart into what you actually need to 
> interop with a real gateway endpoint, exhaustively test what SDP can 
> then be put back in to setLocalDescription and setRemoteDescription 
> (as there is no specification saying what SDP is and is not allowed 
> here, and no error reporting to speak of), and then the signaling on 
> top of that all adds up to more than twice as much Javascript as it 
> takes to use our proposed direct-manipulation API, create SDP that is 
> compatible with the far end you are interoperating with, and start 
> media flowing.
>
> If the existing API actually had a description of what SDP is and is 
> not acceptable, and specified an error reporting mechanism that was 
> actually helpful for developers, then it would *still* take 2x the 
> Javascript to call any endpoint other than one of your own because of 
> the need to manipulate the SDP and the need to implement a state 
> machine in Javascript while the browser is also keeping a bunch of 
> state internally.
>
> I encourage anyone who cares about interoperability to replicate our 
> experiment... build an application using the Chrome beta that calls a 
> gateway speaking SIP for signaling and ICE-lite for consent 
> verification and see how "easy" it is or isn't.
>
> Matthew Kaufman
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">(Chrome developer hat on)<br>
      Thanks for the info - can you point to the bugs you filed on these
      issues with our implementation? I'll try to see that they're
      followed up.<br>
      <br>
      It would be helpful if you specified which endpoint you
      interoperated with, but the most important thing is to show what
      the SDP was and where it was wrong for interoperability.<br>
      <br>
      Frankly, the code in Chrome is a completely new SDP
      parser/generator, and definitely needs testing.<br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
      <br>
      On 10/20/2012 07:21 PM, Matthew Kaufman wrote:<br>
    </div>
    <blockquote cite="mid:5082DDAF.7080102@matthew.at" type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      (Resending from my personal account as the list server dropped a
      bunch of messages and I will be resending from my personal account
      a few that I think are still particularly relevant)<br>
      <br>
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <p class="MsoPlainText">Harald Alvestrand [<a
          moz-do-not-send="true" class="moz-txt-link-abbreviated"
          href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>]:<o:p></o:p></p>
      <p class="MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
      <p class="MsoPlainText">&gt; Because making an interface that
        makes following an existing practice <o:p></o:p></p>
      <p class="MsoPlainText">&gt; easy (but does not enforce all the
        constraints of that practice) is <o:p></o:p></p>
      <p class="MsoPlainText">&gt; better than making an interface that
        makes following that practice <o:p></o:p></p>
      <p class="MsoPlainText">&gt; very complex?<o:p></o:p></p>
      <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
      <p class="MsoPlainText">If this myth were correct, it would be a
        good argument. In practice, the JSEP interface (which claims to
        loosely follow an existing practice) is *more difficult* to use
        when interoperating with endpoints that actually follow that
        practice than the API we proposed does.<o:p></o:p></p>
      <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
      <p class="MsoPlainText">In our testing, the amount of Javascript
        required to unpack the pile of SDP that is emitted by an
        existing implementation of JSEP (the Google Chrome beta), pick
        it apart into what you actually need to interop with a real
        gateway endpoint, exhaustively test what SDP can then be put
        back in to setLocalDescription and setRemoteDescription (as
        there is no specification saying what SDP is and is not allowed
        here, and no error reporting to speak of), and then the
        signaling on top of that all adds up to more than twice as much
        Javascript as it takes to use our proposed direct-manipulation
        API, create SDP that is compatible with the far end you are
        interoperating with, and start media flowing.<o:p></o:p></p>
      <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
      <p class="MsoPlainText">If the existing API actually had a
        description of what SDP is and is not acceptable, and specified
        an error reporting mechanism that was actually helpful for
        developers, then it would *still* take 2x the Javascript to call
        any endpoint other than one of your own because of the need to
        manipulate the SDP and the need to implement a state machine in
        Javascript while the browser is also keeping a bunch of state
        internally.<o:p></o:p></p>
      <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
      <p class="MsoPlainText">I encourage anyone who cares about
        interoperability to replicate our experiment... build an
        application using the Chrome beta that calls a gateway speaking
        SIP for signaling and ICE-lite for consent verification and see
        how "easy" it is or isn't.<o:p></o:p></p>
      <p class="MsoPlainText"><o:p>&nbsp;</o:p></p>
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      <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>
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Harald said:
"Thanks for the info - can you point to the bugs you filed on these=0A=
      issues with our implementation? I'll try to see that they're=0A=
      followed up."
=0A=
     =20
[BA] To determine whether something is a "bug" or a "feature" it helps to h=
ave a specification that states what is expected.  Since we're primarily ta=
lking about SDP usage within the API rather than what travels over the wire=
 (after all=2C signaling is out of scope in WebRTC)=2C http://tools.ietf.or=
g/html/draft-nandakumar-rtcweb-sdp is only a baby step toward determining w=
hether the SDP output by createOffer is conformant or whether input to setL=
ocalDescription or setRemoteDescription should be acceptable or not=2C and =
if not=2C what should happen.=20
For example=2C  posting the SDP that is emitted by various createOffer() im=
plementations might enhance the appreciation of the importance of the issue=
=2C I am not sure how to evaluate the results without understanding whether=
 that output is expected to be valid SDP or not.  For example=2C if trickle=
 ICE is used=2C then perhaps one should not expect createOffer to provide S=
DP with valid addresses and ports filled in=3B instead that information is =
expected to be filled in later.  		 	   		  =

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<html>
<head>
<style><!--
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padding:0px
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body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Harald said:<div><div class=3D"e=
cxmoz-cite-prefix"><br>"Thanks for the info - can you point to the bugs you=
 filed on these=0A=
      issues with our implementation? I'll try to see that they're=0A=
      followed up."<br>=0A=
      <br>[BA] To determine whether something is a "bug" or a "feature" it =
helps to have a specification that states what is expected. &nbsp=3BSince w=
e're primarily talking about SDP usage within the API rather than what trav=
els over the wire (after all=2C signaling is out of scope in WebRTC)=2C&nbs=
p=3B<a href=3D"http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp" styl=
e=3D"font-size: 12pt=3B ">http://tools.ietf.org/html/draft-nandakumar-rtcwe=
b-sdp</a>&nbsp=3Bis only a baby step toward determining whether the SDP out=
put by createOffer is conformant or whether input to setLocalDescription or=
 setRemoteDescription should be acceptable or not=2C and if not=2C what sho=
uld happen.&nbsp=3B</div></div><div class=3D"ecxmoz-cite-prefix"><br></div>=
<div class=3D"ecxmoz-cite-prefix">For example=2C &nbsp=3Bposting the SDP th=
at is emitted by various createOffer() implementations might enhance the ap=
preciation of the importance of the issue=2C I am not sure how to evaluate =
the results without understanding whether that output is expected to be val=
id SDP or not. &nbsp=3BFor example=2C if trickle ICE is used=2C then perhap=
s one should not expect createOffer to provide SDP with valid addresses and=
 ports filled in=3B instead that information is expected to be filled in la=
ter.&nbsp=3B</div> 		 	   		  </div></body>
</html>=

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Subject: Re: [rtcweb] SDP testing
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On 10/20/2012 4:20 PM, Bernard Aboba wrote:
> Harald said:
>
> "Thanks for the info - can you point to the bugs you filed on these 
> issues with our implementation? I'll try to see that they're followed up."
>
> [BA] To determine whether something is a "bug" or a "feature" it helps 
> to have a specification that states what is expected.  Since we're 
> primarily talking about SDP usage within the API rather than what 
> travels over the wire (after all, signaling is out of scope in 
> WebRTC), http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp is 
> only a baby step toward determining whether the SDP output by 
> createOffer is conformant or whether input to setLocalDescription or 
> setRemoteDescription should be acceptable or not, and if not, what 
> should happen.
>

Precisely the problem. It isn't a "bug" with the Google Chrome beta. It 
is a "bug" with the current specification, which says "see SDP O/A RFC" 
rather than "this is what a browser must allow" or even "this is the 
type of error a browser must throw if the rules aren't followed".

I'll pick some choice examples from some SDP that the latest Chrome beta 
just gave me from createOffer (without pasting the entire blob it 
disgorges):


s=-

Can I add "e=matthew@matthew.at" right after this?

How about a "p=" line?

a=group:BUNDLE audio video

Can I remove this before passing it back to setLocal Description?

a=ice-options:google-ice

How about removing this? Or changing it to 
"a=ice-options:matthew-special-ice"?

a=sendrecv

Can I change it to sendonly?

a=rtcp-mux

Remove this?

a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:66AIWBQ/R+UnMc4N/NP/b2hjEB6ydFMMCc8n3DMp

Remove this? Change the key? Set up DTLS-SRTP instead?

a=rtpmap:103 ISAC/16000

Change this to rtpmap: 110 ISAC/16000 because I prefer 110 as the codec ID?

a=ssrc:1018881792 cname:B0VSPpORV75cCTIv

Delete this? Change the ssrc ID because I don't like this one? or the cname?

m=video 1 RTP/SAVPF 100 101 102 c=IN IP4 0.0.0.0

Change this to RTP/SAVP ? [Never mind that "IN IP4 0.0.0.0" is a strange 
thing to have coming from "createOffer"]

NOWHERE in the JSEP specification can I find the answers to these 
questions. That makes working out what is allowed a browser-by-browser 
exhaustive search, which is unacceptable.

Matthew Kaufman

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    <div class="moz-cite-prefix">On 10/20/2012 4:20 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W136BDAECA1A38C4DA5FB5E593740@phx.gbl"
      type="cite">
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      <div dir="ltr">Harald said:
        <div>
          <div class="ecxmoz-cite-prefix"><br>
            "Thanks for the info - can you point to the bugs you filed
            on these issues with our implementation? I'll try to see
            that they're followed up."<br>
            <br>
            [BA] To determine whether something is a "bug" or a
            "feature" it helps to have a specification that states what
            is expected. &nbsp;Since we're primarily talking about SDP usage
            within the API rather than what travels over the wire (after
            all, signaling is out of scope in WebRTC),&nbsp;<a
              moz-do-not-send="true"
              href="http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp"
              style="font-size: 12pt; ">http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp</a>&nbsp;is
            only a baby step toward determining whether the SDP output
            by createOffer is conformant or whether input to
            setLocalDescription or setRemoteDescription should be
            acceptable or not, and if not, what should happen.&nbsp;</div>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    Precisely the problem. It isn't a "bug" with the Google Chrome beta.
    It is a "bug" with the current specification, which says "see SDP
    O/A RFC" rather than "this is what a browser must allow" or even
    "this is the type of error a browser must throw if the rules aren't
    followed".<br>
    <br>
    I'll pick some choice examples from some SDP that the latest Chrome
    beta just gave me from createOffer (without pasting the entire blob
    it disgorges):<br>
    <span style="color: rgb(0, 0, 0); font-family: 'Times New Roman';
      font-size: medium; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-align: start; text-indent: 0px; text-transform:
      none; white-space: normal; widows: 2; word-spacing: 0px;
      -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;
      display: inline !important; float: none;"><br>
    </span><br>
    <span style="color: rgb(0, 0, 0); font-family: 'Times New Roman';
      font-size: medium; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-align: start; text-indent: 0px; text-transform:
      none; white-space: normal; widows: 2; word-spacing: 0px;
      -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;
      display: inline !important; float: none;"><span style="color:
        rgb(0, 0, 0); font-family: 'Times New Roman'; font-size: medium;
        font-style: normal; font-variant: normal; font-weight: normal;
        letter-spacing: normal; line-height: normal; orphans: 2;
        text-align: start; text-indent: 0px; text-transform: none;
        white-space: normal; widows: 2; word-spacing: 0px;
        -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;
        display: inline !important; float: none;">s=-&nbsp;
        <span class="Apple-converted-space"><br>
          <br>
          Can I add <a class="moz-txt-link-rfc2396E" href="mailto:e=matthew@matthew.at">"e=matthew@matthew.at"</a> right after this?<br>
          <br>
          How about a "p=" line?<br>
          <br>
        </span></span>a=group:BUNDLE audio video<br>
      <br>
      Can I remove this before passing it back to setLocal Description?<br>
      <br>
      a=ice-options:google-ice <br>
      <br>
      How about removing this? Or changing it to
      "a=ice-options:matthew-special-ice"?<br>
      <br>
      a=sendrecv <br>
      <br>
      Can I change it to sendonly?<br>
      <br>
      a=rtcp-mux <br>
      <br>
      Remove this?<br>
      <br>
      a=crypto:1 AES_CM_128_HMAC_SHA1_80
      inline:66AIWBQ/R+UnMc4N/NP/b2hjEB6ydFMMCc8n3DMp <br>
      <br>
      Remove this? Change the key? Set up DTLS-SRTP instead?<br>
      <br>
      a=rtpmap:103 ISAC/16000 <br>
      <br>
      Change this to rtpmap: 110 ISAC/16000 because I prefer 110 as the
      codec ID?<br>
      <br>
      a=ssrc:1018881792 cname:B0VSPpORV75cCTIv <br>
      <br>
      Delete this? Change the ssrc ID because I don't like this one? or
      the cname?<br>
      <br>
      m=video 1 RTP/SAVPF 100 101 102 c=IN IP4 0.0.0.0 <br>
      <br>
      Change this to RTP/SAVP ? [Never mind that "IN IP4 0.0.0.0" is a
      strange thing to have coming from "createOffer"]<br>
      <br>
      NOWHERE in the JSEP specification can I find the answers to these
      questions. That makes working out what is allowed a
      browser-by-browser exhaustive search, which is unacceptable.<br>
      <br>
      Matthew Kaufman<br>
    </span>
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On 10/21/2012 01:20 AM, Bernard Aboba wrote:
> Harald said:
>
> "Thanks for the info - can you point to the bugs you filed on these 
> issues with our implementation? I'll try to see that they're followed up."
>
> [BA] To determine whether something is a "bug" or a "feature" it helps 
> to have a specification that states what is expected.  Since we're 
> primarily talking about SDP usage within the API rather than what 
> travels over the wire (after all, signaling is out of scope in 
> WebRTC), http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp is 
> only a baby step toward determining whether the SDP output by 
> createOffer is conformant or whether input to setLocalDescription or 
> setRemoteDescription should be acceptable or not, and if not, what 
> should happen.
>
> For example,  posting the SDP that is emitted by various createOffer() 
> implementations might enhance the appreciation of the importance of 
> the issue, I am not sure how to evaluate the results without 
> understanding whether that output is expected to be valid SDP or not. 
>  For example, if trickle ICE is used, then perhaps one should not 
> expect createOffer to provide SDP with valid addresses and ports 
> filled in; instead that information is expected to be filled in later.
Actually all of the issues I've seen in real testing so far have fallen 
into two broad cateories:

- Issues where the endpoints are known to be noninteroperable - for 
instance by requiring ICE, requiring SRTP, or using different video 
codecs. No amount of SDP munging will fix those.
- Issues where the Google implementors have failed to understand some 
subtlety of SDP syntax, encountered an SDP extension that we didn't 
expect to see, or didn't get around to handling that part of the syntax 
yet - for instance when we were sending "s=" instead of "s= ".

When we started out, I expected a host of issues of the type you 
envision - where people who are reading the SDP specs have serious 
differences of opinion about what should be the right outcome - but so 
far, I haven't seen that happening.

Perhaps we haven't gotten to the hard parts yet, or perhaps the SDP 
specification, when used for the basic stuff that we're so far doing, is 
less troublesome than we thought?

That's why I keep on pushing for bugs to be filed - I believe we can 
identify problem areas more easily by saying "when X tries to negotiate 
a connection with Y, A happens, but B should happen instead".
Also, I learn much more from those discussions than a discussion phrased 
in general terms!

                 Harald




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    <div class="moz-cite-prefix">On 10/21/2012 01:20 AM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU002-W136BDAECA1A38C4DA5FB5E593740@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">Harald said:
        <div>
          <div class="ecxmoz-cite-prefix"><br>
            "Thanks for the info - can you point to the bugs you filed
            on these issues with our implementation? I'll try to see
            that they're followed up."<br>
            <br>
            [BA] To determine whether something is a "bug" or a
            "feature" it helps to have a specification that states what
            is expected. &nbsp;Since we're primarily talking about SDP usage
            within the API rather than what travels over the wire (after
            all, signaling is out of scope in WebRTC),&nbsp;<a
              moz-do-not-send="true"
              href="http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp"
              style="font-size: 12pt; ">http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp</a>&nbsp;is
            only a baby step toward determining whether the SDP output
            by createOffer is conformant or whether input to
            setLocalDescription or setRemoteDescription should be
            acceptable or not, and if not, what should happen.&nbsp;</div>
        </div>
        <div class="ecxmoz-cite-prefix"><br>
        </div>
        <div class="ecxmoz-cite-prefix">For example, &nbsp;posting the SDP
          that is emitted by various createOffer() implementations might
          enhance the appreciation of the importance of the issue, I am
          not sure how to evaluate the results without understanding
          whether that output is expected to be valid SDP or not. &nbsp;For
          example, if trickle ICE is used, then perhaps one should not
          expect createOffer to provide SDP with valid addresses and
          ports filled in; instead that information is expected to be
          filled in later. <br>
        </div>
      </div>
    </blockquote>
    Actually all of the issues I've seen in real testing so far have
    fallen into two broad cateories:<br>
    <br>
    - Issues where the endpoints are known to be noninteroperable - for
    instance by requiring ICE, requiring SRTP, or using different video
    codecs. No amount of SDP munging will fix those.<br>
    - Issues where the Google implementors have failed to understand
    some subtlety of SDP syntax, encountered an SDP extension that we
    didn't expect to see, or didn't get around to handling that part of
    the syntax yet - for instance when we were sending "s=" instead of
    "s= ".<br>
    <br>
    When we started out, I expected a host of issues of the type you
    envision - where people who are reading the SDP specs have serious
    differences of opinion about what should be the right outcome - but
    so far, I haven't seen that happening.<br>
    <br>
    Perhaps we haven't gotten to the hard parts yet, or perhaps the SDP
    specification, when used for the basic stuff that we're so far
    doing, is less troublesome than we thought?<br>
    <br>
    That's why I keep on pushing for bugs to be filed - I believe we can
    identify problem areas more easily by saying "when X tries to
    negotiate a connection with Y, A happens, but B should happen
    instead".<br>
    Also, I learn much more from those discussions than a discussion
    phrased in general terms!<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------010505070800020706020604--

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Subject: Re: [rtcweb] (resend) RE: Draft agenda for RTCWeb session 2 at IETF85
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On 10/20/2012 07:23 PM, Matthew Kaufman wrote:
> Can we just save a bunch of time and have a show of hands at the
> beginning for:
>
> A) People who WILL NOT implement H.264 in their product and cannot be
> swayed by any presentation
>
> B) People who WILL NOT implement VP8 in their product and cannot be
> swayed by any presentation
>
> C) People who are still undecided enough that they want to hear the
> presentations

I think something like this makes sense. We could alternative do a show 
of hand for preference of either alternative, if it seems we're far away 
from any (rough) consensus then we could move on to discuss if we should 
abandon an MTI video codec, or select an alternative decision process.

And hopefully save a lot of time for SDP discussion.
>
> It is quite possible that once we get that out of the way, the rest of
> the presentations will be irrelevant and we can get back to discussing
> important things like who is going to write down the particulars for the
> "SDP profile" we're using as an API surface.
>
> Matthew Kaufman
>
> ________________________________________
>
> From: rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
> [rtcweb-bounces@ietf.org] on behalf of Cullen Jennings (fluffy)
> [fluffy@cisco.com]
>
> Sent: Friday, October 19, 2012 11:59 AM
>
> To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>
> Subject: [rtcweb] Draft agenda for RTCWeb session 2 at IETF85
>
> We have two WebRTC session at the next IETF. We have not worked out the
> agenda for the first session yet but our current plan is to dedicate the
> second session to the MTI video codec question.
>
> The draft agenda for this season is:
>
> Admin issues (5 min)
>
> IPR at the IETF ( 10 min )
>
> - this is a reminder of IETF rules and guidelines around IPR
>
> Presentations ( 80 min )
>
> - a 20 minute presentation from each of the following drafts
>
> draft-alvestrand-rtcweb-vp8-00
>
> draft-burman-rtcweb-h264-proposal-00
>
> draft-dbenham-webrtc-videomti-00
>
> draft-marjou-rtcweb-video-codec-00
>
> - If the authors of drafts recommending the same codec can consult with
> each other to avoid overlap, or even merge presentations, the recovered
> time would be given to the general discussion.
>
> General discussion (30 min )
>
> - general time for everyone to speak at the mic to express information
> important for the decision
>
> Call the question of which mandatory to implement video codec to select(
> 5 min )
>
> Next steps ( 20 min )
>
> Cullen, Magnus, and Ted <RTCWeb Chairs>
>
> _______________________________________________
>
> 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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--_310b2119-0692-45a8-b133-06d8fe2963a1_
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Harald said:=20

"When we started out=2C I expected a host of issues of the type you=0A=
    envision - where people who are reading the SDP specs have serious=0A=
    differences of opinion about what should be the right outcome - but=0A=
    so far=2C I haven't seen that happening."

[BA] Defining what SDP specs are relevant is not difficult.  draft-nandakum=
ar-rtcweb-sdp enumerates the relevant specifications to a good degree of ac=
curacy (I agree with 90+ percent of what is in the draft).   If all we had =
to do was quibble about which SDP references are MUST/SHOULD/MAY within tha=
t document=2C then with sufficient WG attention=2C the probability of conve=
rgence would be high.  However=2C this would not cover:

* Expectations for SDP parser resilience.  Experience indicates that this w=
as not sufficiently well described in RFC 3264 and 4566.  For example=2C my=
 understanding is that Chrome stops parsing in some cases when it hits SDP =
it does not understand.  While this approach is quite common in SDP impleme=
ntations (and may not violate 3264/4566 so as to be classifiable as a "bug"=
)=2C expectations for resilience could be better defined. =20
* Error handling.  What happens when the parser detects an error and how do=
 we know what it objected to?  Is an exception or error callback fired?  Do=
es this provide an indication of what SDP lines caused the error? Is there =
a way to retrieve the SDP that the browser decided to act upon (which may b=
e different from what was input)?=20
* Guidance with respect to API usage=2C both general and specific=2C such a=
s the questions Mathew mentioned.=20

Overall=2C these questions relate more to the API than to existing or in-pr=
ogress SDP specifications within IETF=2C which raises the question of where=
 they would be best addressed.=20

 		 	   		  =

--_310b2119-0692-45a8-b133-06d8fe2963a1_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Harald said: <br><br>"When we st=
arted out=2C I expected a host of issues of the type you=0A=
    envision - where people who are reading the SDP specs have serious=0A=
    differences of opinion about what should be the right outcome - but=0A=
    so far=2C I haven't seen that happening."<br><br>[BA] Defining what SDP=
 specs are relevant is not difficult.&nbsp=3B draft-nandakumar-rtcweb-sdp e=
numerates the relevant specifications to a good degree of accuracy (I agree=
 with 90+ percent of what is in the draft).&nbsp=3B&nbsp=3B If all we had t=
o do was quibble about which SDP references are MUST/SHOULD/MAY within that=
 document=2C then with sufficient WG attention=2C the probability of conver=
gence would be high.&nbsp=3B However=2C this would not cover:<br><br>* Expe=
ctations for SDP parser resilience.&nbsp=3B Experience indicates that this =
was not sufficiently well described in RFC 3264 and 4566.&nbsp=3B For examp=
le=2C my understanding is that Chrome stops parsing in some cases when it h=
its SDP it does not understand.&nbsp=3B While this approach is quite common=
 in SDP implementations (and may not violate 3264/4566 so as to be classifi=
able as a "bug")=2C expectations for resilience could be better defined.&nb=
sp=3B <br>* Error handling.&nbsp=3B What happens when the parser detects an=
 error and how do we know what it objected to?&nbsp=3B Is an exception or e=
rror callback fired?&nbsp=3B Does this provide an indication of what SDP li=
nes caused the error? Is there a way to retrieve the SDP that the browser d=
ecided to act upon (which may be different from what was input)? <br>* Guid=
ance with respect to API usage=2C both general and specific=2C such as the =
questions Mathew mentioned. <br><br>Overall=2C these questions relate more =
to the API than to existing or in-progress SDP specifications within IETF=
=2C which raises the question of where they would be best addressed. <br><b=
r> 		 	   		  </div></body>
</html>=

--_310b2119-0692-45a8-b133-06d8fe2963a1_--

From ron@debian.org  Sun Oct 21 12:57:56 2012
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Hi Matthew,

On Sat, Oct 20, 2012 at 10:26:05AM -0700, Matthew Kaufman wrote:
> Harald Alvestrand [harald@alvestrand.no]*:*
> 
> >At the moment, if we want open, good and  developed in cooperation
> >with a standards body, it seems that we only
> get 2 out of 3. At most.
> 
> For video, H.264 is available in open source implementations and the
> specification is available to read by all, the quality is more than
> good enough, and it was developed in cooperation with a recognized
> standards body.
> 
> I think you're looking for a word other than "open" to explain what
> you might not like about it... but the alternative you've proposed
> meets even fewer of these, and I would claim that the last is more
> important than the missing one you're thinking of.

I'm not sure which dictionary you're taking your definition of "open"
from here, but I can't seem to find one that constrains it to just the
closed sophisticated meanings that you seem to want to dilute it to.
Do you have a reference or suggested alternative word that you'd like
to share with us?

"Requires me to apply for a licence, without which I have no rights"
is pretty much the poster child for the exact opposite of open ...

"Can peep through the window at what I'm missing without one" is not
an acceptable or even approximate substitute by any measure.


It seems pretty clear to me that Open and Good, as judged by this group,
are the only two actual relevant constraints here.  The third is about
as arbitrary as saying "eligible codecs must have a V in their name"
in its current effect on limiting the scope of our choices.

Having the imprimatur of a closed shop organisation is quite irrelevant
to meeting the real demands of developers and end users.  Suggesting it
is somehow essential seems like the sort of conceit that led people to
gather around a more open organisation to look for alternative solutions
in the first place.

Yes, it would have been wonderful if RFC 6386 had been Standards Track
instead of Informational, and if we'd had a working group to oversee
that several years ago.  But that doesn't make it any less of an existing
technology, or any less worthy of consideration for its applicability to
the goals of this group.

We should judge it on technical merits, not its political endorsements.
An inability to deploy something, except at the whim of a private group
of stake-holders, is a pretty insurmountable technical barrier.  That
some other standards body might endorse or even encourage that practice
probably only discounts the value of their approval to us even further.

If the owners of H.264 aren't prepared to remove that barrier to at least
the same extent that the owners of VP8 have, then I don't know why we are
wasting our time still pretending H.264 is even an option here.  If they
are serious about being a viable candidate, then they know what they need
to do and have had plenty of time to do it.

Just redefining the word "open" isn't the solution we're looking for.


 Openly,
 Ron



From ron@debian.org  Sun Oct 21 14:01:51 2012
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From: Ron <ron@debian.org>
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Subject: Re: [rtcweb] (resend) RE: Draft agenda for RTCWeb session 2 at IETF85
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On Sat, Oct 20, 2012 at 10:23:20AM -0700, Matthew Kaufman wrote:
> Can we just save a bunch of time and have a show of hands at the
> beginning for:
> 
> A) People who WILL NOT implement H.264 in their product and cannot
> be swayed by any presentation

Are there actually people saying they *will not* implement H.264?

I can't say that I've seen that here.  The problem being repeatedly
voiced is that under its current licencing regime they simply *CANNOT*
implement and/or distribute it with their products.

> B) People who WILL NOT implement VP8 in their product and cannot be
> swayed by any presentation

That constraint does not apply to VP8, so turning this into a tug of
war between the WILL NOTs and the CANNOTs, doesn't seem to really
address the actual problem we have before us.

> C) People who are still undecided enough that they want to hear the
> presentations

Unless the presentations are going to announce new perpetual RF
licencing conditions for H.264, I don't really see what new facts
might be presented that can change the reality of:

"People invested in H.264 licences or owning its IPR prefer H.264
 and the exclusivity and revenue streams it brings them."

"People who don't qualify for H.264 licences will be screwed unless
 we pick VP8 as the baseline - and will need to either fork RTCWeb,
 or adopt or create an alternative to it."

This isn't a coin toss between choices of equivalent merit.
Framing the question as one of Wills completely ignores the salient
technical point of which choice would give RTCWeb relevance to the
broadest range of possible developers, adopters and users.


I don't want to see discussion of people's real concerns suppressed,
so I think I agree that it would be a great idea for people to begin
their presentations here and now, with the aim of distilling that to
some final brief summations by the time of the meeting.

So far as I can see this is a technical no-brainer.  But not because
I have any deep set preference for one or the other, or an inability
to be swayed.  Simply because H.264 would be *impossible* for many
users to deploy.  That's very different to VP8 being undesirable to
deploy by people who have a competing interest in other codecs.

Reducing that to a question of WILLs is a gross oversimplification
that isn't really very helpful to making a Good decision here or to
showing the careful reasoning behind it.

  Simply,
  Ron



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Subject: Re: [rtcweb] (resend) RE: Draft agenda for RTCWeb session 2 at IETF85
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On Oct 21, 2012, at 8:44 AM, Stefan Hakansson LK =
<stefan.lk.hakansson@ericsson.com> wrote:

> On 10/20/2012 07:23 PM, Matthew Kaufman wrote:
>> Can we just save a bunch of time and have a show of hands at the
>> beginning for:
>>=20
>> A) People who WILL NOT implement H.264 in their product and cannot be
>> swayed by any presentation
>>=20
>> B) People who WILL NOT implement VP8 in their product and cannot be
>> swayed by any presentation
>>=20
>> C) People who are still undecided enough that they want to hear the
>> presentations
>=20
> I think something like this makes sense. We could alternative do a =
show of hand for preference of either alternative, if it seems we're far =
away from any (rough) consensus then we could move on to discuss if we =
should abandon an MTI video codec, or select an alternative decision =
process.

I'd like to point out that rough show of hands was done at a previous =
meeting and it seemed like getting to agreement on a MTI video codec was =
possible.=20

On the question proposed, I don't think this is the right question. The =
right questions would be of the people that want present information one =
way or the other, how many of them think nothing they say will change =
the opinions of anyone in the room. If they feel this way, I'm sure the =
presentations will be short and no one will line up at the mic to =
discuss it. My best guess is that the will be people at the front of the =
room, and at the mic, that feel they have something to say on the topic.=20=


Keep in mind this approach was discussed at the last meeting an there =
seemed to an awful lot of people that felt this was important. We have =
spent an incredible amount of WG meeting time trying to get to the point =
were we can make have a meeting where we try and come to consensus on =
this.


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] RTCWEB Session 2:  A plea for brevity
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Subject: Re: [rtcweb] RTCWEB Session 2:  A plea for brevity
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On Oct 20, 2012, at 10:13 AM, Bernard Aboba <bernard_aboba@hotmail.com> wro=
te:

> Cullen Jennings said:
>=20
> "One of the drafts says when making measurements on quality they found A =
was
> better than B while the another draft says B was better than A.  Several
> people have asked me how this could happen two different groups ended up
> with opposite measurements and they want to understand more about how thi=
s
> could have happened and which is correct.... Let me ask you guys a questi=
on,
> how many people do you think will be in the microphone line up when these
> two are discussed?""
>=20
> [BA] Assuming those at the mic would be asking clarifying questions, why =
do
> we need to wait until IETF 85 to begin that discussion?   If each side we=
re
> to start a thread on the list, at least some of the clarifying questions
> could be gotten out of the way beforehand.  Who knows, the result could b=
e
> akin to letting a child stay up late on Saturday night to watch TV - the
> little devil might be too tired to cause trouble at church on Sunday :)
>=20

+1 for starting list conversation before the meeting =85 It will be a bit c=
runched as many people are still working on -01 drafts till monday, then ha=
ve a few days to read drafts and travel to Europe for W3C meeting then go s=
traight from W3C meeting to IETF. I really hope there can be some good list=
 discussion ahead of time.=20
=20




From fluffy@iii.ca  Sun Oct 21 16:42:09 2012
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From: Cullen Jennings <fluffy@iii.ca>
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On Oct 21, 2012, at 9:10 AM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

> Harald said:=20
>=20
> "When we started out, I expected a host of issues of the type you =
envision - where people who are reading the SDP specs have serious =
differences of opinion about what should be the right outcome - but so =
far, I haven't seen that happening."
>=20
> [BA] Defining what SDP specs are relevant is not difficult.  =
draft-nandakumar-rtcweb-sdp enumerates the relevant specifications to a =
good degree of accuracy (I agree with 90+ percent of what is in the =
draft).   If all we had to do was quibble about which SDP references are =
MUST/SHOULD/MAY within that document, then with sufficient WG attention, =
the probability of convergence would be high.  However, this would not =
cover:
>=20
> * Expectations for SDP parser resilience.  Experience indicates that =
this was not sufficiently well described in RFC 3264 and 4566.  For =
example, my understanding is that Chrome stops parsing in some cases =
when it hits SDP it does not understand.  While this approach is quite =
common in SDP implementations (and may not violate 3264/4566 so as to be =
classifiable as a "bug"), expectations for resilience could be better =
defined. =20

We could put some implementer notes in one of the drafts to try and help =
cover some of the common things developers need to deal with. That would =
be helpful for many things that use SDP. Having some example SDP that =
included "bogus future extensions" that a parse should successfully =
ignore might help people test their SDP implementations. Note that =
unlike the IETF, the W3C tends to have have a conformance test suite to =
deal with things like this so in some weird way, the W3C might end up =
improving the interoperability of the SDP.=20

> * Error handling.  What happens when the parser detects an error and =
how do we know what it objected to?  Is an exception or error callback =
fired?  Does this provide an indication of what SDP lines caused the =
error? Is there a way to retrieve the SDP that the browser decided to =
act upon (which may be different from what was input)?=20

I think the combination of the JSEP and WebRTC spec need to get very =
clear on all the issues you raised. I'd like to see error handling a =
major part of the discussion at the W3C meeting as we have not really =
tacked that yet in the spec. Last time I looked, no one had replied to =
Anant's proposal.=20

> * Guidance with respect to API usage, both general and specific, such =
as the questions Mathew mentioned.=20

The issue of what can be changed that Mathew raised has been discussed =
as an open issue and we know that we need to nail that down in the =
specification. It seemed like one of the things that could be left till =
a bit later in the process when we had more information about what the =
use cases were and what to put. The one use case that has been raised =
very clearly is people want to be able to remove a particular codec for =
various reasons. I suspect that lots of people will argue to make sure =
that type of change of SDP is supported. I'm not sure what use case =
drive the rest but 100% agree this is a topic we need to put some time =
into.  Right now, it does not seem to be holding up implications which =
are still trying to nail down all the basic stuff along with ICE, DTLS, =
etc.=20

>=20
> Overall, these questions relate more to the API than to existing or =
in-progress SDP specifications within IETF, which raises the question of =
where they would be best addressed.=20

I suspect most of them are in the WebRTC spec and some of them in ietf =
drafts but either way, I agree all the points you raised need to be =
specified.=20

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


From john@jlc.net  Sun Oct 21 20:04:31 2012
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From: John Leslie <john@jlc.net>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] RTCWEB Session 2:  A plea for brevity
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Cullen Jennings <fluffy@iii.ca> wrote:
> 
> The MTI video codec is a pretty big questions we are asking and my
> observation about getting consensus is that people need to have been
> given the time to make their key argument or it is very hard to get
> to consensus.

   Some arguments are endless... :^(

> I realize that some people believe, or even hope, that we will not
> get to consensus on this topic

   "or even hope" wasn't necessary!

> so it is all a waste of time but the WG has made it clear it does
> want to get to have a MTI video codec. 

   Absent agreement on _which_ video codec will be MTI, that's a dead
end.

> So consider the situation we are in. One of the drafts says when
> making measurements on quality they found A was better than B while
> the another draft says B was better than A.

   Welcome to research-land!

> Several people have asked me how this could happen two different groups
> ended up with opposite measurements and they want to understand more
> about how this could have happened and which is correct. 

   Not I...

> Let me ask you guys a question, how many people do you think will be
> in the microphone line up when these two are discussed?

   Say four (more than once apiece, of course)...

> You have both seen plenty of controversial topics discussed at other
> ietf meetings. What would you recommend to people making slides for
> a slot like this - how many slides do you think they can reasonably
> get through in a time slot like this?

   Depends on the slides!

> What would your best guess be of how long we need to make these slots
> such that the chairs don't have to cut off the mic lineup? 

   45 minutes. (Myself, I'd rather you cut the line at 10 minutes.)

> I know you could take these as vaguely cynical rhetorical questions
> but I don't mean them that way. I'm straight up asking you how much
> time you think we need such that authors of the drafts can make their
> key argument and people that have questions or disagreements that
> they think are important for the WG to understand can make ask their
> question.

   Zero minutes, IMHO: these questions should be on the list.

   Some large companies with deep pockets have every reason to prefer
the H.264 licensing paragigm. Some small organizations doing open-source
have every reason to prefer the AV8 paradigm. The actual decisions will
be made by folks won't be in the room.

   Those decisions won't be driven by which codec is better: they'll
be driven by what _risks_ are taken by implementing one or the other.

   I'd like to air a mostly-serious way-out.

   Both H.264 and AV8 deserve a SHOULD implement. Some implementors
_will_ chicken out due to the risks, waiting to see what the market
demands.

   Mandatory-to-Implement isn't about which codec is best _today_ --
it's about a lowest-common-denominator for years to come. We could
perfectly well designate as Mandatory-to-Implement something so
brain-dead that nobody could be bothered to sue for patent infringement.
H.261 comes to mind -- more than 20 years old, can be implemented in
20-year-old software. Prove that you used 20-year-old software line-
for-line, and no patent troll will want to let it get to court.

   The tussle between H.264 and AV8 deserves to play out in the
marketplace, not in an IETF meeting-room. I don't believe any IETF
RFC _can_ settle this issue.

   YMMV, of course...

--
John Leslie <john@jlc.net>

From matthew@matthew.at  Sun Oct 21 20:50:15 2012
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Subject: Re: [rtcweb] (resend) RE: Draft agenda for RTCWeb session 2 at IETF85
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On 10/21/2012 2:01 PM, Ron wrote:
> Unless the presentations are going to announce new perpetual RF
> licencing conditions for H.264, I don't really see what new facts
> might be presented that can change the reality of:
>
> "People invested in H.264 licences or owning its IPR prefer H.264
>   and the exclusivity and revenue streams it brings them."
>
> "People who don't qualify for H.264 licences will be screwed unless
>   we pick VP8 as the baseline - and will need to either fork RTCWeb,
>   or adopt or create an alternative to it."

If only it were this simple. My employer spends more on H.264 licensing 
than it receives in return, and it isn't about "exclusivity" or "revenue 
streams" ( 
http://blogs.msdn.com/b/ie/archive/2010/05/03/follow-up-on-html5-video-in-ie9.aspx 
)

John Leslie said:

"The actual decisions will
be made by folks [who] won't be in the room.

    Those decisions won't be driven by which codec is better: they'll
be driven by what _risks_ are taken by implementing one or the other."

And this is a much more accurate way of saying it... and why, for anyone 
in the room actually on the "implementation" side, there's not much 
point in continuing to debate the potential merits unless the landscape 
is *significantly* changed from its current state (for example, RF 
licenses for H.264 or total indemnification provided by VP8's owner.)

Matthew Kaufman


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On 10/21/2012 4:41 PM, Cullen Jennings wrote:
>
> The issue of what can be changed that Mathew raised has been discussed as an open issue and we know that we need to nail that down in the specification. It seemed like one of the things that could be left till a bit later in the process when we had more information about what the use cases were and what to put. The one use case that has been raised very clearly is people want to be able to remove a particular codec for various reasons. I suspect that lots of people will argue to make sure that type of change of SDP is supported. I'm not sure what use case drive the rest but 100% agree this is a topic we need to put some time into.  Right now, it does not seem to be holding up implications which are still trying to nail down all the basic stuff along with ICE, DTLS, etc.
>

The API surface exposed in the browser must be *completely* specified. 
While I am glad that you "100% agree this is a topic we need to put some 
time into," I also know that in the years since SDP O/A was first 
proposed a concrete set of interoperability specifications has yet to 
materialize, and so it seems unlikely that this WG or the W3C WG will 
manage to fully document what changes to the SDP emitted by createOffer 
and the SDP accepted by setLocalDescription and setRemoteDescription are 
permitted without a *significant* effort.

I will note that the JSEP draft says that calling createOffer is 
optional, and so it must be possible to, with only access to the 
specification and the referenced specifications, generate a blob of SDP 
which is accepted by setLocalDescription in *every* browser 
implementation. We are nowhere near that point.

All of this is of course easily avoided by choosing an API surface which 
is not an underspecified blob of SDP.

Matthew Kaufman

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On Wed, Oct 17, 2012 at 12:20 PM, Ron <ron@debian.org> wrote:

>
> Hi Erik,
>
> On Wed, Oct 17, 2012 at 01:56:55AM -0700, Erik Lagerway wrote:
> > Thought it important to point out the fact that FREE-P8 (VP8) would
> likely
> > be the preferred codec implemented by independent developers. Royalties
> and
> > admin headaches create a barrier to entry. Independent dev shops equate
> to
> > a great deal of innovation and some of these have been responsible for
> > dictating the standard in market with hundreds of millions of users. Yes,
> > there is plenty of interoperability with 264 today but that doesn't mean
> it
> > will be that way tomorrow.
>
> Very true.  Though for developers who don't click-track and count users
> of their software it is more than just preferred.  An RF solution is
> actually the only viable way for this standard to be accessible to them.
>
> Any standard that immediately excludes a large portion of potential
> developers and adopters can only ever be of niche value and is certain
> to be replaced before long by an alternative that doesn't have that
> constraint as a fundamental flaw preventing its broadest use.
>
> If the whole point of this group was to avoid further fracturing of
> the developer efforts in this space, then solving this problem now
> would be much preferable to forcing another group to develop another
> competing standard that does address that issue.  It would be a sad
> waste of the invested time of the people here if we didn't recognise
> this necessity.
>
> I think the points you make above are very relevant here.
>
>
> > Can't we do both, as we have done for Audio MTI?
>
> Well, the M in MTI stands for Mandatory.  So mandating both would be
> mandating the barriers to entry that you so clearly identified above?
>
> Mandating an RF video codec, and permitting any other to be negotiated
> as an extension would be more like what we decided was best for audio.
>

Yes, I see your point and agree.


>
>
>
>
> > RTCweb / WebRTC is supposed to be the next big thing, why hamper
> > innovation like this?
>
> Indeed we should not, and I for one certainly hope consensus is
> that we will not.
>

Seems to me that if we want to see the indie webrtc developers flourish, we
need to implement a RF codec = VP8.  Unless of course the MPEG LA decides
to grant a RF license on 264/AVC before the WG votes on the MTI? Which does
not seem likely given the time frame. That plus the fact this 264/AVC non
royalty codec concept was tabled in the past and went nowhere, it seems.

I would like to see us foster grassroots innovation by making this tech
accessible to all developers, which essentially means zero royalty and
admin overhead.


>
>
>  Stay sharp!
>  Ron
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d0420a695dbdfd304cca7a735
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<div class=3D"gmail_quote">On Wed, Oct 17, 2012 at 12:20 PM, Ron <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ron@debian.org" target=3D"_blank">ron@debian=
.org</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">
<br>
Hi Erik,<br>
<div class=3D"im"><br>
On Wed, Oct 17, 2012 at 01:56:55AM -0700, Erik Lagerway wrote:<br>
&gt; Thought it important to point out the fact that FREE-P8 (VP8) would li=
kely<br>
&gt; be the preferred codec implemented by independent developers. Royaltie=
s and<br>
&gt; admin headaches create a barrier to entry. Independent dev shops equat=
e to<br>
&gt; a great deal of innovation and some of these have been responsible for=
<br>
&gt; dictating the standard in market with hundreds of millions of users. Y=
es,<br>
&gt; there is plenty of interoperability with 264 today but that doesn&#39;=
t mean it<br>
&gt; will be that way tomorrow.<br>
<br>
</div>Very true. =A0Though for developers who don&#39;t click-track and cou=
nt users<br>
of their software it is more than just preferred. =A0An RF solution is<br>
actually the only viable way for this standard to be accessible to them.<br=
>
<br>
Any standard that immediately excludes a large portion of potential<br>
developers and adopters can only ever be of niche value and is certain<br>
to be replaced before long by an alternative that doesn&#39;t have that<br>
constraint as a fundamental flaw preventing its broadest use.<br>
<br>
If the whole point of this group was to avoid further fracturing of<br>
the developer efforts in this space, then solving this problem now<br>
would be much preferable to forcing another group to develop another<br>
competing standard that does address that issue. =A0It would be a sad<br>
waste of the invested time of the people here if we didn&#39;t recognise<br=
>
this necessity.<br>
<br>
I think the points you make above are very relevant here.<br>
<div class=3D"im"><br>
<br>
&gt; Can&#39;t we do both, as we have done for Audio MTI?<br>
<br>
</div>Well, the M in MTI stands for Mandatory. =A0So mandating both would b=
e<br>
mandating the barriers to entry that you so clearly identified above?<br>
<br>
Mandating an RF video codec, and permitting any other to be negotiated<br>
as an extension would be more like what we decided was best for audio.<br><=
/blockquote><div><br>Yes, I see your point and agree.<br>=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">

<div class=3D"im"><br>
</div><br><div class=3D"im"><br>
<br>
&gt; RTCweb / WebRTC is supposed to be the next big thing, why hamper<br>
&gt; innovation like this?<br>
<br>
</div>Indeed we should not, and I for one certainly hope consensus is<br>
that we will not.<br></blockquote><div><br>Seems to me that if we want to s=
ee the indie webrtc developers flourish,
 we need to implement a RF codec =3D VP8.=A0 Unless of course the MPEG LA=
=20
decides to grant a RF license on 264/AVC before the WG votes on the=20
MTI? Which does not seem likely given the time frame. That plus the fact th=
is 264/AVC non royalty codec concept was tabled in the past and went nowher=
e, it seems.<br><br>I would like to see us foster grassroots innovation by =
making this tech accessible to all developers, which essentially means zero=
 royalty and admin overhead. <br>
=A0<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex">
<br>
<br>
=A0Stay sharp!<br>
<span class=3D"HOEnZb"><font color=3D"#888888">=A0Ron<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><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>

--f46d0420a695dbdfd304cca7a735--

From internet-drafts@ietf.org  Mon Oct 22 09:27:19 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : RTCWEB Security Architecture
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-04.txt
	Pages           : 37
	Date            : 2012-10-22

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (e.g
   JavaScript).  The major use cases for RTCWEB technology are real-time
   audio and/or video calls, Web conferencing, and direct data transfer.
   Unlike most conventional real-time systems (e.g., SIP-based soft
   phones) RTCWEB communications are directly controlled by some Web
   server, which poses new security challenges.  For instance, a Web
   browser might expose a JavaScript API which allows a server to place
   a video call.  Unrestricted access to such an API would allow any
   site which a user visited to "bug" a user's computer, capturing any
   activity which passed in front of their camera.  [I-D.ietf-rtcweb-
   security] defines the RTCWEB threat model.  This document defines an
   architecture which provides security within that threat model.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-security-arch-04


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


From ekr@rtfm.com  Mon Oct 22 09:35:20 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 22 Oct 2012 09:34:33 -0700
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Please ignore this version. I had a version control hiccup. Will post the
correct version in a few minutes after I unscrew things.

-Ekr


On Mon, Oct 22, 2012 at 9:27 AM,  <internet-drafts@ietf.org> wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>         Title           : RTCWEB Security Architecture
>         Author(s)       : Eric Rescorla
>         Filename        : draft-ietf-rtcweb-security-arch-04.txt
>         Pages           : 37
>         Date            : 2012-10-22
>
> Abstract:
>    The Real-Time Communications on the Web (RTCWEB) working group is
>    tasked with standardizing protocols for enabling real-time
>    communications within user-agents using web technologies (e.g
>    JavaScript).  The major use cases for RTCWEB technology are real-time
>    audio and/or video calls, Web conferencing, and direct data transfer.
>    Unlike most conventional real-time systems (e.g., SIP-based soft
>    phones) RTCWEB communications are directly controlled by some Web
>    server, which poses new security challenges.  For instance, a Web
>    browser might expose a JavaScript API which allows a server to place
>    a video call.  Unrestricted access to such an API would allow any
>    site which a user visited to "bug" a user's computer, capturing any
>    activity which passed in front of their camera.  [I-D.ietf-rtcweb-
>    security] defines the RTCWEB threat model.  This document defines an
>    architecture which provides security within that threat model.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-security-arch-04
>
>
> 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

From internet-drafts@ietf.org  Mon Oct 22 10:13:19 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : RTCWEB Security Architecture
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-05.txt
	Pages           : 38
	Date            : 2012-10-22

Abstract:
   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (e.g
   JavaScript).  The major use cases for RTCWEB technology are real-time
   audio and/or video calls, Web conferencing, and direct data transfer.
   Unlike most conventional real-time systems (e.g., SIP-based soft
   phones) RTCWEB communications are directly controlled by some Web
   server, which poses new security challenges.  For instance, a Web
   browser might expose a JavaScript API which allows a server to place
   a video call.  Unrestricted access to such an API would allow any
   site which a user visited to "bug" a user's computer, capturing any
   activity which passed in front of their camera.  [I-D.ietf-rtcweb-
   security] defines the RTCWEB threat model.  This document defines an
   architecture which provides security within that threat model.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-security-arch-05


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


From internet-drafts@ietf.org  Mon Oct 22 14:32:42 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-05.txt
	Pages           : 61
	Date            : 2012-10-22

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-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-05


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


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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : RTCWeb Datagram Connection
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-02.txt
	Pages           : 12
	Date            : 2012-10-22

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-channel-02


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


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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Javascript Session Establishment Protocol
	Author(s)       : Justin Uberti
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-jsep-02.txt
	Pages           : 33
	Date            : 2012-10-22

Abstract:
   This document proposes a mechanism for allowing a Javascript
   application to fully control the signaling plane of a multimedia
   session, and discusses how this would work with existing signaling
   protocols.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-02


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


From ron@debian.org  Mon Oct 22 18:18:23 2012
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From: Ron <ron@debian.org>
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On Sun, Oct 21, 2012 at 08:50:11PM -0700, Matthew Kaufman wrote:
> On 10/21/2012 2:01 PM, Ron wrote:
> >Unless the presentations are going to announce new perpetual RF
> >licencing conditions for H.264, I don't really see what new facts
> >might be presented that can change the reality of:
> >
> >"People invested in H.264 licences or owning its IPR prefer H.264
> >  and the exclusivity and revenue streams it brings them."
> >
> >"People who don't qualify for H.264 licences will be screwed unless
> >  we pick VP8 as the baseline - and will need to either fork RTCWeb,
> >  or adopt or create an alternative to it."
> 
> If only it were this simple. My employer spends more on H.264
> licensing than it receives in return, and it isn't about
> "exclusivity" or "revenue streams"

It is fairly normal to need to pay for exclusivity :)

It certainly means that I could not ship a Free RTCWeb implementation
with H.264 as MTI to the same volume of users that your employer might.
If the cost of simply tracking all those users didn't bankrupt me all
by itself, then the licence fees certainly would try to finish the job.

That sounds like exclusivity to me.  If not a little Deja Vu of the
good old days of the First Browser Wars ...


> John Leslie said:
> 
> "The actual decisions will
> be made by folks [who] won't be in the room.
> 
>    Those decisions won't be driven by which codec is better: they'll
> be driven by what _risks_ are taken by implementing one or the other."
> 
> And this is a much more accurate way of saying it... and why, for
> anyone in the room actually on the "implementation" side, there's
> not much point in continuing to debate the potential merits unless
> the landscape is *significantly* changed from its current state (for
> example, RF licenses for H.264 or total indemnification provided by
> VP8's owner.)

I don't really understand this repeated insistence that VP8 is somehow
more risky unless it comes with Total Indemnification?

H.264 certainly doesn't come with total indemnification, and MPEG-LA
is very clear that absolutely nothing which they licence comes with
even token indemnification from them or anyone else - or even with a
licence from all the people you might actually need a licence from.

This requirement is totally unprecedented, and such offers are near
to totally non-existent in the real world.  The landscape would most
certainly be changed if we woke up to find it covered in unicorns,
but I don't think anybody would take that seriously as a technical
prerequisite for some candidate technology.


The reality today is that H.264 is currently being prosecuted in the
courts over infringement claims -- while the only apparent threat to
VP8 is a mythical pool, that hasn't actually been formed, that hasn't
actually issued any licencing demands, and that the people who wanted
to form it have refused to say anything about for over a year, since
they last mumbled "yeah, some people responded.  really they did."

H.264 being released under an RF licence would indeed be a game-changer,
though even if that happened, on the evidence at hand today it is still
clearly the more patently risky (no pun intended) of the two unless you
happen to be one of its IPR holders (and even then, so it would seem).

If the current lawsuit against H.264 had instead been prosecuted against
a user of VP8, the prosecuting party would have lost their licence to
use VP8.  No such protection exists for me if I use H.264.

So the balance of both risk and protection would already seem tilted
in favour of VP8 too.  If anyone should be required to provide a total
indemnity, then surely H.264 is in more evident and urgent need of such
a thing, given the risk to it is already *known* to not be hypothetical.


What am I missing that means VP8 should need to jump over an even
higher bar for acceptance in this respect?  So far as I can see it
already has, hasn't it?


 Cheers,
 Ron



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--_1f65d363-6809-4b73-beaf-11e0a248ed8f_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Ron said:=20
> If the cost of simply tracking all those users didn't bankrupt me all
> by itself=2C then the licence fees certainly would try to finish the job.

[BA]  Pardon me if I'm obtuse=2C but I'd like to better understand the situ=
ation that is resulting in your being concerned about having to pay license=
 fees and track users.=20

Are you developing HTML5 applications?  f so=2C are you envisaging that as =
an HTML5 developer writing a WebRTC application for a browser that supports=
 H.264 you would need to pay licensing fees?=20

Are you working for a browser vendor?  If so=2C are you envisaging that you=
 would need to pay licensing fees based on the number of users of your brow=
ser?=20

Are you a developer of applications looking to interoperate with WebRTC?  I=
f so=2C are you calling APIs that enable you to use the H.264 implementatio=
n in the OS platform=2C or accessing the hardware encode/decode functionali=
ty (available on most SoC platforms)?  If so=2C do you believe that you nee=
d to pay license fees?=20

Or do you primarily develop on platforms without H.264 support in the OS=2C=
 and on hardware with no H.264 encode/decode support? =20


 		 	   		  =

--_1f65d363-6809-4b73-beaf-11e0a248ed8f_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Ron said: <br><div>&gt=3B If the=
 cost of simply tracking all those users didn't bankrupt me all<br>&gt=3B b=
y itself=2C then the licence fees certainly would try to finish the job.<br=
><br>[BA]&nbsp=3B Pardon me if I'm obtuse=2C but I'd like to better underst=
and the situation that is resulting in your being concerned about having to=
 pay license fees and track users. <br><br>Are you developing HTML5 applica=
tions?&nbsp=3B f so=2C are you envisaging that as an HTML5 developer writin=
g a WebRTC application for a browser that supports H.264 you would need to =
pay licensing fees? <br><br>Are you working for a browser vendor?&nbsp=3B I=
f so=2C are you envisaging that you would need to pay licensing fees based =
on the number of users of your browser? <br><br>Are you a developer of appl=
ications looking to interoperate with WebRTC?&nbsp=3B If so=2C are you call=
ing APIs that enable you to use the H.264 implementation in the OS platform=
=2C or accessing the hardware encode/decode functionality (available on mos=
t SoC platforms)?&nbsp=3B If so=2C do you believe that you need to pay lice=
nse fees? <br><br>Or do you primarily develop on platforms without H.264 su=
pport in the OS=2C and on hardware with no H.264 encode/decode support?&nbs=
p=3B <br><br><br></div> 		 	   		  </div></body>
</html>=

--_1f65d363-6809-4b73-beaf-11e0a248ed8f_--

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#1: RFC 4734 citation

 Section 3 states:

    WebRTC clients are REQUIRED to implement the following audio codecs.

    o  Opus [RFC6716], with any ptime value up to 120 ms

    o  G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
       ptime of 20 - see section 4.5.14 of [RFC3551]

    o  Telephone Event - [RFC4734]

 [BA] I believe the citation should be to "RTP Payload for DTMF Digits,
 Telephony Tones, and Telephony Signals" [RFC4733], not to "Definition of
 Events for Modem, Fax, and Text Telephony Signals" [RFC4734].

 If the desire is in fact to mandate support for RFC 4734, please explain
 why it is necessary for browsers to include support for digital encoding
 of modem and fax signals.

-- 
--------------------------------+------------------------
 Reporter:  bernard_aboba@…     |      Owner:  jmvalin@…
     Type:  defect              |     Status:  new
 Priority:  critical            |  Milestone:  milestone1
Component:  audio               |    Version:  1.0
 Severity:  Active WG Document  |   Keywords:
--------------------------------+------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/1>
rtcweb <http://tools.ietf.org/rtcweb/>


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#2: Section 7.1

 Section 7.1 states:

    The circuit breaker defines a conservative boundary condition for safe
 operation,
    chosen such that applications that trigger the circuit breaker will
    almost certainly be causing severe network congestion.

 [BA] While it is true that when circuit breakers are triggered it is
 likely that very severe network congestion is occurring along the path,
 severe network congestion (or even a congestive collapse) will not
 necessarily trigger the circuit breakers. For example, one of the circuit
 breakers is only triggered if *no* packets are received; even in a
 congestive collapse, *some* packets will get through. Therefore the lack
 of circuit breaker triggering does not imply that an application is
 behaving "conservatively" or safely.  Recommend change to:

 "Circuit breakers are designed to enable applications to recognize and
 react to situations of extreme network congestion.  However, since circuit
 breakers may not be triggered until congestion becomes extreme, circuit
 breakers are not a substitute for congestion control and applications
 implementing circuit breakers cannot be considered "conservative" or
 "safe"."

-- 
--------------------------------+------------------------
 Reporter:  bernard_aboba@…     |      Owner:  csp@…
     Type:  defect              |     Status:  new
 Priority:  major               |  Milestone:  milestone1
Component:  rtp-usage           |    Version:  1.0
 Severity:  Active WG Document  |   Keywords:
--------------------------------+------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/2>
rtcweb <http://tools.ietf.org/rtcweb/>


From randell-ietf@jesup.org  Tue Oct 23 06:39:36 2012
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On 10/22/2012 6:36 PM, rtcweb issue tracker wrote:
> #1: RFC 4734 citation
>
>   Section 3 states:
>
>      WebRTC clients are REQUIRED to implement the following audio codecs.
>
>      o  Opus [RFC6716], with any ptime value up to 120 ms
>
>      o  G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
>         ptime of 20 - see section 4.5.14 of [RFC3551]
>
>      o  Telephone Event - [RFC4734]
>
>   [BA] I believe the citation should be to "RTP Payload for DTMF Digits,
>   Telephony Tones, and Telephony Signals" [RFC4733], not to "Definition of
>   Events for Modem, Fax, and Text Telephony Signals" [RFC4734].
>
>   If the desire is in fact to mandate support for RFC 4734, please explain
>   why it is necessary for browsers to include support for digital encoding
>   of modem and fax signals.
>

I'm certain the intent was RFC 4733.

-- 
Randell Jesup
randell-ietf@jesup.org


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To: "bernard_aboba@hotmail.com" <bernard_aboba@hotmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Bria doesn't support VP8 on the desktop.   But perhaps can relook at linpho=
ne.

Also just noticed a URL error for this Bria result (2nd on the list) ...

<url correction>
   Bria using VP8 on iPhone to linphone on a fast desktop PC.=20
   http://dl.dropbox.com/u/17089001/video-samples/iphone-vp8.mov


----------------------------------------------
Looking at the videos from Section 4, using the same software on both sides=
 might help tease out implementation effects.  My understanding is that Bri=
a (at least on the iPad) supports both H.264 and VP8, and it would appear t=
hat Linphone also supports both.  Would it be possible to compare a Bria/Li=
nphone conversation with H.264 to one using VP8?

>From Section 4:

Captures of Mobile Tests;

   FaceTime using H.264 on an iPhone to linphone on a fast desktop PC.
   http://dl.dropbox.com/u/17089001/video-samples/facetime.mov

   Bria using VP8 on iPhone to linphone on a fast desktop PC. http://
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov

   Bria using VP8 on Galaxy Nexus to linphone on a fast desktop PC.
   http://dl.dropbox.com/u/17089001/video-samples/galaxy-nexus.mov

   Shown side-by-side, Bria using VP8 on iPhone (left) and FaceTime with
   H.264/AVC on iPhone (right), going to linphone and FaceTime. http://
   dl.dropbox.com/u/17089001/video-samples/iphone-vp8-vs-iphone-264.mov

   Captures of Desktop-only Tests;

   FaceTime H.264/AVC from a fast notebook computer to fast desktop. htt
   p://dl.dropbox.com/u/17089001/video-samples/facetime-264-desktop.mov

   -or- http://dl.dropbox.com/u/17089001/video-samples/
   facetime-264-desktop.webm

   Chrome with VP8 from same fast notebook computer to same fast
   desktop.
   http://dl.dropbox.com/u/17089001/video-samples/chrome-vp8-desktop.mov

   -or- http://dl.dropbox.com/u/17089001/video-samples/
   chrome-vp8-desktop.webm

   Two fast notebook computers - one running Chrome and the other
   running Fictive - both sending to the same desktop computer shown
   side by side. http://dl.dropbox.com/u/17089001/video-samples/
   chrome_and_factime_desktop.mov>

   -or- http://dl.dropbox.com/u/17089001/video-samples/
   chrome_and_factime_desktop.webm

From ron@debian.org  Tue Oct 23 22:42:46 2012
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Hi Bernard,

On Mon, Oct 22, 2012 at 07:33:29PM -0700, Bernard Aboba wrote:
> Ron said: 
> > If the cost of simply tracking all those users didn't bankrupt me all
> > by itself, then the licence fees certainly would try to finish the job.
> 
> [BA]  Pardon me if I'm obtuse, but I'd like to better understand the
> situation that is resulting in your being concerned about having to pay
> license fees and track users. 
> 
> Are you developing HTML5 applications?  f so, are you envisaging that as an
> HTML5 developer writing a WebRTC application for a browser that supports
> H.264 you would need to pay licensing fees? 
> 
> Are you working for a browser vendor?  If so, are you envisaging that you
> would need to pay licensing fees based on the number of users of your
> browser? 
> 
> Are you a developer of applications looking to interoperate with WebRTC?  If
> so, are you calling APIs that enable you to use the H.264 implementation in
> the OS platform, or accessing the hardware encode/decode functionality
> (available on most SoC platforms)?  If so, do you believe that you need to
> pay license fees? 
> 
> Or do you primarily develop on platforms without H.264 support in the OS, and
> on hardware with no H.264 encode/decode support?  

Yes, I see the distinction that you're aiming to point out there,
but I hadn't got the impression that anybody here actually was
confused by the difference between:

 "If a licenced implementation of H.264 is available, you can use it."

and 

 "RTCWeb can only be used on platforms with a licenced H.264 codec."


The former does not require H.264 to be MTI.  Like any other codec
it can always be negotiated when support for it exists at both ends.

The latter is a consequence of making H.264 MTI and would make this
standard useless to anyone who did not buy a licence to it themselves
or were not happy to restrict their userbase to platforms that had
purchased one for them in advance, whether they wanted it or not.

That doesn't seem like a good starting point for a standard that
likes to describe SIP as "legacy technology", and aims to become a
(if not The) ubiquitous communication technology for the future.


I don't see what is confusing about the idea that the baseline spec
should be RF, and open to as wide a field of users and developers
as possible -- with the ability to negotiate the optional use of
other (possibly royalty encumbered) technologies where some more
constrained demand or deployment or rationale for those also exists.

This would be consistent with the criteria applied for selection
of the MTI audio codec.



To return just briefly to the question of "risk" that was raised,
one other advantage to making VP8 the choice of MTI codec which
occurs to me is that since the known VP8 IPR comes with protective
conditions for its users in the form of the "poison pill" clauses
of its RF licence - then by making it MTI here, we'd also confer
that protection more broadly to RTCWeb users, since anyone suing
them for using VP8 would then also lose their own "licence" to
implement the RTCWeb standard too.

If RTCWeb is as successful at unifying communications as we all
hope it will be, that would certainly be a strong disincentive
to hostile action for anyone likely to actually hold patents on
any relevant communication technology.  That sounds like a much
stronger form of protection to me than "MPEG-LA can licence you
lots of, but not all of, the patents for H.264".  Doesn't it?


 Acutely,
 Ron



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Hi Cullen and all

If time is permitted, I suggest in JSEP part we can discuss the Rehydration=
 scenario.
I have an alternative approach in draft-li-rtcweb-ice-page-reload-02, which=
 suggests=20
to re-use the old connection to recovery the session. Also I see that in th=
e JSEP-02, it also=20
indicate an open issue for the Rehydration scenario, so I think it is worth=
 to discuss it during=20
the meeting. I can provide some slides for it if needed.=20

Commends for my approach are also welcomed.

Regards
shitao
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Cullen Jennings (fluffy)
> Sent: Thursday, October 04, 2012 11:44 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Agenda requests for Atlanta meeting
>=20
>=20
> With my individual contributor hat on .
>=20
> Justin and I would like to have some time for JSEP. I think there are a b=
unch of
> issues, particularly around the use of SDP and timing of various things t=
hat
> need discussion.
>=20
> Thanks, Cullen
>=20
>=20
>=20
> On Oct 2, 2012, at 9:08 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>=20
> > WG,
> >
> > We chairs like to have input into what you see should be on the Agenda
> > for our two meeting slots in Atlanta. Both document editors/authors and
> > participant are welcome to request or inform the WG or the chairs only
> > about issues that needs to be discussed in the meeting.
> >
> > Two items we chairs believe should be on the agenda are:
> >
> > - Video codec selection
> > - JSEP
> >
> >
> >
> > Cheers
> >
> > Magnus Westerlund
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > 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 stewe@stewe.org  Wed Oct 24 02:40:20 2012
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From: Stephan Wenger <stewe@stewe.org>
To: "video-codec@ietf.org" <video-codec@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Incoming liaison statement from MPEG
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Date: Wed, 24 Oct 2012 09:40:12 +0000
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Subject: [rtcweb] Incoming liaison statement from MPEG
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Hi all,
Please find attached an incoming liaison statement from MPEG.  It's short, =
and therefore attached.  For those not familiar with MPEG's terminology: "A=
VC" stands for "Advanced Video Codec" and is used in certain circles for wh=
at other people know as H.264, and sometimes, depending on context, only fo=
r the non-scalable and non-multiview profiles of H.264.  (H.264 is the ITU =
designation for the joint standard).  MPEG-4 part 29, or WebVC is, as descr=
ibed in the statement, what most people know as "constrained baseline H.264=
", i.e. H.264 baseline without the rarely used tools known as Arbitrary Sli=
ce Order, Flexible Macroblock Order, and Redundant Slices.
Also, in order to avoid possible confusion, let me emphasize that this liai=
son statement comes from the standardization body known as MPEG (ISO/IEC JT=
C1/SC29/WG11 ) and not from the licensing administrator known as MPEG-LA.
Regards,
Stephan




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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>Hi all,</div>
<div>Please find attached an incoming liaison statement from MPEG. &nbsp;It=
's short, and therefore attached. &nbsp;For those not familiar with MPEG's =
terminology: &quot;AVC&quot; stands for &quot;Advanced Video Codec&quot; an=
d is used in certain circles for what other people know as H.264,
 and sometimes, depending on context, only for the non-scalable and non-mul=
tiview profiles of H.264. &nbsp;(H.264 is the ITU designation for the joint=
 standard). &nbsp;MPEG-4 part 29, or WebVC is, as described in the statemen=
t, what most people know as &quot;constrained baseline
 H.264&quot;, i.e. H.264 baseline without the rarely used tools known as Ar=
bitrary Slice Order, Flexible Macroblock Order, and Redundant Slices.</div>
<div>Also, in order to avoid possible confusion, let me emphasize that this=
 liaison statement comes from the standardization body known as MPEG (ISO/I=
EC JTC1/SC29/WG11 ) and not from the licensing administrator known as MPEG-=
LA.</div>
<div>Regards,</div>
<div>Stephan</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</body>
</html>

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From: "David Benham (dbenham)" <dbenham@cisco.com>
To: Stephan Wenger <stewe@stewe.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Incoming liaison statement from MPEG
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To further connect dots, this part 29 effort in MPEG/IEC/ISO towards a non-=
royalty bearing solution was mentioned in our draft below as a potential up=
side to selecting H264/AVC constrained baseline profile today.  Unfortunate=
ly, the outcome will not be known soon enough, but to the extent future sur=
prises weigh on our decision ... this one is positive!



http://tools.ietf.org/html/draft-dbenham-webrtc-videomti-00
6.4.  Future IPR Status Consideration
   For additional informative purposes, the MPEG/IEC/ISO made a call for
   a royalty-free video coding solution for the Internet.  As of its
   98th meeting, two tracks forward are being pursued; one of which is
   based on the H.264/AVC constrained baseline profile [AVC06].  MPEG is
   presently collecting patent licensing declarations that would include
   patent holder's royalty expectations unbundled from other higher
   profile mechanisms.  Should this effort result in a royalty-free
   constrained baseline profile, this would benefit RTCweb in the future
   if H.264/AVC is selected as mandatory today.



From: Stephan Wenger [mailto:stewe@stewe.org]=20
Sent: Wednesday, October 24, 2012 2:40 AM
To: video-codec@ietf.org; rtcweb@ietf.org
Subject: [rtcweb] Incoming liaison statement from MPEG

Hi all,
Please find attached an incoming liaison statement from MPEG. =A0It's short=
, and therefore attached. =A0For those not familiar with MPEG's terminology=
: "AVC" stands for "Advanced Video Codec" and is used in certain circles fo=
r what other people know as H.264, and sometimes, depending on context, onl=
y for the non-scalable and non-multiview profiles of H.264. =A0(H.264 is th=
e ITU designation for the joint standard). =A0MPEG-4 part 29, or WebVC is, =
as described in the statement, what most people know as "constrained baseli=
ne H.264", i.e. H.264 baseline without the rarely used tools known as Arbit=
rary Slice Order, Flexible Macroblock Order, and Redundant Slices.
Also, in order to avoid possible confusion, let me emphasize that this liai=
son statement comes from the standardization body known as MPEG (ISO/IEC JT=
C1/SC29/WG11 ) and not from the licensing administrator known as MPEG-LA.
Regards,
Stephan




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To: "video-codec@ietf.org" <video-codec@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [video-codec] Incoming liaison statement from MPEG
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--_002_C5E08FE080ACFD4DAE31E4BDBF944EB11189E17Bxmbalnx02ciscoc_
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On Oct 24, 2012, at 3:40 AM, Stephan Wenger <stewe@stewe.org> wrote:

> Hi all,
> Please find attached an incoming liaison statement from MPEG.  It's short=
, and therefore attached.  For those not familiar with MPEG's terminology: =
"AVC" stands for "Advanced Video Codec" and is used in certain circles for =
what other people know as H.264, and sometimes, depending on context, only =
for the non-scalable and non-multiview profiles of H.264.  (H.264 is the IT=
U designation for the joint standard).  MPEG-4 part 29, or WebVC is, as des=
cribed in the statement, what most people know as "constrained baseline H.2=
64", i.e. H.264 baseline without the rarely used tools known as Arbitrary S=
lice Order, Flexible Macroblock Order, and Redundant Slices.
> Also, in order to avoid possible confusion, let me emphasize that this li=
aison statement comes from the standardization body known as MPEG (ISO/IEC =
JTC1/SC29/WG11 ) and not from the licensing administrator known as MPEG-LA.
> Regards,
> Stephan
>=20
>=20
>=20
> <29n130661.doc>_______________________________________________
> video-codec mailing list
> video-codec@ietf.org
> https://www.ietf.org/mailman/listinfo/video-codec

PDF version of the liaison is attached for folks that requested it.


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Subject: Re: [rtcweb] Incoming liaison statement from MPEG
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On 10/24/2012 10:58 PM, David Benham (dbenham) wrote:
> To further connect dots, this part 29 effort in MPEG/IEC/ISO towards a non-royalty bearing solution was mentioned in our draft below as a potential upside to selecting H264/AVC constrained baseline profile today.  Unfortunately, the outcome will not be known soon enough, but to the extent future surprises weigh on our decision ... this one is positive!
FWIW, there have been 11 IPR submissions into the ISO registry on MPEG-4 
part 29 to date; 1 of them specifies that the IPR is available on a 
royalty free basis, 10 do not. Unless I miscounted.

Apple deserves to be commended for making a royalty-free commitment, but 
it's a lonely position.

The registry of IPR submissions is available at 
http://www.iso.org/iso/standards_development/patents

>
>
>
> http://tools.ietf.org/html/draft-dbenham-webrtc-videomti-00
> 6.4.  Future IPR Status Consideration
>     For additional informative purposes, the MPEG/IEC/ISO made a call for
>     a royalty-free video coding solution for the Internet.  As of its
>     98th meeting, two tracks forward are being pursued; one of which is
>     based on the H.264/AVC constrained baseline profile [AVC06].  MPEG is
>     presently collecting patent licensing declarations that would include
>     patent holder's royalty expectations unbundled from other higher
>     profile mechanisms.  Should this effort result in a royalty-free
>     constrained baseline profile, this would benefit RTCweb in the future
>     if H.264/AVC is selected as mandatory today.
>
>
>
> From: Stephan Wenger [mailto:stewe@stewe.org]
> Sent: Wednesday, October 24, 2012 2:40 AM
> To: video-codec@ietf.org; rtcweb@ietf.org
> Subject: [rtcweb] Incoming liaison statement from MPEG
>
> Hi all,
> Please find attached an incoming liaison statement from MPEG.  It's short, and therefore attached.  For those not familiar with MPEG's terminology: "AVC" stands for "Advanced Video Codec" and is used in certain circles for what other people know as H.264, and sometimes, depending on context, only for the non-scalable and non-multiview profiles of H.264.  (H.264 is the ITU designation for the joint standard).  MPEG-4 part 29, or WebVC is, as described in the statement, what most people know as "constrained baseline H.264", i.e. H.264 baseline without the rarely used tools known as Arbitrary Slice Order, Flexible Macroblock Order, and Redundant Slices.
> Also, in order to avoid possible confusion, let me emphasize that this liaison statement comes from the standardization body known as MPEG (ISO/IEC JTC1/SC29/WG11 ) and not from the licensing administrator known as MPEG-LA.
> Regards,
> Stephan
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 10/24/2012 10:58 PM, David Benham
      (dbenham) wrote:<br>
    </div>
    <blockquote
cite="mid:0683D6CB32AC424D8AF52C0F660E5DC51509CFE0@xmb-aln-x10.cisco.com"
      type="cite">
      <pre wrap="">To further connect dots, this part 29 effort in MPEG/IEC/ISO towards a non-royalty bearing solution was mentioned in our draft below as a potential upside to selecting H264/AVC constrained baseline profile today.  Unfortunately, the outcome will not be known soon enough, but to the extent future surprises weigh on our decision ... this one is positive!</pre>
    </blockquote>
    FWIW, there have been 11 IPR submissions into the ISO registry on
    MPEG-4 part 29 to date; 1 of them specifies that the IPR is
    available on a royalty free basis, 10 do not. Unless I miscounted.<br>
    <br>
    Apple deserves to be commended for making a royalty-free commitment,
    but it's a lonely position.<br>
    <br>
    The registry of IPR submissions is available at
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a href="http://www.iso.org/iso/standards_development/patents">http://www.iso.org/iso/standards_development/patents</a><br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <blockquote
cite="mid:0683D6CB32AC424D8AF52C0F660E5DC51509CFE0@xmb-aln-x10.cisco.com"
      type="cite">
      <pre wrap="">



<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-dbenham-webrtc-videomti-00">http://tools.ietf.org/html/draft-dbenham-webrtc-videomti-00</a>
6.4.  Future IPR Status Consideration
   For additional informative purposes, the MPEG/IEC/ISO made a call for
   a royalty-free video coding solution for the Internet.  As of its
   98th meeting, two tracks forward are being pursued; one of which is
   based on the H.264/AVC constrained baseline profile [AVC06].  MPEG is
   presently collecting patent licensing declarations that would include
   patent holder's royalty expectations unbundled from other higher
   profile mechanisms.  Should this effort result in a royalty-free
   constrained baseline profile, this would benefit RTCweb in the future
   if H.264/AVC is selected as mandatory today.



From: Stephan Wenger [<a class="moz-txt-link-freetext" href="mailto:stewe@stewe.org">mailto:stewe@stewe.org</a>] 
Sent: Wednesday, October 24, 2012 2:40 AM
To: <a class="moz-txt-link-abbreviated" href="mailto:video-codec@ietf.org">video-codec@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
Subject: [rtcweb] Incoming liaison statement from MPEG

Hi all,
Please find attached an incoming liaison statement from MPEG. &nbsp;It's short, and therefore attached. &nbsp;For those not familiar with MPEG's terminology: "AVC" stands for "Advanced Video Codec" and is used in certain circles for what other people know as H.264, and sometimes, depending on context, only for the non-scalable and non-multiview profiles of H.264. &nbsp;(H.264 is the ITU designation for the joint standard). &nbsp;MPEG-4 part 29, or WebVC is, as described in the statement, what most people know as "constrained baseline H.264", i.e. H.264 baseline without the rarely used tools known as Arbitrary Slice Order, Flexible Macroblock Order, and Redundant Slices.
Also, in order to avoid possible confusion, let me emphasize that this liaison statement comes from the standardization body known as MPEG (ISO/IEC JTC1/SC29/WG11 ) and not from the licensing administrator known as MPEG-LA.
Regards,
Stephan



_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
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Hi,

Some comments on JSEP-02.


Q1: Cloning

I saw that you have removed the text about cloning from version -02.

That is a rather fundamental change, so I think the change log should say a=
 little more than "Clarified section on forking.".


Q2: Gone to W3C

In the change log you say "Removed stuff that has moved to W3C specificaito=
n". I think it would be good with a least some key words on exactly what ha=
s been moved.

I know I can do a diff with the previous version, but as one of the issues =
have been what belongs to IETF and what belongs to W3C I don't think I shou=
ld have to do that.


Q3: Subsequent offer

The text says:

        "In [RFC3264], the constraints at the signaling level is that only =
one
        offer can be outstanding for a given session but from the media sta=
ck
        level, a new offer can be generated at any point.  For example, whe=
n
        using SIP for signaling, if one offer is sent, then cancelled using=
 a
        SIP CANCEL, another offer can be generated even though no answer wa=
s
        received for the first offer.  To support this, the JSEP media laye=
r
        can provide an offer whenever the Javascript application needs one
        for the signaling.  The answerer can send back zero or more
        provisional answers, and finally end the offer-answer exchange by
        sending a final answer."

This is one of the issues previously discussed, but IF we want support gene=
rating a new offer while the previous one is still pending, then you need t=
o state that only answers associated with the latest offer are provided bac=
k to the browser.


Q4: Remove offer

This is related to the PRANSWER issue.

Unless I've missed it, the draft does not address the issue raised on the l=
ist, where an offer is received from the remote peer while in PRANSWER stat=
e.

(If you solved it on the list, I applogize for not having read it. I've had=
 a flu the whole week, so I am a little behind with my e-mails)


Q5: Trickle ICE

Don't get me wrong - I support the work on trickle ICE :) '

BUT, the trickle ICE draft is still "fresh from the owen", so I suggest to =
e.g. add a note which says that the mechanism has still not been adopted, a=
nd that people need to be aware that things may change. It may be clear to =
people in IETF, but maybe not to others.



Q6: ROAP

Section A.1.1 shows a call flow example with ROAP.

The latest version of draft-jennings-roap expired in may. Are we going to c=
ontinue the work on ROAP?



Regards,

Christer


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<font face=3D"Arial" size=3D"3"><span style=3D"font-size:12pt;">
<div>&nbsp;</div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Hi,</span></font></di=
v>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Some comments on JSEP=
-02.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q1: Cloning</b></s=
pan></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">I saw that you have r=
emoved the text about cloning from version -02.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">That is a rather fund=
amental change, so I think the change log should say a little more than &qu=
ot;Clarified section on forking.&quot;.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q2: Gone to W3C</b=
></span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">In the change log you=
 say &quot;Removed stuff that has moved to W3C specificaiton&quot;. I think=
 it would be good with a least some key words on exactly what has been move=
d.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">I know I can do a dif=
f with the previous version, but as one of the issues have been what belong=
s to IETF and what belongs to W3C I don't think I should have to do that.</=
span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q3: Subsequent off=
er</b></span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">The text says:</span>=
</font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; </span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; &quot;In [RFC3264], the constraints at the signaling =
level is that only one</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; offer can be outstanding for a given session bu=
t from the media stack</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; level, a new offer can be generated at any poin=
t.&nbsp; For example, when</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; using SIP for signaling, if one offer is sent, =
then cancelled using a</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; SIP CANCEL, another offer can be generated even=
 though no answer was</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; received for the first offer.&nbsp; To support =
this, the JSEP media layer</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; can provide an offer whenever the Javascript ap=
plication needs one</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; for the signaling.&nbsp; The answerer can send =
back zero or more</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; provisional answers, and finally end the offer-=
answer exchange by</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; sending a final answer.&quot;</span></font></di=
v>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">This is one of the is=
sues previously discussed, but IF we want support generating a new offer wh=
ile the previous one is still pending, then you need to state that only ans=
wers associated with the latest offer
are provided back to the browser.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q4: Remove offer</=
b></span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">This is related to th=
e PRANSWER issue.</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Unless I've missed it=
, the draft does not address the issue raised on the list, where an offer i=
s received from the remote peer while in PRANSWER state.</span></font></div=
>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">(If you solved it on =
the list, I applogize for not having read it. I've had a flu the whole week=
, so I am a little behind with my e-mails)</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q5: Trickle ICE</b=
></span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Don't get me wrong - =
I support the work on trickle ICE :) '</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">BUT, the trickle ICE =
draft is still &quot;fresh from the owen&quot;, so I suggest to e.g. add a =
note which says that the mechanism has still not been adopted, and that peo=
ple need to be aware that things may change. It
may be clear to people in IETF, but maybe not to others.</span></font></div=
>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;"><b>Q6: ROAP</b></span=
></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Section A.1.1 shows a=
 call flow example with ROAP. </span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">The latest version of=
 draft-jennings-roap expired in may. Are we going to continue the work on R=
OAP?</span></font></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Regards,</span></font=
></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">Christer</span></font=
></div>
<div><font size=3D"2"><span style=3D"font-size:10pt;">&nbsp;</span></font><=
/div>
</span></font>
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Ok, so now that we've had one round of the usual debate that won't 
terminate, *how* much time on the agenda do we really want to spend 
debating the potential merits or flaws of codecs that some of us either 
will or will not use no matter what happens in the meeting?

I really think there's more important things to discuss and specify.

Matthew Kaufman

From ted.ietf@gmail.com  Thu Oct 25 16:14:20 2012
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On Thu, Oct 25, 2012 at 3:26 PM, Matthew Kaufman <matthew@matthew.at> wrote:
> Ok, so now that we've had one round of the usual debate that won't
> terminate, *how* much time on the agenda do we really want

The chairs are currently discussing how to compress the presentation
time, and we hope to have a query to the authors very soon.  That,
presuming it works out, will regain some time.  If there are short mic
lines after the presentations, we will also gain time.

We hope to have an update of the draft agenda that's currently update
by early next week.  Additional topics that you wish to see covered in
any time we make available would be welcome.

Ted

From ron@debian.org  Thu Oct 25 17:15:48 2012
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On Thu, Oct 25, 2012 at 03:26:31PM -0700, Matthew Kaufman wrote:
> Ok, so now that we've had one round of the usual debate that won't
> terminate, *how* much time on the agenda do we really want to spend
> debating the potential merits or flaws of codecs that some of us
> either will or will not use no matter what happens in the meeting?

If you feel that the discussion has exhausted all the rationale
that you have to offer, perhaps you'd like to summarise what you
see as the potential merits and flaws of each candidate for us?

"I will not use it no matter what", isn't much of a technical
argument.

"I cannot use it because people are still fighting in the courts
 over who actually has a licence and how much one should cost",
is more than a technical problem, it's a real and present, and
apparently quite enduring danger haunting H.264, and not just for
the people otherwise ineligible to get a licence for it at all.

Am I forgetting one, or is it really now the most litigated codec
in the history of all codecs?


But perhaps you have some other list of pros and cons that
somehow changes that balance?  I'd certainly like to see a
concise summary of what I'm supposedly missing that means
VP8 isn't technically the no-brainer choice to make here, 
in pretty much every respect that has been discussed.


> I really think there's more important things to discuss and specify.

I agree.  But until we get past the question of MTI codecs that
will ensure interoperability, the risk remains that those things
will just be Rearranging The Deck Chairs.  Again.

So let's get some consensus on a summary of pros and cons for
each, that will give us some reasoned basis to back whatever
decision we do arrive at on this, and let us really move on.

 Double Hulled and Hopeful This Time,
 Ron



From matthew@matthew.at  Thu Oct 25 17:25:48 2012
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On 10/25/2012 5:15 PM, Ron wrote:
> On Thu, Oct 25, 2012 at 03:26:31PM -0700, Matthew Kaufman wrote:
>> Ok, so now that we've had one round of the usual debate that won't
>> terminate, *how* much time on the agenda do we really want to spend
>> debating the potential merits or flaws of codecs that some of us
>> either will or will not use no matter what happens in the meeting?
> If you feel that the discussion has exhausted all the rationale
> that you have to offer, perhaps you'd like to summarise what you
> see as the potential merits and flaws of each candidate for us?
>

I think I made this clear in an earlier post. VP8 is not the product of 
a recognized standards body. This has negative technical implications 
and negative legal implications. I am not a lawyer, and this is an IETF 
list, so that's two reasons for me to not go into more detail on the latter.

My employer happens to be a major browser vendor, has published blog 
postings that I've previously referenced on this topic, and nothing 
substantive has changed since those were posted as far as I can tell.

As was pointed out by someone else, and quoted by me previously, the 
decision to include or not include a particular codec in a shipping 
product is not going to be made by any of the technical people on this 
list, and so discussing it here or at the meeting is pointless. And 
presenting *technical* arguments as to the superiority of one or another 
is likewise pointless.

Matthew Kaufman


From stewe@stewe.org  Thu Oct 25 19:07:37 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Ron <ron@debian.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi Ron,
A few comments inline.

On 10.26.2012 08:15 , "Ron" <ron@debian.org> wrote:

[...]
>
>"I cannot use it because people are still fighting in the courts
> over who actually has a licence and how much one should cost",
>is more than a technical problem, it's a real and present, and
>apparently quite enduring danger haunting H.264, and not just for
>the people otherwise ineligible to get a licence for it at all.
>
>Am I forgetting one, or is it really now the most litigated codec
>in the history of all codecs?

Oh yes, you are forgetting MPEG-2.  Out of the top of my head, I can
recollect four high profile lawsuits regarding MPEG-2 video essential
claims.  I believe there have been several more, and enforcement is going
on even in the most recent past (the Nero antitrust dispute being the most
prominent one in the last three years).

Many of those MPEG-2 related lawsuits were the result of enforcement
action in an emerging licensing ecosystem that is based on royalties.  I'm
not aware of any video codec patent disputes, nor of any significant
royalty-bearing licensing deals in the video codec space, before the
advent of MPEG-2, formation of the related MPEG-LA pool, and the resulting
wave of litigation.

While there have been a few more, the single most prominent H.264 related
lawsuit today is the action brought by Motorola/google against a number of
companies that could be viewed as threatening the Android ecosystem.
(Most of the older stuff is straightforward enforcement in a settled
royalty-bearing ecosystem.)  The Motorola lawsuits are IMO strategic
disputes, and the use of video codec related patents is just a means to
achieve a higher goal: keep the Android system competitive.

Also, with respect to the Motorola lawsuits, it is my understanding that
the patent claims being litigated are related to interlace technologies
and, therefore, are not relevant to the practice of constrained baseline.
I'm not going to discuss this further; those interested may want to
consult the web or, much better, your trusted patent lawyer.

[...]

Stephan

>But perhaps you have some other list of pros and cons that
>somehow changes that balance?  I'd certainly like to see a
>concise summary of what I'm supposedly missing that means
>VP8 isn't technically the no-brainer choice to make here,
>in pretty much every respect that has been discussed.
>
>
>> I really think there's more important things to discuss and specify.
>
>I agree.  But until we get past the question of MTI codecs that
>will ensure interoperability, the risk remains that those things
>will just be Rearranging The Deck Chairs.  Again.
>
>So let's get some consensus on a summary of pros and cons for
>each, that will give us some reasoned basis to back whatever
>decision we do arrive at on this, and let us really move on.
>
> Double Hulled and Hopeful This Time,
> Ron
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From ron@debian.org  Thu Oct 25 19:53:04 2012
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On Thu, Oct 25, 2012 at 05:25:44PM -0700, Matthew Kaufman wrote:
> On 10/25/2012 5:15 PM, Ron wrote:
> >On Thu, Oct 25, 2012 at 03:26:31PM -0700, Matthew Kaufman wrote:
> >>Ok, so now that we've had one round of the usual debate that won't
> >>terminate, *how* much time on the agenda do we really want to spend
> >>debating the potential merits or flaws of codecs that some of us
> >>either will or will not use no matter what happens in the meeting?
> >If you feel that the discussion has exhausted all the rationale
> >that you have to offer, perhaps you'd like to summarise what you
> >see as the potential merits and flaws of each candidate for us?
> >
> 
> I think I made this clear in an earlier post. VP8 is not the product
> of a recognized standards body. This has negative technical
> implications and negative legal implications. I am not a lawyer, and
> this is an IETF list, so that's two reasons for me to not go into
> more detail on the latter.
> 
> As was pointed out by someone else, and quoted by me previously, the
> decision to include or not include a particular codec in a shipping
> product is not going to be made by any of the technical people on
> this list, and so discussing it here or at the meeting is pointless.
> And presenting *technical* arguments as to the superiority of one or
> another is likewise pointless.

That's a pretty good summary of the problems I'm seeing with the
arguments against VP8 indeed.  And exactly why I asked to see a
more substantial summary of them.

In the one breath you are saying "not the product of an SDO == Bad"
is your only direct argument against it.

And then you immediately follow that with "we should ignore anything
this SDO has to say about it anyway".  And then wonder why I'm more
than a little bewildered about what exactly your argument is here?

The decision of which codec should be MTI for this proposed standard
not being the product of this recognised standards body, I completely
agree would have negative technical and legal implications for it.
Serious ones at that.


It is far less clear in practice how much harm this does to a codec
that was carefully designed by experts in that field though.  It is
clear that working within the IETF process did wonderful things for
Opus -- but it is much less clear that the same people working
together in the same way would have produced total crap if the
efforts to oppose formation of the CODEC WG had been "successful"
at that and they had done their work outside of an SDO instead.

Where is the offensive patent pool against Theora that we've been
hearing about for so many years?  Or the single lawsuit against a
user of it?  What about Vorbis?  Any problems it had were neither
technical nor legal.  Can nobody at all shed any light on the
status of the mythical pool that wanted to attack VP8 which seems
to have vanished in a whimper over a year ago?  Did anyone sue
Skype for SILK?  How does the success of those compare to OOXML?


I don't really see how your argument there is coherent with either
itself or the observable reality.  But you're insisting on it very
strongly so I'm hoping you can clue us in on the missing link that
makes it actually make some sense.



> My employer happens to be a major browser vendor, has published blog
> postings that I've previously referenced on this topic, and nothing
> substantive has changed since those were posted as far as I can
> tell.

So that's the basis on which you think we should decide this?

There are two other major browser vendors contributing here,
and as I understand it, they both prefer VP8.  Since they have
the lion's share of users, why shouldn't their preference stand
if this logic is applied?

You say nothing substantive has changed since then.
The slope on this graph looks like pretty substantive change to me:
http://upload.wikimedia.org/wikipedia/commons/8/86/Usage_share_of_web_browsers_%28Source_StatCounter%29.svg

The numbers here look pretty substantive too:
http://www.w3schools.com/browsers/browsers_stats.asp


But personally I'd still much rather see us build consensus around
a clear list of agreed pros and cons, so that when people look back
and wonder "how did the RTCWeb WG make that decision" we will have
something a bit more substantive to show for it than "via a farce
where industry heavyweights had their preference rubber-stamped
without discussion or justification that held up to even the most
casual scrutiny."

I'm pretty sure this still is a technical standards group, and that
technical arguments to back up its decisions mostly aren't pointless.

  Unabashedly,
  Ron



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Thanks for the review and suggestions, inline =85

On Oct 25, 2012, at 7:50 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> =20
> Hi,
> =20
> Some comments on JSEP-02.
> =20
> =20
> Q1: Cloning
> =20
> I saw that you have removed the text about cloning from version -02.
> =20
> That is a rather fundamental change, so I think the change log should say=
 a little more than "Clarified section on forking.".

agree

> =20
> =20
> Q2: Gone to W3C
> =20
> In the change log you say "Removed stuff that has moved to W3C specificai=
ton". I think it would be good with a least some key words on exactly what =
has been moved.
> =20
> I know I can do a diff with the previous version, but as one of the issue=
s have been what belongs to IETF and what belongs to W3C I don't think I sh=
ould have to do that.

fair enough - to be honest, we ran out of time on some of this. You will no=
tice the draft went in seconds before the deadline and the references are a=
lso woefully inadequate.=20

> =20
> =20
> Q3: Subsequent offer
> =20
> The text says:
>       =20
>         "In [RFC3264], the constraints at the signaling level is that onl=
y one
>          offer can be outstanding for a given session but from the media =
stack
>          level, a new offer can be generated at any point.  For example, =
when
>          using SIP for signaling, if one offer is sent, then cancelled us=
ing a
>          SIP CANCEL, another offer can be generated even though no answer=
 was
>          received for the first offer.  To support this, the JSEP media l=
ayer
>          can provide an offer whenever the Javascript application needs o=
ne
>          for the signaling.  The answerer can send back zero or more
>          provisional answers, and finally end the offer-answer exchange b=
y
>          sending a final answer."
> =20
> This is one of the issues previously discussed, but IF we want support ge=
nerating a new offer while the previous one is still pending, then you need=
 to state that only answers associated with the latest offer are provided b=
ack to the browser.

I agree with what you are saying but I think it gets a little more complica=
ted once we include the roll back. I hope to get the roll back in the next =
version but yes, agree with this along with some exceptions to it when doin=
g roll back.=20

> =20
> =20
> Q4: Remove offer
> =20
> This is related to the PRANSWER issue.
> =20
> Unless I've missed it, the draft does not address the issue raised on the=
 list, where an offer is received from the remote peer while in PRANSWER st=
ate.
> =20
> (If you solved it on the list, I applogize for not having read it. I've h=
ad a flu the whole week, so I am a little behind with my e-mails)]

that's a signaling problem. I doubt the draft should address it any more th=
an there are a few ways of dealing with it and Javascript can do whatever i=
t wants to choose which one it wants.=20

> =20
> =20
> Q5: Trickle ICE
> =20
> Don't get me wrong - I support the work on trickle ICE :) '
> =20
> BUT, the trickle ICE draft is still "fresh from the owen", so I suggest t=
o e.g. add a note which says that the mechanism has still not been adopted,=
 and that people need to be aware that things may change. It may be clear t=
o people in IETF, but maybe not to others.
> =20

agree - (actually, I think the ICE draft is still getting ready to go in th=
e oven :-)

> =20
> =20
> Q6: ROAP
> =20
> Section A.1.1 shows a call flow example with ROAP.
> =20
> The latest version of draft-jennings-roap expired in may. Are we going to=
 continue the work on ROAP?
> =20

I really have no idea. At first I thought I should take this out. Then I wa=
nted to show something that mapped to another signaling protocol. This shou=
ld probably go, but something should be put there to help illustrate this t=
ype of flow.=20

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


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
Thread-Topic: [rtcweb] Relaxing SDP O/A (was RE: Agenda requests for Atlanta meeting)
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On Oct 19, 2012, at 3:23 AM, "Hutton, Andrew" <andrew.hutton@siemens-enterp=
rise.com> wrote:

>=20
> RFC3624 requires that the session reverts to the state prior to the offer=
 if the offer/answer exchange fails.=20
>=20
> If an applications attempts to modify a session but the updated offer is =
rejected by the far end there should be no interruption to the existing ses=
sion. Like Roman suggested I am also thinking it would be nice if the API p=
rovided some way of cancelling an offer and reverting to the original state=
 but if this is putting too much state back in the browser we would need a =
clearly documented mechanism.

We talked about this before and I think just about everyone agrees that we =
need rollback in JSEP.  And it's not just needed for gatewaying with SIP, a=
ny signaling protocol that wanted to allow simultaneous changes is likely n=
eed it.  It's not yet fully defined in the current draft because we need to=
 get stable on what we do the initial "O/A" pair before we step into the mo=
difications of it. In SIP terms this would be let make sure we understand i=
nitial INVITE before taking on UPDATE.=20

I agree this is 1) needed and 2) needs to be well defined in JSEP 3) it's n=
ot well defined in current draft. The idea/hope is that the current API wil=
l support rollback with no change to the API, we just have to write down th=
e semantics and check all the details work. =

From christer.holmberg@ericsson.com  Mon Oct 29 05:44:07 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] JSEP-02 Comments
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Hi,
=20
>> Q1: Cloning
>> =20
>> I saw that you have removed the text about cloning from version -02.
>> =20
>> That is a rather fundamental change, so I think the change log should sa=
y a little more than "Clarified section on forking.".
>
> agree

And, do we have a WG concensus to remove cloning?

>> Q4: Remove offer
>> =20
>> This is related to the PRANSWER issue.
>> =20
>> Unless I've missed it, the draft does not address the issue raised on th=
e list, where an offer is received from the remote peer while in PRANSWER s=
tate.
>> =20
>> (If you solved it on the list, I applogize for not having read it. I've =
had a flu the whole week, so I am a little behind with my e-mails)]
>
> that's a signaling problem. I doubt the draft should address it any more =
than there are a few ways of dealing with it and Javascript can do whatever=
 it wants to choose which one it wants.=20

It's a use-case I'd like the API to support.

But, even without PRANSWER it would be tricky, as you have removed cloning.

Regards,

Christer

From magnus.westerlund@ericsson.com  Mon Oct 29 08:12:47 2012
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WG,

I spent a little time reviewing what dependencies we have in our current
set of WG drafts on I-Ds in other WGs. The below is the result of that
analysis. I can't promise that all listed are actually still relevant
normative references, they can equally be left overs. I for example
found two reference issues in my own WG document on the RTP usage. But I
think it gives us good indication that the WG memebers needs to be
active and contribute to work in at least MMUSIC, AVTCORE and TSVWG to
enable our WG to finish its work.

There are more WGs where things goes on that are relevant, like the OPUS
payload format in the PAYLOAD WG. draft-spittka-payload-rtp-opus-01
which wasn't referenced by either the audio-codec doc or the RTP usage one.

Please review and comment on these documents so they make progress.

MMUSIC
draft-ietf-mmusic-sctp-sdp-02
draft-ietf-mmusic-traffic-class-for-sdp-02
draft-ietf-mmusic-sdp-bundle-negotiation-01
draft-alvestrand-mmusic-msid-01
draft-rescorla-mmusic-ice-trickle-00


AVTCORE
draft-ietf-avtcore-rtp-circuit-breakers-00
draft-ietf-avtcore-srtp-encrypted-header-ext-02
draft-ietf-avtext-multiple-clock-rates-06
draft-rescorla-avtcore-6222bis-00
draft-terriberry-avp-codecs-00
draft-ietf-avtcore-multi-media-rtp-session-01

Potentially some of these:
draft-lennox-avtcore-rtp-multi-stream-01
draft-westerlund-avtcore-transport-multiplexing-04
draft-westerlund-avtcore-rtp-topologies-update-01
draft-westerlund-avtcore-multiplex-architecture-02
draft-ietf-avt-srtp-ekt-03

TSVWG
draft-tuexen-tsvwg-sctp-dtls-encaps-01
draft-ietf-tsvwg-sctp-udp-encaps-06

Behave
draft-muthu-behave-consent-freshness-01

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From petithug@acm.org  Mon Oct 29 09:09:34 2012
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

Please note that http://www.w3.org/TR/webrtc/ have normative references to:

http://tools.ietf.org/html/draft-petithuguenin-behave-turn-uris
http://tools.ietf.org/html/draft-nandakumar-rtcweb-stun-uri

On 10/29/2012 08:12 AM, Magnus Westerlund wrote:
> WG,
> 
> I spent a little time reviewing what dependencies we have in our current 
> set of WG drafts on I-Ds in other WGs. The below is the result of that 
> analysis. I can't promise that all listed are actually still relevant 
> normative references, they can equally be left overs. I for example found
> two reference issues in my own WG document on the RTP usage. But I think it
> gives us good indication that the WG memebers needs to be active and
> contribute to work in at least MMUSIC, AVTCORE and TSVWG to enable our WG
> to finish its work.
> 
> There are more WGs where things goes on that are relevant, like the OPUS 
> payload format in the PAYLOAD WG. draft-spittka-payload-rtp-opus-01 which
> wasn't referenced by either the audio-codec doc or the RTP usage one.
> 
> Please review and comment on these documents so they make progress.
> 
> MMUSIC draft-ietf-mmusic-sctp-sdp-02 
> draft-ietf-mmusic-traffic-class-for-sdp-02 
> draft-ietf-mmusic-sdp-bundle-negotiation-01 
> draft-alvestrand-mmusic-msid-01 draft-rescorla-mmusic-ice-trickle-00
> 
> 
> AVTCORE draft-ietf-avtcore-rtp-circuit-breakers-00 
> draft-ietf-avtcore-srtp-encrypted-header-ext-02 
> draft-ietf-avtext-multiple-clock-rates-06 
> draft-rescorla-avtcore-6222bis-00 draft-terriberry-avp-codecs-00 
> draft-ietf-avtcore-multi-media-rtp-session-01
> 
> Potentially some of these: draft-lennox-avtcore-rtp-multi-stream-01 
> draft-westerlund-avtcore-transport-multiplexing-04 
> draft-westerlund-avtcore-rtp-topologies-update-01 
> draft-westerlund-avtcore-multiplex-architecture-02 
> draft-ietf-avt-srtp-ekt-03
> 
> TSVWG draft-tuexen-tsvwg-sctp-dtls-encaps-01 
> draft-ietf-tsvwg-sctp-udp-encaps-06
> 
> Behave draft-muthu-behave-consent-freshness-01
> 
> Cheers
> 
> Magnus Westerlund
> 
> ---------------------------------------------------------------------- 
> Multimedia Technologies, Ericsson Research EAB/TVM 
> ---------------------------------------------------------------------- 
> Ericsson AB                | Phone  +46 10 7148287 Färögatan 6
> | Mobile +46 73 0949079 SE-164 80 Stockholm, Sweden| mailto:
> magnus.westerlund@ericsson.com 
> ----------------------------------------------------------------------
> 
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 


- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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From: "Eggert, Lars" <lars@netapp.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] RTCWEB WG document's dependencies outside of the WG
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On Oct 29, 2012, at 16:12, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
> I spent a little time reviewing what dependencies we have in our =
current
> set of WG drafts on I-Ds in other WGs. The below is the result of that
> analysis.

Great overview!

You will probably in the near future also grow some dependencies to =
RMCAT documents, once we have started adopting some.

Lars=

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From: Stephan Wenger <stewe@stewe.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWEB WG document's dependencies outside of the WG
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If you are an VP8 fan, you may also want to add the VP8 payload format:
http://datatracker.ietf.org/doc/draft-ietf-payload-vp8/
This doc is currently in last call in the payload WG.
Stephan

On 10.29.2012 08:12 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>WG,
>
>I spent a little time reviewing what dependencies we have in our current
>set of WG drafts on I-Ds in other WGs. The below is the result of that
>analysis. I can't promise that all listed are actually still relevant
>normative references, they can equally be left overs. I for example
>found two reference issues in my own WG document on the RTP usage. But I
>think it gives us good indication that the WG memebers needs to be
>active and contribute to work in at least MMUSIC, AVTCORE and TSVWG to
>enable our WG to finish its work.
>
>There are more WGs where things goes on that are relevant, like the OPUS
>payload format in the PAYLOAD WG. draft-spittka-payload-rtp-opus-01
>which wasn't referenced by either the audio-codec doc or the RTP usage
>one.
>
>Please review and comment on these documents so they make progress.
>
>MMUSIC
>draft-ietf-mmusic-sctp-sdp-02
>draft-ietf-mmusic-traffic-class-for-sdp-02
>draft-ietf-mmusic-sdp-bundle-negotiation-01
>draft-alvestrand-mmusic-msid-01
>draft-rescorla-mmusic-ice-trickle-00
>
>
>AVTCORE
>draft-ietf-avtcore-rtp-circuit-breakers-00
>draft-ietf-avtcore-srtp-encrypted-header-ext-02
>draft-ietf-avtext-multiple-clock-rates-06
>draft-rescorla-avtcore-6222bis-00
>draft-terriberry-avp-codecs-00
>draft-ietf-avtcore-multi-media-rtp-session-01
>
>Potentially some of these:
>draft-lennox-avtcore-rtp-multi-stream-01
>draft-westerlund-avtcore-transport-multiplexing-04
>draft-westerlund-avtcore-rtp-topologies-update-01
>draft-westerlund-avtcore-multiplex-architecture-02
>draft-ietf-avt-srtp-ekt-03
>
>TSVWG
>draft-tuexen-tsvwg-sctp-dtls-encaps-01
>draft-ietf-tsvwg-sctp-udp-encaps-06
>
>Behave
>draft-muthu-behave-consent-freshness-01
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From ted.ietf@gmail.com  Mon Oct 29 11:08:19 2012
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Subject: [rtcweb] Form of the video codec question
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We got feedback after the audio codec discussion that folks would have
benefited from seeing the question we intended to ask before the
meeting.  In order to clarify that for this discussion, the chairs are
putting forward this draft of the question which will be asked in the
meeting (and then confirmed on the list).

"1) If you support H.264 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

2) If you support VP8 as the mandatory to implement codec or are
willing to live with it as the MTI, please raise your hand now.

You may raise your hand more than once and we encourage you to do so
if you can live with either, even if you have a preference for one
over the other."

Because all 3 chairs and our responsible AD could be perceived to have
conflicts of interest here, we have asked Robert Sparks to judge the
consensus of the room, and he has kindly agreed.  If you have major
heartburn about the form of the question, please let us know.

regards,

Ted Hardie, Cullen Jennings, Magnus Westerlund

From stewe@stewe.org  Mon Oct 29 11:51:53 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Form of the video codec question
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Hi Ted, chairs,

I think these are the wrong questions, as the result of such a poll is not
going to solve anything.  It's trivial to predict that there will be a
significant show of hands in favor of each question.  What does that tell
you?  Asking the questions as presented is going to be a waste of time.

The more informative questions to as would be:

1) Do you object to H.264 as MTI, regardless of its technical and IPR
merits (if any); and
2) Do you object to VP8 as MTI, regardless of its technical and IPR merits
(if any)
(both in more politically correct formulation, if you want.)

If you chairs know that there are objections to both codecs by a
significant number of people that cannot be swayed by technical or IPR
arguments, a presentation of such technical and IPR merits can be omitted,
and the WG can focus on stuff where progress is likely.

Regards,
Stephan


On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:

>We got feedback after the audio codec discussion that folks would have
>benefited from seeing the question we intended to ask before the
>meeting.  In order to clarify that for this discussion, the chairs are
>putting forward this draft of the question which will be asked in the
>meeting (and then confirmed on the list).
>
>"1) If you support H.264 as the mandatory to implement codec or are
>willing to live with it as the MTI, please raise your hand now.
>
>2) If you support VP8 as the mandatory to implement codec or are
>willing to live with it as the MTI, please raise your hand now.
>
>You may raise your hand more than once and we encourage you to do so
>if you can live with either, even if you have a preference for one
>over the other."
>
>Because all 3 chairs and our responsible AD could be perceived to have
>conflicts of interest here, we have asked Robert Sparks to judge the
>consensus of the room, and he has kindly agreed.  If you have major
>heartburn about the form of the question, please let us know.
>
>regards,
>
>Ted Hardie, Cullen Jennings, Magnus Westerlund
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From ted.ietf@gmail.com  Mon Oct 29 13:17:28 2012
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Hi Stephan,


On Mon, Oct 29, 2012 at 11:51 AM, Stephan Wenger <stewe@stewe.org> wrote:
> Hi Ted, chairs,
>
> I think these are the wrong questions, as the result of such a poll is not
> going to solve anything.  It's trivial to predict that there will be a
> significant show of hands in favor of each question.  What does that tell
> you?

It tells us whether there is rough consensus or not.

Asking the questions as presented is going to be a waste of time.
>
> The more informative questions to as would be:
>
> 1) Do you object to H.264 as MTI, regardless of its technical and IPR
> merits (if any); and
> 2) Do you object to VP8 as MTI, regardless of its technical and IPR merits
> (if any)
> (both in more politically correct formulation, if you want.)
>

So, this seems to be the inverse ("Can you not live with X as MTI").
Why is asking this as the inverse going to give more information?

We understand, of course, that the group has folks with strong
opinions.  They are vocal enough that no one is confused on this
point.  But we hope to understand, after a set of technical
presentations, whether there is a sufficiently large group with a
willingness to live with one or the other to create a rough consensus.

regards,

Ted Hardie

> If you chairs know that there are objections to both codecs by a
> significant number of people that cannot be swayed by technical or IPR
> arguments, a presentation of such technical and IPR merits can be omitted,
> and the WG can focus on stuff where progress is likely.
>
> Regards,
> Stephan
>
>
> On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>
>>We got feedback after the audio codec discussion that folks would have
>>benefited from seeing the question we intended to ask before the
>>meeting.  In order to clarify that for this discussion, the chairs are
>>putting forward this draft of the question which will be asked in the
>>meeting (and then confirmed on the list).
>>
>>"1) If you support H.264 as the mandatory to implement codec or are
>>willing to live with it as the MTI, please raise your hand now.
>>
>>2) If you support VP8 as the mandatory to implement codec or are
>>willing to live with it as the MTI, please raise your hand now.
>>
>>You may raise your hand more than once and we encourage you to do so
>>if you can live with either, even if you have a preference for one
>>over the other."
>>
>>Because all 3 chairs and our responsible AD could be perceived to have
>>conflicts of interest here, we have asked Robert Sparks to judge the
>>consensus of the room, and he has kindly agreed.  If you have major
>>heartburn about the form of the question, please let us know.
>>
>>regards,
>>
>>Ted Hardie, Cullen Jennings, Magnus Westerlund
>>_______________________________________________
>>rtcweb mailing list
>>rtcweb@ietf.org
>>https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>

From john@jlc.net  Mon Oct 29 14:04:44 2012
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Ted Hardie <ted.ietf@gmail.com> wrote:
> 
> In order to clarify that for this discussion, the chairs are
> putting forward this draft of the question which will be asked in the
> meeting (and then confirmed on the list).
> 
> "1) If you support H.264 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
> 
> 2) If you support VP8 as the mandatory to implement codec or are
> willing to live with it as the MTI, please raise your hand now.
> 
> You may raise your hand more than once and we encourage you to do so
> if you can live with either, even if you have a preference for one
> over the other."

   I find myself bothered (especially in a US election week) with the
suggestion that there are only two possible outcomes.

   I view both being MTI as somewhat reasonable, and neither being MTI
as a likely best-choice (assuming we don't get real consensus for
either H.264 or VP8).

   I emphasize again that RTCWEB is designing for a long future, and
that we're almost certain to find a better choice than either of these
during that future. Thus I feel no obligation to choose the better
technical choice as MTI.

   Clearly there are technical differences; and it's possible that a
significant portion of the WG participants find a compelling argument
among the technical issues. (It would be nice to learn whether that's
true.)

   It's more obvious that there are IPR-related differences; and it
seems there are folks that find compelling arguments among these.
(It would be nice to learn whether clarifying any IPR issue would
help.)

   If we in fact _do_ find near-consensus for one or the other, I'm
happy to pursue consensus on the list -- but I'm led to doubt that
will be the result of discussion during IETF-85. If we're not near
consensus after the discussion, I'd _really_ like to have some
direction to the WGCs of how to proceed. Simple-Majority-Vote is
_not_ appropriate. Flipping a coin is not appropriate.

   I'm not fool enough to think my "something-like-H.261" suggestion
is popular. But it _is_ a plausible way forward...

> Because all 3 chairs and our responsible AD could be perceived to have
> conflicts of interest here, we have asked Robert Sparks to judge the
> consensus of the room, and he has kindly agreed.

   Clearly Robert should be empowered to ask follow-up questions if
appropriate. I won't ask him to pre-determine what these might be,
but I do hope he'll explore what might be an acceptable way forward.

--
John Leslie <john@jlc.net>

From matthew.kaufman@skype.net  Mon Oct 29 14:21:11 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Stephan Wenger <stewe@stewe.org>, Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Form of the video codec question
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Subject: Re: [rtcweb] Form of the video codec question
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stephan Wenger
> Sent: Monday, October 29, 2012 7:52 PM
> To: Ted Hardie; rtcweb@ietf.org
> Cc: Cullen Jennings
> Subject: Re: [rtcweb] Form of the video codec question
>=20
> Hi Ted, chairs,
>=20
> I think these are the wrong questions, as the result of such a poll is no=
t going
> to solve anything.  It's trivial to predict that there will be a signific=
ant show of
> hands in favor of each question.  What does that tell you?  Asking the
> questions as presented is going to be a waste of time.
>=20
> The more informative questions to as would be:
>=20
> 1) Do you object to H.264 as MTI, regardless of its technical and IPR mer=
its (if
> any); and
> 2) Do you object to VP8 as MTI, regardless of its technical and IPR merit=
s (if
> any) (both in more politically correct formulation, if you want.)
>

Agree. These are the questions I suggested originally. If the vast majority=
 of the people in the room raise their hands for *either one* of these, the=
n we should spend the time on other topics instead.

If one gets a significantly larger number than we can go on to whether or n=
ot we might actually get consensus.

Matthew Kaufman


From matthew.kaufman@skype.net  Mon Oct 29 14:23:57 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Ted Hardie <ted.ietf@gmail.com>, Stephan Wenger <stewe@stewe.org>
Thread-Topic: [rtcweb] Form of the video codec question
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ted Hardie
>=20
>=20
> So, this seems to be the inverse ("Can you not live with X as MTI").
> Why is asking this as the inverse going to give more information?

It isn't exactly the inverse. CANNOT is not the opposite of WOULD.

>=20
> We understand, of course, that the group has folks with strong opinions.
> They are vocal enough that no one is confused on this point.  But we hope=
 to
> understand, after a set of technical presentations, whether there is a
> sufficiently large group with a willingness to live with one or the other=
 to
> create a rough consensus.

I hope to avoid the set of technical presentations altogether if spending t=
ime on them is unlikely to change anyone's mind.

Matthew Kaufman


From ted.ietf@gmail.com  Mon Oct 29 14:33:56 2012
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On Mon, Oct 29, 2012 at 2:04 PM, John Leslie <john@jlc.netwrote:
>
>    I find myself bothered (especially in a US election week) with the
> suggestion that there are only two possible outcomes.
>

These are the only two proposals for which we have drafts, which was
noted before as a step along the path for this decision.  We have
previously established a consensus to have a mandatory-to-implement in
order to avoid negotiation failure.

>    I view both being MTI as somewhat reasonable, and neither being MTI
> as a likely best-choice (assuming we don't get real consensus for
> either H.264 or VP8).
>
>    I emphasize again that RTCWEB is designing for a long future, and
> that we're almost certain to find a better choice than either of these
> during that future. Thus I feel no obligation to choose the better
> technical choice as MTI.
>    Clearly there are technical differences; and it's possible that a
> significant portion of the WG participants find a compelling argument
> among the technical issues. (It would be nice to learn whether that's
> true.)
>
>    It's more obvious that there are IPR-related differences; and it
> seems there are folks that find compelling arguments among these.
> (It would be nice to learn whether clarifying any IPR issue would
> help.)
>
>    If we in fact _do_ find near-consensus for one or the other, I'm
> happy to pursue consensus on the list -- but I'm led to doubt that
> will be the result of discussion during IETF-85. If we're not near
> consensus after the discussion, I'd _really_ like to have some
> direction to the WGCs of how to proceed. Simple-Majority-Vote is
> _not_ appropriate. Flipping a coin is not appropriate.
>

We will discuss next steps after the result, whatever the result may
be.  Unless there is a long mic line advocating flipping a coin, I do
not see it as a likely next step.

>    I'm not fool enough to think my "something-like-H.261" suggestion
> is popular. But it _is_ a plausible way forward...
>
>> Because all 3 chairs and our responsible AD could be perceived to have
>> conflicts of interest here, we have asked Robert Sparks to judge the
>> consensus of the room, and he has kindly agreed.
>
>    Clearly Robert should be empowered to ask follow-up questions if
> appropriate. I won't ask him to pre-determine what these might be,
> but I do hope he'll explore what might be an acceptable way forward.
>

Follow-up questions are, of course, a possibility, and we will
obviously confirm anything on the list.

regards,

Ted Hardie

> --
> John Leslie <john@jlc.net>

From sergio.garcia.murillo@gmail.com  Mon Oct 29 15:40:19 2012
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Hi Ted,

Maybe I am completely wrong, but my understanding about the current 
situation is that there are two groups of people:

Group A: H264 folks, that want to have H264 as MTI (mainly for 
compatibility with current deployed base) but won't oppose to have VP8 
MTI in addition.
Group B: VP8 folks, that want to have VP8 as MTI but will oppose H264 to 
be also MTI.

So in the first sets of questions, A would raise the hands for both 1 
and 2, and B only for 2, which would be considered as a consensus to go 
for VP8.

In the second set of questions, A would not raise hands in neither 1 nor 
2, and B would only rise hands for 1, which would be considered as a 
consensus to go for VP8.

So at the end, both sets of question seems biased toward VP8 supporters.

Best regards
Sergio

El 29/10/2012 19:51, Stephan Wenger escribi:
> Hi Ted, chairs,
>
> I think these are the wrong questions, as the result of such a poll is not
> going to solve anything.  It's trivial to predict that there will be a
> significant show of hands in favor of each question.  What does that tell
> you?  Asking the questions as presented is going to be a waste of time.
>
> The more informative questions to as would be:
>
> 1) Do you object to H.264 as MTI, regardless of its technical and IPR
> merits (if any); and
> 2) Do you object to VP8 as MTI, regardless of its technical and IPR merits
> (if any)
> (both in more politically correct formulation, if you want.)
>
> If you chairs know that there are objections to both codecs by a
> significant number of people that cannot be swayed by technical or IPR
> arguments, a presentation of such technical and IPR merits can be omitted,
> and the WG can focus on stuff where progress is likely.
>
> Regards,
> Stephan
>
>
> On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>
>> We got feedback after the audio codec discussion that folks would have
>> benefited from seeing the question we intended to ask before the
>> meeting.  In order to clarify that for this discussion, the chairs are
>> putting forward this draft of the question which will be asked in the
>> meeting (and then confirmed on the list).
>>
>> "1) If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2) If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> You may raise your hand more than once and we encourage you to do so
>> if you can live with either, even if you have a preference for one
>> over the other."
>>
>> Because all 3 chairs and our responsible AD could be perceived to have
>> conflicts of interest here, we have asked Robert Sparks to judge the
>> consensus of the room, and he has kindly agreed.  If you have major
>> heartburn about the form of the question, please let us know.
>>
>> regards,
>>
>> Ted Hardie, Cullen Jennings, Magnus Westerlund
>> _______________________________________________
>> 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 hangzhou.chenxin@huawei.com  Mon Oct 29 21:14:42 2012
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From: Chenxin <hangzhou.chenxin@huawei.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Form of the video codec question
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+1=20
option 3 should be added to the choice : support both H.264 and VP8 as the =
mandatory to implement codec.

Best Regards,
     Xin=20


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf O=
f
>Sergio Garcia Murillo
>Sent: Tuesday, October 30, 2012 6:40 AM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Form of the video codec question
>
>Hi Ted,
>
>Maybe I am completely wrong, but my understanding about the current
>situation is that there are two groups of people:
>
>Group A: H264 folks, that want to have H264 as MTI (mainly for
>compatibility with current deployed base) but won't oppose to have VP8
>MTI in addition.
>Group B: VP8 folks, that want to have VP8 as MTI but will oppose H264 to
>be also MTI.
>
>So in the first sets of questions, A would raise the hands for both 1
>and 2, and B only for 2, which would be considered as a consensus to go
>for VP8.
>
>In the second set of questions, A would not raise hands in neither 1 nor
>2, and B would only rise hands for 1, which would be considered as a
>consensus to go for VP8.
>
>So at the end, both sets of question seems biased toward VP8 supporters.
>
>Best regards
>Sergio
>
>El 29/10/2012 19:51, Stephan Wenger escribi=F3:
>> Hi Ted, chairs,
>>
>> I think these are the wrong questions, as the result of such a poll is n=
ot
>> going to solve anything.  It's trivial to predict that there will be a
>> significant show of hands in favor of each question.  What does that tel=
l
>> you?  Asking the questions as presented is going to be a waste of time.
>>
>> The more informative questions to as would be:
>>
>> 1) Do you object to H.264 as MTI, regardless of its technical and IPR
>> merits (if any); and
>> 2) Do you object to VP8 as MTI, regardless of its technical and IPR meri=
ts
>> (if any)
>> (both in more politically correct formulation, if you want.)
>>
>> If you chairs know that there are objections to both codecs by a
>> significant number of people that cannot be swayed by technical or IPR
>> arguments, a presentation of such technical and IPR merits can be omitte=
d,
>> and the WG can focus on stuff where progress is likely.
>>
>> Regards,
>> Stephan
>>
>>
>> On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>>
>>> We got feedback after the audio codec discussion that folks would have
>>> benefited from seeing the question we intended to ask before the
>>> meeting.  In order to clarify that for this discussion, the chairs are
>>> putting forward this draft of the question which will be asked in the
>>> meeting (and then confirmed on the list).
>>>
>>> "1) If you support H.264 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> 2) If you support VP8 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>
>>> You may raise your hand more than once and we encourage you to do so
>>> if you can live with either, even if you have a preference for one
>>> over the other."
>>>
>>> Because all 3 chairs and our responsible AD could be perceived to have
>>> conflicts of interest here, we have asked Robert Sparks to judge the
>>> consensus of the room, and he has kindly agreed.  If you have major
>>> heartburn about the form of the question, please let us know.
>>>
>>> regards,
>>>
>>> Ted Hardie, Cullen Jennings, Magnus Westerlund
>>> _______________________________________________
>>> 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 Markus.Isomaki@nokia.com  Tue Oct 30 03:12:49 2012
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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: [rtcweb] Form of the video codec question
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Hi,

Sergio Garcia Murillo wrote:
>
>Hi Ted,
>
>Maybe I am completely wrong, but my understanding about the current
>situation is that there are two groups of people:
>
>Group A: H264 folks, that want to have H264 as MTI (mainly for compatibili=
ty
>with current deployed base) but won't oppose to have VP8 MTI in addition.
>Group B: VP8 folks, that want to have VP8 as MTI but will oppose H264 to b=
e
>also MTI.
>

I know that there are definitely people also in:
Group C: H264 folks that want to have H264 as MTI and will oppose VP8 as MT=
I.

Otherwise the issue would be much easier.

Markus

From Markus.Isomaki@nokia.com  Tue Oct 30 03:42:48 2012
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From: <Markus.Isomaki@nokia.com>
To: <stewe@stewe.org>, <ted.ietf@gmail.com>, <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Form of the video codec question
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Hi,

I prefer Stephan's formulation of the questions. To me the main issue does =
not seem to be the amount of people favoring a particular codec but objecti=
ng one. In case both H.264 and VP8 will get a large number of objections, a=
s I suspect, I think it is worhwhile also to ask once more something like:

* Raise your hand if you think the WG must decide on at least one MTI video=
 codec (at this time)
* Raise your hand if you think the WG should give up on MTI video codec (fo=
r now) and focus on making progress in other areas =20

Markus=20

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Stephan Wenger
>Sent: 29 October, 2012 20:52
>To: Ted Hardie; rtcweb@ietf.org
>Cc: Cullen Jennings
>Subject: Re: [rtcweb] Form of the video codec question
>
>Hi Ted, chairs,
>
>I think these are the wrong questions, as the result of such a poll is not=
 going
>to solve anything.  It's trivial to predict that there will be a significa=
nt show of
>hands in favor of each question.  What does that tell you?  Asking the
>questions as presented is going to be a waste of time.
>
>The more informative questions to as would be:
>
>1) Do you object to H.264 as MTI, regardless of its technical and IPR meri=
ts (if
>any); and
>2) Do you object to VP8 as MTI, regardless of its technical and IPR merits=
 (if
>any) (both in more politically correct formulation, if you want.)
>
>If you chairs know that there are objections to both codecs by a significa=
nt
>number of people that cannot be swayed by technical or IPR arguments, a
>presentation of such technical and IPR merits can be omitted, and the WG c=
an
>focus on stuff where progress is likely.
>
>Regards,
>Stephan
>
>
>On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>
>>We got feedback after the audio codec discussion that folks would have
>>benefited from seeing the question we intended to ask before the
>>meeting.  In order to clarify that for this discussion, the chairs are
>>putting forward this draft of the question which will be asked in the
>>meeting (and then confirmed on the list).
>>
>>"1) If you support H.264 as the mandatory to implement codec or are
>>willing to live with it as the MTI, please raise your hand now.
>>
>>2) If you support VP8 as the mandatory to implement codec or are
>>willing to live with it as the MTI, please raise your hand now.
>>
>>You may raise your hand more than once and we encourage you to do so if
>>you can live with either, even if you have a preference for one over
>>the other."
>>
>>Because all 3 chairs and our responsible AD could be perceived to have
>>conflicts of interest here, we have asked Robert Sparks to judge the
>>consensus of the room, and he has kindly agreed.  If you have major
>>heartburn about the form of the question, please let us know.
>>
>>regards,
>>
>>Ted Hardie, Cullen Jennings, Magnus Westerlund
>>_______________________________________________
>>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 fluffy@iii.ca  Tue Oct 30 03:52:37 2012
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So lets say we got agreement on something qualified with "regardless of =
its technical and IPR merits (if any)", I can't see how that helps us. =
We need to get agreement without that sort of qualification.=20

Can folks walk me through how this questions is better?


On Oct 30, 2012, at 11:42 , <Markus.Isomaki@nokia.com> wrote:

> Hi,
>=20
> I prefer Stephan's formulation of the questions. To me the main issue =
does not seem to be the amount of people favoring a particular codec but =
objecting one. In case both H.264 and VP8 will get a large number of =
objections, as I suspect, I think it is worhwhile also to ask once more =
something like:
>=20
> * Raise your hand if you think the WG must decide on at least one MTI =
video codec (at this time)
> * Raise your hand if you think the WG should give up on MTI video =
codec (for now) and focus on making progress in other areas =20
>=20
> Markus=20
>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>> Of ext Stephan Wenger
>> Sent: 29 October, 2012 20:52
>> To: Ted Hardie; rtcweb@ietf.org
>> Cc: Cullen Jennings
>> Subject: Re: [rtcweb] Form of the video codec question
>>=20
>> Hi Ted, chairs,
>>=20
>> I think these are the wrong questions, as the result of such a poll =
is not going
>> to solve anything.  It's trivial to predict that there will be a =
significant show of
>> hands in favor of each question.  What does that tell you?  Asking =
the
>> questions as presented is going to be a waste of time.
>>=20
>> The more informative questions to as would be:
>>=20
>> 1) Do you object to H.264 as MTI, regardless of its technical and IPR =
merits (if
>> any); and
>> 2) Do you object to VP8 as MTI, regardless of its technical and IPR =
merits (if
>> any) (both in more politically correct formulation, if you want.)
>>=20
>> If you chairs know that there are objections to both codecs by a =
significant
>> number of people that cannot be swayed by technical or IPR arguments, =
a
>> presentation of such technical and IPR merits can be omitted, and the =
WG can
>> focus on stuff where progress is likely.
>>=20
>> Regards,
>> Stephan
>>=20
>>=20
>> On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>>=20
>>> We got feedback after the audio codec discussion that folks would =
have
>>> benefited from seeing the question we intended to ask before the
>>> meeting.  In order to clarify that for this discussion, the chairs =
are
>>> putting forward this draft of the question which will be asked in =
the
>>> meeting (and then confirmed on the list).
>>>=20
>>> "1) If you support H.264 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>=20
>>> 2) If you support VP8 as the mandatory to implement codec or are
>>> willing to live with it as the MTI, please raise your hand now.
>>>=20
>>> You may raise your hand more than once and we encourage you to do so =
if
>>> you can live with either, even if you have a preference for one over
>>> the other."
>>>=20
>>> Because all 3 chairs and our responsible AD could be perceived to =
have
>>> conflicts of interest here, we have asked Robert Sparks to judge the
>>> consensus of the room, and he has kindly agreed.  If you have major
>>> heartburn about the form of the question, please let us know.
>>>=20
>>> regards,
>>>=20
>>> Ted Hardie, Cullen Jennings, Magnus Westerlund
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Cullen Jennings <fluffy@iii.ca>, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [rtcweb] Form of the video codec question
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Cc: "fluffy@cisco.com" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Form of the video codec question
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Cullen Jennings:
>=20
> So lets say we got agreement on something qualified with "regardless of i=
ts
> technical and IPR merits (if any)", I can't see how that helps us. We nee=
d to
> get agreement without that sort of qualification.
>=20
> Can folks walk me through how this questions is better?
>

It doesn't hardly matter *why*... my point of a qualification was covered i=
n my original questions:

Raise your hand if you want only H.264 and no amount of presentation time w=
ill change that to VP8
=09
Raise your hand if you want only VP8 and no amount of presentation time wil=
l change that to H.264

If the sum of those hands is the majority, we have better ways to use our t=
ime than presenting to people who won't change their mind.

Matthew Kaufman
=20


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Oct 2012 14:02:30 +0100
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Subject: Re: [rtcweb] JSEP-02 Comments
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2012/10/25 Christer Holmberg <christer.holmberg@ericsson.com>:
> Some comments on JSEP-02.
>
> Q1: Cloning
>
> I saw that you have removed the text about cloning from version -02.

So is it already decided that local PeerConnection cloning will not exist?

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

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Subject: Re: [rtcweb] #1: RFC 4734 citation
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2012/10/23 rtcweb issue tracker <trac+rtcweb@trac.tools.ietf.org>:
>  Telephone Event - [RFC4734]

Could somebody give some points about why DTMFs over media are
required in WWW? Isn't DTMF mechanism OLD enough to incorporate it
into WebRTC?

Autoanswer: interoperability with legacy telephony world.

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

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Form of the video codec question
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On Oct 30, 2012, at 11:42 , Markus.Isomaki@nokia.com wrote:

>=20
> * Raise your hand if you think the WG must decide on at least one MTI vid=
eo codec (at this time)
> * Raise your hand if you think the WG should give up on MTI video codec (=
for now) and focus on making progress in other areas =20

Agree this is critical but we took that previously and there was consensus =
for a MTI video codec.=20



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Subject: Re: [rtcweb] #1: RFC 4734 citation
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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: [rtcweb] Form of the video codec question
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Subject: Re: [rtcweb] Form of the video codec question
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Hi,

Cullen Jennings (fluffy) [mailto:fluffy@cisco.com] wrote:
>
>On Oct 30, 2012, at 11:42 , Markus.Isomaki@nokia.com wrote:
>
>>
>> * Raise your hand if you think the WG must decide on at least one MTI vi=
deo
>codec (at this time)
>> * Raise your hand if you think the WG should give up on MTI video codec
>(for now) and focus on making progress in other areas
>
>Agree this is critical but we took that previously and there was consensus=
 for a
>MTI video codec.
>

True. I would just note that if I recall correctly that took place 15 month=
s ago, which is a long time. Not sure if it has been refreshed afterwards.=
=20

Markus


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#1: RFC 4734 citation


Comment (by fluffy@…):

 This is a bug on draft-ietf-rtcweb-audio-00.txt

-- 
--------------------------------+-------------------------
 Reporter:  bernard_aboba@…     |       Owner:  jmvalin@…
     Type:  defect              |      Status:  new
 Priority:  critical            |   Milestone:  milestone1
Component:  audio               |     Version:  1.0
 Severity:  Active WG Document  |  Resolution:
 Keywords:                      |
--------------------------------+-------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/1#comment:1>
rtcweb <http://tools.ietf.org/rtcweb/>


From ted.ietf@gmail.com  Tue Oct 30 07:14:04 2012
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Form of the video codec question
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On Tue, Oct 30, 2012 at 5:30 AM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:

> If the sum of those hands is the majority, we have better ways to use our time than presenting to people who won't change their mind.
>

The problem in determining that is the baseline.  If we have 10 people
object to each in a room full at 150, it looks like things are
actually fine for picking either.  This is not true, however, if the
total number of folks who would have raised their hand in support of
anything was 30.  By asking the positive question, we get a better
shot at understanding what the baseline support is, which is critical.

I also note that working in an SDO that requires consensus (even rough
consensus) frequently requires compromise.  There are lots of aspects
of our work that cause us to put the goal of advancing the
interoperable standard over advancing our own technical preferences.
The form of your question would cause us to downplay both that need
and the ability of our participants to take that step if needed.   I
certainly hope that each of our participants is considering that as
they way their answers.

regards,

Ted

From ron@debian.org  Tue Oct 30 10:33:23 2012
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From: Ron <ron@debian.org>
To: rtcweb@ietf.org
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Subject: Re: [rtcweb] Form of the video codec question
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On Tue, Oct 30, 2012 at 12:30:48PM +0000, Matthew Kaufman wrote:
> Cullen Jennings:
> > 
> > So lets say we got agreement on something qualified with "regardless of its
> > technical and IPR merits (if any)", I can't see how that helps us. We need to
> > get agreement without that sort of qualification.
> > 
> > Can folks walk me through how this questions is better?
> >
> 
> It doesn't hardly matter *why*... my point of a qualification was covered in
> my original questions:

It really does matter *why*.  Probably even more so than what.

> Raise your hand if you want only H.264 and no amount of presentation time
> will change that to VP8
> 
> Raise your hand if you want only VP8 and no amount of presentation time will
> change that to H.264

Giving weight to an objection that isn't required to come with any sort of
justification whatsoever is not a good method of technical decision-making.

Presupposing that objections simply attached to either of those questions
should carry equal weight and/or be of equal merit would also seem to lack
both engineering and statistical rigour.

> If the sum of those hands is the majority, we have better ways to use our
> time than presenting to people who won't change their mind.

If the sum of those hands is a majority, we should have used our time to
formulate more discriminating questions (and consensus answers) about the
problem space.

I agree that posing a question with one in a nonillion chance of arriving
at a coherent answer is not a good use of time.  From the discussion so
far though, there does seem to be a reasonable amount of consensus about
what the dilemmas of either choice are - so perhaps we should summarise
that and make some analysis of the merits and problems that we *do* all
seem to be in basic agreement about.

We might be surprised at how easily an actual, rational, answer could
become obviously apparent from that once we peel back the skin of
simplistic Do Not Want objections.

  Cheers,
  Ron



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To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Form of the video codec question
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On 30 October 2012 15:14, Ted Hardie <ted.ietf@gmail.com> wrote:
> The problem in determining that is the baseline.  If we have 10 people
> object to each in a room full at 150, it looks like things are
> actually fine for picking either.

Not as much so if the 10 comprise the browser vendors.  (Yes, we're
all individuals...)

From stewe@stewe.org  Tue Oct 30 11:12:13 2012
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From: Stephan Wenger <stewe@stewe.org>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>, "fluffy@cisco.com" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Form of the video codec question
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Hi, please see inline.

On 10.30.2012 06:51 , "Markus.Isomaki@nokia.com"
<Markus.Isomaki@nokia.com> wrote:

>Hi,
>
>Cullen Jennings (fluffy) [mailto:fluffy@cisco.com] wrote:
>>
>>On Oct 30, 2012, at 11:42 , Markus.Isomaki@nokia.com wrote:
>>
>>>
>>> * Raise your hand if you think the WG must decide on at least one MTI
>>>video
>>codec (at this time)
>>> * Raise your hand if you think the WG should give up on MTI video codec
>>(for now) and focus on making progress in other areas
>>
>>Agree this is critical but we took that previously and there was
>>consensus for a
>>MTI video codec.
>>
>
>True. I would just note that if I recall correctly that took place 15
>months ago, which is a long time. Not sure if it has been refreshed
>afterwards.=20

Yes.  Even more importantly, arguments were exchanged for 15 months.
Individuals had 15 months time to formulate their positions.  More
importantly for me (albeit moderately politically incorrect), the
employers of those individual also have had 15 months time to formulate
their positions, and advise their employees accordingly.  I would hope we
all understand that both individual positions and guidance are in most
cases "hard".  No half hour presentation will swing those positions.

Stephan


>
>Markus
>
>



From partha@parthasarathi.co.in  Tue Oct 30 11:18:06 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>
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Date: Tue, 30 Oct 2012 23:47:56 +0530
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Cullen,

PRANSWER open issue is related to JSEP State machine wherein PRANSWER state
could not handle OFFER event. Offer event in PRANSWER state is required for
the early-dialog UPDATE Offer and on the other hand, ANSWER event in
PRANSWER state is required by SIP serial forking. Also, the issue is not
related to signaling problem.  

Christer,

IIRC, cloning is removed as there is a consensus not to have parallel
forking in one of the earlier interim meeting.

Thanks
Partha

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Christer Holmberg
Sent: Monday, October 29, 2012 6:14 PM
To: Cullen Jennings (fluffy)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-02 Comments

Hi,
 
>> Q1: Cloning
>>  
>> I saw that you have removed the text about cloning from version -02.
>>  
>> That is a rather fundamental change, so I think the change log should say
a little more than "Clarified section on forking.".
>
> agree

And, do we have a WG concensus to remove cloning?

>> Q4: Remove offer
>>  
>> This is related to the PRANSWER issue.
>>  
>> Unless I've missed it, the draft does not address the issue raised on the
list, where an offer is received from the remote peer while in PRANSWER
state.
>>  
>> (If you solved it on the list, I applogize for not having read it. I've
had a flu the whole week, so I am a little behind with my e-mails)]
>
> that's a signaling problem. I doubt the draft should address it any more
than there are a few ways of dealing with it and Javascript can do whatever
it wants to choose which one it wants. 

It's a use-case I'd like the API to support.

But, even without PRANSWER it would be tricky, as you have removed cloning.

Regards,

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


From bernard_aboba@hotmail.com  Tue Oct 30 11:47:16 2012
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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: "'Parthasarathi R'" <partha@parthasarathi.co.in>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] JSEP-02 Comments
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Partha said: 

"Christer,

IIRC, cloning is removed as there is a consensus not to have parallel
forking in one of the earlier interim meeting.

Thanks
Partha"

[BA] Can you send a pointer to the consensus determination you are referring
to?  Also, I cannot find a consensus call on this subject sent to the RTCWEB
mailing list.   If in fact a consensus determination has been made (either
in IETF or W3C), can someone send a pointer to it? 


From christer.holmberg@ericsson.com  Tue Oct 30 12:14:13 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Bernard Aboba' <bernard_aboba@hotmail.com>, 'Parthasarathi R' <partha@parthasarathi.co.in>
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Hi,

There has been discussions about using cloning for SERIAL forking, because =
it would solve some issues.

Regards,

Christer=20

-----Original Message-----
From: Bernard Aboba [mailto:bernard_aboba@hotmail.com]=20
Sent: 30. lokakuuta 2012 20:47
To: 'Parthasarathi R'; Christer Holmberg
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] JSEP-02 Comments

Partha said:=20

"Christer,

IIRC, cloning is removed as there is a consensus not to have parallel forki=
ng in one of the earlier interim meeting.

Thanks
Partha"

[BA] Can you send a pointer to the consensus determination you are referrin=
g to?  Also, I cannot find a consensus call on this subject sent to the RTC=
WEB
mailing list.   If in fact a consensus determination has been made (either
in IETF or W3C), can someone send a pointer to it?=20


From adam@nostrum.com  Tue Oct 30 15:33:53 2012
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On 10/29/12 16:04, Oct 29, John Leslie wrote:
>     I view both being MTI as somewhat reasonable, and neither being MTI
> as a likely best-choice

That's an interesting set of assertions. I, for one, cannot see how you 
reached either conclusion.

Selecting "both" solves none of the problems that have been raised to 
date. It takes on the entire union of disadvantages that have been 
raised about the candidate codecs, while providing the benefits of neither.

Selecting "neither" is intentionally aiming the ship for the rocks: we 
guarantee failure. If we publish a specification to which two 
implementations can comply but cannot subsequently inter-operate, then 
the entire point of standardization is defeated.

/a

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On 10/30/12 07:30, Oct 30, Matthew Kaufman wrote:
> my point of a qualification was covered in my original questions:
>
> Raise your hand if you want only H.264 and no amount of presentation time will change that to VP8
> 	
> Raise your hand if you want only VP8 and no amount of presentation time will change that to H.264
>
> If the sum of those hands is the majority, we have better ways to use our time than presenting to people who won't change their mind.

*If* those questions were asked (and I like Ted's formulation better), 
*and* the sum of the hands comprised the majority, then I would agree 
that we have better ways to use our time than presenting technical 
issues to the entire WG. In my opinion, foremost among those would be 
selecting which of the mechanisms in section 4 of RFC 3929 the WG thinks 
is most applicable.

/a

From stewe@stewe.org  Tue Oct 30 18:41:24 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Adam Roach <adam@nostrum.com>, John Leslie <john@jlc.net>
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Hi Adam,

On 10.30.2012 15:33 , "Adam Roach" <adam@nostrum.com> wrote:

>On 10/29/12 16:04, Oct 29, John Leslie wrote:
>>     I view both being MTI as somewhat reasonable, and neither being MTI
>> as a likely best-choice
>
>That's an interesting set of assertions. I, for one, cannot see how you
>reached either conclusion.
>
>Selecting "both" solves none of the problems that have been raised to
>date. It takes on the entire union of disadvantages that have been
>raised about the candidate codecs, while providing the benefits of
>neither.
>
>Selecting "neither" is intentionally aiming the ship for the rocks: we
>guarantee failure. If we publish a specification to which two
>implementations can comply but cannot subsequently inter-operate, then
>the entire point of standardization is defeated.

I disagree.  It just takes a commercial decision making process out of the
hands of IETF geeks and into the hands of (hopefully) competent business
people.  Given the money and risk involved, that is sensible.

SDOs can do wonderful things, but sorting out this dispute is not one of
them.

Stephan

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



From ron@debian.org  Tue Oct 30 21:43:42 2012
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Subject: Re: [rtcweb] Form of the video codec question
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On Wed, Oct 31, 2012 at 01:41:10AM +0000, Stephan Wenger wrote:
> On 10.30.2012 15:33 , "Adam Roach" <adam@nostrum.com> wrote:
> 
> >On 10/29/12 16:04, Oct 29, John Leslie wrote:
> >>     I view both being MTI as somewhat reasonable, and neither being MTI
> >> as a likely best-choice
> >
> >That's an interesting set of assertions. I, for one, cannot see how you
> >reached either conclusion.
> >
> >Selecting "both" solves none of the problems that have been raised to
> >date. It takes on the entire union of disadvantages that have been
> >raised about the candidate codecs, while providing the benefits of
> >neither.
> >
> >Selecting "neither" is intentionally aiming the ship for the rocks: we
> >guarantee failure. If we publish a specification to which two
> >implementations can comply but cannot subsequently inter-operate, then
> >the entire point of standardization is defeated.
> 
> I disagree.  It just takes a commercial decision making process out of the
> hands of IETF geeks and into the hands of (hopefully) competent business
> people.

Didn't this group form precisely because the hopefully competent business
people have so far only produced a diverse number of unacceptable solutions
and people *really* wanted something better than that?

The choice of MTI codec here isn't a commercial decision, it's a technical
one.  If we make the right choice, we will ensure the broadest number of
implementors have the option to use it, and the broadest number of
implementations will be interoperable.

There are probably perfectly good "commercial decision making processes"
which might arrive at the conclusion that this is not in their best
interest and is something that should be opposed and stalled at every
step of the way.  That is their right.  But it's not the charter of this
group ...


I think the concerns Adam voiced are both real and outcomes to be avoided.


 Ron



From matthew.kaufman@skype.net  Tue Oct 30 23:26:56 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Ron <ron@debian.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Form of the video codec question
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Subject: Re: [rtcweb] Form of the video codec question
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> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ron
> Sent: Wednesday, October 31, 2012 5:43 AM
>
> Didn't this group form precisely because the hopefully competent business
> people have so far only produced a diverse number of unacceptable
> solutions and people *really* wanted something better than that?

No, I don't believe that was the motivation for forming the RTCWEB working =
group.
=20
> The choice of MTI codec here isn't a commercial decision, it's a technica=
l one.
> If we make the right choice, we will ensure the broadest number of
> implementors have the option to use it, and the broadest number of
> implementations will be interoperable.

No, we won't. No matter what this WG decides, people who are not in the roo=
m will ultimately make the decision as to what business model they wish to =
pursue and what business risks they are willing to take... and thus whether=
 or not their implementation will interoperate with others.
 =09
Matthew Kaufman


From ron.even.tlv@gmail.com  Tue Oct 30 23:38:42 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Adam Roach'" <adam@nostrum.com>, "'John Leslie'" <john@jlc.net>
References: <CA+9kkMCQBw27XwvZ=hARs1DyyQufY2H7R4FwYq-3P-Tp7T7W+w@mail.gmail.com>	<20121029210443.GB68232@verdi> <509055C4.6060500@nostrum.com>
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Hi,
Just to be precise, if we do not chose a common video codec the
interoperability point will be an audio call
Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Adam Roach
Sent: 31 October, 2012 12:34 AM
To: John Leslie
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Form of the video codec question

On 10/29/12 16:04, Oct 29, John Leslie wrote:
>     I view both being MTI as somewhat reasonable, and neither being 
> MTI as a likely best-choice

That's an interesting set of assertions. I, for one, cannot see how you
reached either conclusion.

Selecting "both" solves none of the problems that have been raised to date.
It takes on the entire union of disadvantages that have been raised about
the candidate codecs, while providing the benefits of neither.

Selecting "neither" is intentionally aiming the ship for the rocks: we
guarantee failure. If we publish a specification to which two
implementations can comply but cannot subsequently inter-operate, then the
entire point of standardization is defeated.

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


From lorenzo@meetecho.com  Tue Oct 30 23:56:32 2012
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Date: Wed, 31 Oct 2012 07:56:26 +0100
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: "Roni Even" <ron.even.tlv@gmail.com>
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On Wed, 31 Oct 2012 08:36:26 +0200
"Roni Even" <ron.even.tlv@gmail.com> wrote:

> Hi,
> Just to be precise, if we do not chose a common video codec the
> interoperability point will be an audio call
> Roni
> 


Not if we pick an "older" video codec as a MTI one as we did for G.711 (well maybe not *that* old).

L.


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Adam Roach
> Sent: 31 October, 2012 12:34 AM
> To: John Leslie
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Form of the video codec question
> 
> On 10/29/12 16:04, Oct 29, John Leslie wrote:
> >     I view both being MTI as somewhat reasonable, and neither being 
> > MTI as a likely best-choice
> 
> That's an interesting set of assertions. I, for one, cannot see how you
> reached either conclusion.
> 
> Selecting "both" solves none of the problems that have been raised to date.
> It takes on the entire union of disadvantages that have been raised about
> the candidate codecs, while providing the benefits of neither.
> 
> Selecting "neither" is intentionally aiming the ship for the rocks: we
> guarantee failure. If we publish a specification to which two
> implementations can comply but cannot subsequently inter-operate, then the
> entire point of standardization is defeated.
> 
> /a
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Lorenzo Miniero, COB

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

From oej@edvina.net  Wed Oct 31 00:23:33 2012
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To: Stephan Wenger <stewe@stewe.org>
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Subject: Re: [rtcweb] Form of the video codec question
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31 okt 2012 kl. 02:41 skrev Stephan Wenger <stewe@stewe.org>:

> Hi Adam,
>=20
> On 10.30.2012 15:33 , "Adam Roach" <adam@nostrum.com> wrote:
>=20
>> On 10/29/12 16:04, Oct 29, John Leslie wrote:
>>>    I view both being MTI as somewhat reasonable, and neither being =
MTI
>>> as a likely best-choice
>>=20
>> That's an interesting set of assertions. I, for one, cannot see how =
you
>> reached either conclusion.
>>=20
>> Selecting "both" solves none of the problems that have been raised to
>> date. It takes on the entire union of disadvantages that have been
>> raised about the candidate codecs, while providing the benefits of
>> neither.
>>=20
>> Selecting "neither" is intentionally aiming the ship for the rocks: =
we
>> guarantee failure. If we publish a specification to which two
>> implementations can comply but cannot subsequently inter-operate, =
then
>> the entire point of standardization is defeated.
>=20
> I disagree.  It just takes a commercial decision making process out of =
the
> hands of IETF geeks and into the hands of (hopefully) competent =
business
> people.  Given the money and risk involved, that is sensible.
>=20
Ending up with a situation where users have to find out which browser to
use in order to interoperate with one specific WebRTC application is a =
big failure.

Like in the days of web pages saying "Sorry, this web page only supports =
on Huckzilla Explorer v5."

I agree with Adam, that would defeat the point of standardization. We =
have
that mess with SIP SIMPLE - there are many implementations that fail to
interoperate, thus SIP SIMPLE is not taking off outside of walled =
gardens.

We want interoperability, we don't want more plugins or having to select
the right web browser to reach your application.

/O=

From ron@debian.org  Wed Oct 31 01:15:02 2012
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From: Ron <ron@debian.org>
To: rtcweb@ietf.org
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Subject: Re: [rtcweb] Form of the video codec question
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On Wed, Oct 31, 2012 at 06:26:26AM +0000, Matthew Kaufman wrote:
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Ron
> > Sent: Wednesday, October 31, 2012 5:43 AM
> >
> > Didn't this group form precisely because the hopefully competent business
> > people have so far only produced a diverse number of unacceptable
> > solutions and people *really* wanted something better than that?
> 
> No, I don't believe that was the motivation for forming the RTCWEB working group.

 "Description of Working Group:

  There are a number of proprietary implementations that provide direct
  interactive rich communication using audio, video, collaboration,
  games, etc. between two peers' web-browsers. These are not
  interoperable, as they require non-standard extensions or plugins to
  work.  There is a desire to standardize the basis for such
  communication so that interoperable communication can be established
  between any compatible browsers.  ..."

You can read the rest here: http://tools.ietf.org/wg/rtcweb/charters

Which bit of that do you not believe, or think is unclear?


> > The choice of MTI codec here isn't a commercial decision, it's a technical one.
> > If we make the right choice, we will ensure the broadest number of
> > implementors have the option to use it, and the broadest number of
> > implementations will be interoperable.
> 
> No, we won't. No matter what this WG decides, people who are not in the room
> will ultimately make the decision as to what business model they wish to
> pursue and what business risks they are willing to take... and thus whether
> or not their implementation will interoperate with others.

And that's fine, but totally irrelevant to the technical decisions of the
working group.  If those people want their opinions and technical arguments
to be taken into account, then hiding in another room might not be the best
strategy they could employ to air them.

It would certainly be odd to abandon the charter because of rumours that
people not here didn't want interoperability with others.  Wouldn't it?

We can't stop them from persisting with being part of the problem that the
group was chartered to solve.  But we likewise shouldn't let them stop us
from solving it in the best and most broadly accessible way possible anyhow.

There are always going to be business models that feel threatened by an
open standard that can be freely implemented by anyone.  That doesn't
mean that people don't want and need them, or that we shouldn't make them
and collaborate to make them the best they can be.


  Colloidially,
  Ron



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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2012/10/30 Christer Holmberg <christer.holmberg@ericsson.com>:
> There has been discussions about using cloning for SERIAL forking, becaus=
e it would solve some issues.

PRANSWER just solves a specific *SIP* usecase or need (but not all).

Cloning would solve all the related problems and would allow
implementing any kind of logic on top of WebRTC (including full SIP).


So IMHO PRANSWER looks like a hack introduced by SIP guys into WebRTC,
regardless it makes no sense at all in the WWW world.

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

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 31 Oct 2012 12:13:18 +0100
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Subject: Re: [rtcweb] JSEP-02 Comments
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2012/10/31 Matthew Kaufman <matthew.kaufman@skype.net>:
> Well, since using SDP Offer/Answer as the API is a "hack introduced by SI=
P guys into WebRTC", I'm not at all surprised to see requirements for thing=
s like PRANSWER showing up.

Yes, I understand, but the real requirement for really respecting RFC
3264 was implementing cloning (see below please), instead of PRANSWER
which is a pseudo-hack for allowing some specific and existing use
cases of SIP (but not all).

If I make a call, receive two 183 with early media (different
early-dialogs), and want to render both audio streams locally, how to
do that with PRANSWER? Cloning was the key for that.


> Next you're going to wonder why a single RTCPeerConnection object represe=
nts open ICE sessions with RTP media flowing between this browser and *any =
number* of remote peers.

AFAIR the "cloning" proposal means that you end with N
RTCPeerConnection instances, each of them having media with a single
remote peer. The only they share is the same local SDP.


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

From singer@apple.com  Wed Oct 31 06:08:35 2012
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Subject: Re: [rtcweb] Form of the video codec question
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On Oct 31, 2012, at 5:43 , Ron <ron@debian.org> wrote:

> 
> The choice of MTI codec here isn't a commercial decision, it's a technical
> one.  

Not at all;  there are any number of codecs which would be adequate as the catch-all 'fallback' codec.  The willingness, or ability, to license, and implement, are key -- commercial -- questions, and the essence of the debate. 

> If we make the right choice, we will ensure the broadest number of
> implementors have the option to use it, and the broadest number of
> implementations will be interoperable.

Our choice is only part of the solution; it has to be viable and actually implemented.  I would hazard to guess that what most will implement is a choice made almost entirely indpendently of any MTI tag in the spec.

> 
> There are probably perfectly good "commercial decision making processes"
> which might arrive at the conclusion that this is not in their best
> interest and is something that should be opposed and stalled at every
> step of the way.  That is their right.  But it's not the charter of this
> group ...
> 
> 
> I think the concerns Adam voiced are both real and outcomes to be avoided.

David Singer
Multimedia and Software Standards, Apple Inc.


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Subject: Re: [rtcweb] Form of the video codec question
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-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Matthew Kaufman
Sent: Wednesday, October 31, 2012 2:26 AM
To: Ron; rtcweb@ietf.org
Subject: Re: [rtcweb] Form of the video codec question

> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
> Behalf Of Ron
> Sent: Wednesday, October 31, 2012 5:43 AM
>
> Didn't this group form precisely because the hopefully competent 
> business people have so far only produced a diverse number of 
> unacceptable solutions and people *really* wanted something better than
that?

No, I don't believe that was the motivation for forming the RTCWEB working
group.
 
> The choice of MTI codec here isn't a commercial decision, it's a technical
one.
> If we make the right choice, we will ensure the broadest number of 
> implementors have the option to use it, and the broadest number of 
> implementations will be interoperable.

No, we won't. No matter what this WG decides, people who are not in the room
will ultimately make the decision as to what business model they wish to
pursue and what business risks they are willing to take... and thus whether
or not their implementation will interoperate with others.
[RS> ] 
[RS> ] <sigh>  This is true of any technical/engineering specification. So
what's your point. This particular line of objection is fundamentally
irrelevant to the topic at hand. Adam is correct. Selecting "whatever you
want to do" is not a option. 

 	
Matthew Kaufman

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


From ted.ietf@gmail.com  Wed Oct 31 07:21:44 2012
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On Tue, Oct 30, 2012 at 11:26 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
> No, we won't. No matter what this WG decides, people who are not in the r=
oom will ultimately make the decision as to what business model they wish t=
o pursue and what business risks they are willing to take... and thus wheth=
er or not their implementation will interoperate with others.


Perhaps one way to recast this would be that the working group's
decision will affect whether or not an implementation which decides
not to interoperate can claim to support WebRTC or not.  That may have
an impact on the business decision; it may not.  As you say, there are
other forces.  But I believe it is clear that if the working group
decides to abandon the effort to achieve interoperability that the
chances of the market producing it go down, not up.

Speaking personally, neither as chair nor as company representative,

Ted Hardie

From harald@alvestrand.no  Wed Oct 31 08:19:02 2012
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Subject: Re: [rtcweb] Form of the video codec question
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On 10/29/2012 07:51 PM, Stephan Wenger wrote:
> Hi Ted, chairs,
>
> I think these are the wrong questions, as the result of such a poll is not
> going to solve anything.  It's trivial to predict that there will be a
> significant show of hands in favor of each question.  What does that tell
> you?  Asking the questions as presented is going to be a waste of time.
>
> The more informative questions to as would be:
>
> 1) Do you object to H.264 as MTI, regardless of its technical and IPR
> merits (if any); and
> 2) Do you object to VP8 as MTI, regardless of its technical and IPR merits
> (if any)
> (both in more politically correct formulation, if you want.)

Stephan,

I would not raise my hand on any of these questions, but would 
definitely protest a decision made based on the outcome of that question.

All my objections to H.264 as MTI are based on its technical and IPR merits.

If those change, my opinion may change.

However, I believe that my opinion is based enough in facts that it's 
unlikely that I'll be swayed by a presentation given at the IETF; 
reading the drafts of the H.264 proponents certainly did not change it.


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Subject: Re: [rtcweb] #1: RFC 4734 citation
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On 10/30/2012 02:06 PM, Iñaki Baz Castillo wrote:
> 2012/10/23 rtcweb issue tracker <trac+rtcweb@trac.tools.ietf.org>:
>>   Telephone Event - [RFC4734]
> Could somebody give some points about why DTMFs over media are
> required in WWW? Isn't DTMF mechanism OLD enough to incorporate it
> into WebRTC?
>
> Autoanswer: interoperability with legacy telephony world.
>
Autoanswer (and you should know this by heart now, it's been said so 
many times):

Use case 4.3.2 from draft-ietf-rtcweb-use-cases-and-requirements-09:

4.3.2.  Fedex Call

4.3.2.1.  Description

    Alice uses her web browser with a service something like Skype to be
    able to phone PSTN numbers.  Alice calls 1-800-gofedex.  Alice should
    be able to hear the initial prompts from the fedex IVR and when the
    IVR says press 1, there should be a way for Alice to navigate the
    IVR.

If you want to argue for having that use case deleted from the 
requirements, start a debate on that topic. If you have an alternative 
solution, state that in the positive as what should go into the -audio- 
draft instead of RFC 4734.

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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Form of the video codec question
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Harald Alvestrand:
>
> However, I believe that my opinion is based enough in facts that it's unl=
ikely
> that I'll be swayed by a presentation given at the IETF; reading the draf=
ts of
> the H.264 proponents certainly did not change it.
>

Perfect. If everyone can read the drafts ahead of time and make up their mi=
nd then we can spend valuable face-to-face time on the existing issues plus=
 the three (or more) new items that are coming out of W3C this week instead=
 of on the predetermined codec discussion.

Matthew Kaufman


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2012/10/31 Harald Alvestrand <harald@alvestrand.no>

> On 10/30/2012 02:06 PM, I=C3=B1aki Baz Castillo wrote:
>
>> 2012/10/23 rtcweb issue tracker <trac+rtcweb@trac.tools.ietf.**org<trac%=
2Brtcweb@trac.tools.ietf.org>
>> >:
>>
>>>   Telephone Event - [RFC4734]
>>>
>> Could somebody give some points about why DTMFs over media are
>> required in WWW? Isn't DTMF mechanism OLD enough to incorporate it
>> into WebRTC?
>>
>> Autoanswer: interoperability with legacy telephony world.
>>
>>  Autoanswer (and you should know this by heart now, it's been said so
> many times):
>
> Use case 4.3.2 from draft-ietf-rtcweb-use-cases-**and-requirements-09:
>
> 4.3.2.  Fedex Call
>
> 4.3.2.1.  Description
>
>    Alice uses her web browser with a service something like Skype to be
>    able to phone PSTN numbers.  Alice calls 1-800-gofedex.  Alice should
>    be able to hear the initial prompts from the fedex IVR and when the
>    IVR says press 1, there should be a way for Alice to navigate the
>    IVR.
>
> If you want to argue for having that use case deleted from the
> requirements, start a debate on that topic. If you have an alternative
> solution, state that in the positive as what should go into the -audio-
> draft instead of RFC 4734.



Well, as we all assume that media gateways are requried for
interoperability with other VoIP networks, I assume it would be not so hard
that the gateway receives the "DTMF's" via WWW means (HTTP, WebSocket...)
and generates RTP DTMF's in the legacy leg.

Anyhow I expect that it's not so hard to implement RTP DTMF's in a media
stack, but I would not like to hear about "my PBX's does not detect
Chrome's DTMF's". :)


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

--bcaec54ee77ceb279804cd5e0344
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2012/10/31 Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald=
@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span><br><d=
iv class=3D"gmail_extra"><div class=3D"gmail_quote"><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"im">On 10/30/2012 02:06 PM, I=C3=B1aki Baz Castillo wrote:<br=
>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
2012/10/23 rtcweb issue tracker &lt;<a href=3D"mailto:trac%2Brtcweb@trac.to=
ols.ietf.org" target=3D"_blank">trac+rtcweb@trac.tools.ietf.<u></u>org</a>&=
gt;:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 Telephone Event - [RFC4734]<br>
</blockquote>
Could somebody give some points about why DTMFs over media are<br>
required in WWW? Isn&#39;t DTMF mechanism OLD enough to incorporate it<br>
into WebRTC?<br>
<br>
Autoanswer: interoperability with legacy telephony world.<br>
<br>
</blockquote></div>
Autoanswer (and you should know this by heart now, it&#39;s been said so ma=
ny times):<br>
<br>
Use case 4.3.2 from draft-ietf-rtcweb-use-cases-<u></u>and-requirements-09:=
<br>
<br>
4.3.2. =C2=A0Fedex Call<br>
<br>
4.3.2.1. =C2=A0Description<br>
<br>
=C2=A0 =C2=A0Alice uses her web browser with a service something like Skype=
 to be<br>
=C2=A0 =C2=A0able to phone PSTN numbers. =C2=A0Alice calls 1-800-gofedex. =
=C2=A0Alice should<br>
=C2=A0 =C2=A0be able to hear the initial prompts from the fedex IVR and whe=
n the<br>
=C2=A0 =C2=A0IVR says press 1, there should be a way for Alice to navigate =
the<br>
=C2=A0 =C2=A0IVR.<br>
<br>
If you want to argue for having that use case deleted from the requirements=
, start a debate on that topic. If you have an alternative solution, state =
that in the positive as what should go into the -audio- draft instead of RF=
C 4734.</blockquote>

<div><br></div><div><br></div><div>Well, as we all assume that media gatewa=
ys are requried for interoperability with other VoIP networks, I assume it =
would be not so hard that the gateway receives the &quot;DTMF&#39;s&quot; v=
ia WWW means (HTTP, WebSocket...) and generates RTP DTMF&#39;s in the legac=
y leg.</div>

<div><br></div><div>Anyhow I expect that it&#39;s not so hard to implement =
RTP DTMF&#39;s in a media stack, but I would not like to hear about &quot;m=
y PBX&#39;s does not detect Chrome&#39;s DTMF&#39;s&quot;. :)</div></div>

<br clear=3D"all"><div><br></div>-- <br>I=C3=B1aki Baz Castillo<br>&lt;<a h=
ref=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div>

--bcaec54ee77ceb279804cd5e0344--

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From roman@telurix.com  Wed Oct 31 10:41:49 2012
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] #1: RFC 4734 citation
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On Wed, Oct 31, 2012 at 1:12 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Well, as we all assume that media gateways are requried for
> interoperability with other VoIP networks, I assume it would be not so ha=
rd
> that the gateway receives the "DTMF's" via WWW means (HTTP, WebSocket...)
> and generates RTP DTMF's in the legacy leg.
>
> Anyhow I expect that it's not so hard to implement RTP DTMF's in a media
> stack, but I would not like to hear about "my PBX's does not detect
> Chrome's DTMF's". :)
>

We had an extensive discussion few months back on why RFC 4733 tones are
required. The main reason is time synchronization of audio and DTMF. If you
have an application which does something like "record your name and press
pound", and DTMF is delivered over a separate path, you end up with name
either truncated or padded with silence. In any case, I believe, there was
a group consensus that RFC 4733 DTMF tones will be supported.
_____________
Roman Shpount

--e89a8ffbaaf5a990ee04cd5e6aac
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<br clear=3D"all">On Wed, Oct 31, 2012 at 1:12 PM, I=F1aki 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><div class=3D"gmail_quote"><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex">
<div class=3D"gmail_extra"><div class=3D"gmail_quote"><div>Well, as we all =
assume that media gateways are requried for interoperability with other VoI=
P networks, I assume it would be not so hard that the gateway receives the =
&quot;DTMF&#39;s&quot; via WWW means (HTTP, WebSocket...) and generates RTP=
 DTMF&#39;s in the legacy leg.</div>


<div><br></div><div>Anyhow I expect that it&#39;s not so hard to implement =
RTP DTMF&#39;s in a media stack, but I would not like to hear about &quot;m=
y PBX&#39;s does not detect Chrome&#39;s DTMF&#39;s&quot;. :)</div></div>
</div></blockquote><div><br></div><div>We had an extensive discussion few m=
onths back on why RFC 4733 tones are required. The main reason is time=A0sy=
nchronization=A0of audio and DTMF. If you have an application which does so=
mething like &quot;record your name and press pound&quot;, and DTMF is deli=
vered over a separate path, you end up with name either truncated or padded=
 with silence. In any case, I believe, there was a group consensus that=A0R=
FC 4733 DTMF tones will be supported.</div>
_____________<br>Roman Shpount<br><div>=A0</div></div><br>

--e89a8ffbaaf5a990ee04cd5e6aac--

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

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From andrew.hutton@siemens-enterprise.com  Wed Oct 31 15:39:09 2012
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Subject: [rtcweb]  #3: JSEP-02 Review (from Andy Hutton)
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#3: JSEP-02 Review (from Andy Hutton)

 I have read through the latest jsep-02 draft and have the following
 comments/questions.

 1. Section 1 Introduction

 The introduction makes various references to offer/answer but does not
 mention RFC 3264.  It needs such a reference and to explain the relevance
 of RFC 3264 and how JSEP provides the tools to implement a RFC 3264
 conformant application but does not itself implement RFC 3264.


 2. Section 1 Introduction 2nd Para states:

 "The browser environment also has its own challenges that cause problems
 for an embedded signaling state machine.  One of these is that the user
 may reload the web page at any time.  If this happens, and the state
 machine is being run at a server, the server can simply push the current
 state back down to the page and resume the call where it left off".

 The last sentence is an over simplification as there are problems with
 regard to rehydration even when the signaling state machine is run at the
 server. Also the sentence refers to two different types of state as it is
 not the signaling state that is pushed back to the browser.  I suggest
 removing the last sentence or referring to the section on hydration.


 3. Section 1. Introduction 7th Para states:

 "The JSEP approach does come with a minor downside.  As the application
 now is responsible for driving the signaling state  machine, slightly more
 application code is necessary to perform call setup; the application must
 call the right APIs at the right times, and convert the session
 descriptions and ICE information into the  defined messages of its chosen
 signaling protocol, instead of simply forwarding the messages emitted from
 the browser."

 Yes but maybe the introduction should explain that the API will provide
 valid SDP that an application could use within a RFC 3264 based protocol
 without the application needing to understand the syntax of SDP.


 4. Section 1. Intoduction 8th Para states:

 "For example, this library could convert easily adapt the JSEP API..."

 Remove "convert".

 5. Section 4.1. Signaling model 1st Para states:

 "JSEP does not specify a particular signaling model or state machine,
 other than the generic need to exchange RFC 3264 offers and answers in
 order for both sides of the session to know how to conduct the session".

 I think this needs rewording and suggest the following “JSEP does not
 specify a particular signaling model or signalling state machine. JSEP
 provides the means for an application to build a signaling state machine
 which complies with the requirements for offer/answers as specified in RFC
 3264".  This emphasises that JSEP is required to provide the tools to
 enable conformance to 3264 even if it allows some flexibility to diverge
 from 3264.

 6. Section 4.2. Session Descriptions and State Machine states:

 "Lastly, while the exact media parameters are only known only after a
 offer and an answer have been exchanged, it is possible for the offerer to
 receive media after they have sent an offer and before they have received
 an answer. To properly process incoming media in this case, the offerer's
 media handler must be aware of the details of the offerer before the
 answer arrives."

 I don't see the relevance of the statement "it is possible for the offerer
 to receive media after they have sent an offer and before they have
 received an answer". Why is the purpose of this statement?

 7. Section 4.2. Session Descriptions and State Machine states:

 "In [RFC3264], the constraints at the signaling level is that only one
 offer can be outstanding for a given session but from the media stack
 level, a new offer can be generated at any point.  For example, when using
 SIP for signaling, if one offer is sent, then cancelled using a  SIP
 CANCEL, another offer can be generated even though no answer was  received
 for the first offer. To support this, the JSEP media layer  can provide an
 offer whenever the Javascript application needs one for the signaling."

 Regarding the example SIP sequence in this scenario if the CANCEL is used
 to terminate a SIP dialog initiating INVITE I would assume that normally
 the peer connection would be terminated and a new peer connection used for
 any subsequent call so it does not seem to be a good example. If the
 example is meant to indicate that such a sequence could be used whilst
 establishing a SIP dialog it is a really horrible example. If it is
 intended to be an example mid dialog then the offer needs to be cancelled
 and the media state reverted to what it was previously.

 8. Section 4.2. Session Descriptions and State Machine.

 I think that some text is needed to describe how the version number in the
 SDP o line is managed and whether the application has responsibility for
 changing the version number if it changes the SDP between createOffer and
 setLocalDescription etc.


 9. Section 4.6. Session Rehydration states:

 "At this point a new offer can be generated by the new PeerConnection,
 with new ICE and SDES  credentials. This can then be used to re-initiate
 the session with the existing remote endpoint, who simply sees the new
 offer as an in-call renegotiation..."

 Whether this is possible depends on the signalling system used and the
 offer/answer state for example it might not be possible to send another
 offer if there is already an outstanding offer waiting for an answer so
 maybe the session would have to be terminated. This seems to suggest that
 Ekr's idea to somehow maintain the PeerConnection is more attractive.


 10. Section 5.1 SDP Requirements.

 I miss a statement related to the data channel. Mandatory/optional?
 [draft-ietf-rtcweb-data-channel]

 Bundling of media streams is also not mentioned here (bundle/mmt drafts)
 only is mentioned further down in section 6.

 DTLS-SRTP (RFC5763/64)? It's mentioned throughout the text, but not
 spelled out here. DTLS-SRTP would use as transport in the m-line
 DTLS/UDP/RTP/SAVP(F) as specified in RFC 5764.


 11. Section 5.2.1 Create Offer states:

 "In the event createOffer is called after the session is established,
 createOffer will generate an offer that is compatible with the current
 session, incorporating any changes that have been made to the session
 since the last complete offer-answer exchange, such as addition or removal
 of streams.  If no changes have been made, the offer will be identical to
 the current local description".

 What is meant by "compatible with the current session" in these sentence?
 If for example the offer included G.711 and OPUS audio codecs but the
 answer had only G.711 does this mean that any subsequent offer only has
 G.711 if so this I think is the wrong approach as the new offer should
 provide all the capabilities.  Also I assume that a new set of constraints
 could be used on the createOffer for example to change the media direction
 etc.


 12. Section 5.2.1 Create Offer states:

 "Calling this method may do things such as generate new ICE credentials,
 but does not change media state."

 So am I correct in thinking that this method MUST be called. If so given
 the statement in previous drafts that it does not have to be used it would
 be good to make an explicit statement that it must be used.


 13. Section 5.2.2 createAnswer.

 Presumably constraints can also be set on the answer in which case this
 should be stated like in the createOffer section.

 14. Section 5.2.3 SessionDescriptionType

 ""pranswer" indicates that a description should be parsed as an answer,
 but not a final answer, and so should not result in the freeing of
 allocated resources.  It may result in the start of media   transmission,
 if the answer does not specify an inactive media direction".

 Please note that "inactive" still results in the initiating the RTP
 Session and the transmission of RTCP so if you include RTCP as media this
 statement is not strictly correct.


 15. Section 5.2.3.1 creating Answers states:

 "When an endpoint receives an offer with these channels, it could send an
 answer accepting the data channel for two-way data, and accepting the
 audio and video tracks as receive-only".

 Whether “receive-only” can be used depends on the direction in the
 incoming offer and it should be noted that sending an answer with receive-
 only commits the local browser to sending RTCP so I am not sure this
 really works because the answer with receive-only still needs to contain a
 valid transport address.


 16. Section 5.2.4 setLocalDescription states:

 "If setRemoteDescription was previous called with an offer, and
 setLocalDescription is called with an answer (provisional or final), and
 the media directions are compatible this will result in the starting of
 media transmission."

 I assume by compatible that it is meant that the browser will enforce the
 RFC 3264 rules for setting the direction in an answer. If so please make
 this explicit if not what is meant by “compatible”?


 17. Section 5.2.5 setRemoteDescription - Same comment as above.


 Regards
 Andy

-- 
--------------------------------+--------------------------------------
 Reporter:  bernard_aboba@…     |      Owner:  draft-ietf-rtcweb-jsep@…
     Type:  defect              |     Status:  new
 Priority:  major               |  Milestone:  milestone1
Component:  jsep                |    Version:  1.0
 Severity:  Active WG Document  |   Keywords:
--------------------------------+--------------------------------------

Ticket URL: <http://trac.tools.ietf.org/wg/rtcweb/trac/ticket/3>
rtcweb <http://tools.ietf.org/rtcweb/>


From pkyzivat@alum.mit.edu  Wed Oct 31 18:58:35 2012
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Date: Wed, 31 Oct 2012 21:58:31 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Form of the video codec question
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On 10/29/12 2:51 PM, Stephan Wenger wrote:
> Hi Ted, chairs,
>
> I think these are the wrong questions, as the result of such a poll is not
> going to solve anything.  It's trivial to predict that there will be a
> significant show of hands in favor of each question.  What does that tell
> you?  Asking the questions as presented is going to be a waste of time.
>
> The more informative questions to as would be:
>
> 1) Do you object to H.264 as MTI, regardless of its technical and IPR
> merits (if any); and
> 2) Do you object to VP8 as MTI, regardless of its technical and IPR merits
> (if any)
> (both in more politically correct formulation, if you want.)
>
> If you chairs know that there are objections to both codecs by a
> significant number of people that cannot be swayed by technical or IPR
> arguments, a presentation of such technical and IPR merits can be omitted,
> and the WG can focus on stuff where progress is likely.

Stephan,

Maybe I am dense, but what arguments remain after you remove "technical" 
and "IPR"?

Presumably what is left is "business". But don't business considerations 
in this case devolve to either technical or IPR ones? (E.g. "I favor X 
because I have a license for it and some potential competitors do not" 
is ultimately an IPR question.)

	Thanks,
	Paul

> Regards,
> Stephan
>
>
> On 10.29.2012 11:08 , "Ted Hardie" <ted.ietf@gmail.com> wrote:
>
>> We got feedback after the audio codec discussion that folks would have
>> benefited from seeing the question we intended to ask before the
>> meeting.  In order to clarify that for this discussion, the chairs are
>> putting forward this draft of the question which will be asked in the
>> meeting (and then confirmed on the list).
>>
>> "1) If you support H.264 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> 2) If you support VP8 as the mandatory to implement codec or are
>> willing to live with it as the MTI, please raise your hand now.
>>
>> You may raise your hand more than once and we encourage you to do so
>> if you can live with either, even if you have a preference for one
>> over the other."
>>
>> Because all 3 chairs and our responsible AD could be perceived to have
>> conflicts of interest here, we have asked Robert Sparks to judge the
>> consensus of the room, and he has kindly agreed.  If you have major
>> heartburn about the form of the question, please let us know.
>>
>> regards,
>>
>> Ted Hardie, Cullen Jennings, Magnus Westerlund
>> _______________________________________________
>> 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
>

