Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

<stephane.proust@orange.com> Wed, 13 March 2013 19:37 UTC

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From: <stephane.proust@orange.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>, MARJOU Xavier OLNC/OLN <xavier.marjou@orange.com>
Thread-Topic: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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Date: Wed, 13 Mar 2013 19:36:52 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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Hello

Our understanding is that the reason of the "no consensus" on additional recommended codecs was the additional costs for browsers.
The proposal is then to make these "MUST" fully conditional to the case of no (or very reduced) additional costs, when the codecs are already available on the device and when no additional license fee is required

We could even live with lower level of "requirements" with respectively May and Should (instead of Should and shall) but we think that this proposal is a way to take into account both browser manufacturers concerns on browsers costs and telcos concerns on transcoding costs and some other companies share this view.

Stéphane	






-----Message d'origine-----
De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part de Jean-Marc Valin
Envoyé : mercredi 13 mars 2013 20:24
À : MARJOU Xavier OLNC/OLN
Cc : rtcweb@ietf.org
Objet : Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

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Hi,

I'd really like to understand how the chairs coming to the conclusion that there was *no consensus* on recommended codecs can result in a draft that includes 3 MUSTs and 1 SHOULD. This draft effectively makes
3 new codecs MTI for a range of devices. I understand that it's an individual draft and you can write whatever you like in there, but it definitely goes against the result of the WG discussion.

Cheers,

	Jean-Marc

On 03/13/2013 09:14 AM, Xavier Marjou wrote:
> Here is a summary of the
> draft-marjou-rtcweb-audio-codecs-for-interop-00 presentation that I 
> had prepared for yesterday's session:
> 
> - The co-authors want to underline that non-WebRTC voice endpoints 
> usually use one of the following codecs: AMR, AMR-WB or G.722, which 
> will result in massive transcoding when there will be communications 
> between WebRTC endpoints and non-WebRTC endpoints.
> 
> - On one side, transcoding is bad for many reasons discussed in the 
> draft (cost issues, intrinsic quality degradation, degraded 
> interactivity, fallback from HD to G.711...);
> 
> - On the other side, it is recognized that implementing additional 
> codecs in the browsers can generate additional costs.
> 
> - In order to reach a compromise, we would like to add some text in 
> the WG draft draft-ietf-rtcweb-audio providing incentives for the 
> browser to use these three codecs: make them mandatory to implement 
> when there is no cost impact on the browser (e.g. if codec already 
> installed, paid by the device vendor...).
> 
> Any opinion on that?
> 
> Cheers,
> 
> Xavier
> 
> PS: I will be ready to present the slides on Thursday if time permits 
> it.
> 
> (c.f. http://tools.ietf.org/agenda/86/slides/slides-86-rtcweb-6.pdf
> )
> 
> 
> 
> On Thu, Mar 7, 2013 at 11:18 AM, Ted Hardie <ted.ietf@gmail.com 
> <mailto:ted.ietf@gmail.com>> wrote:
> 
> Magnus and I discussed this this morning, and we encourage you to 
> prepare something.  If the discussion of working group last call items 
> runs short, we may be able to fit this in at that time or at the end 
> of day one if its full agenda his finished.  This is not a commitment, 
> however, so please try and get discussion on the list on the points 
> from the draft you feel need resolution.
> 
> regards,
> 
> Ted
> 
> 
> On Mon, Mar 4, 2013 at 2:31 PM, Espen Berger (espeberg) 
> <espeberg@cisco.com <mailto:espeberg@cisco.com>> wrote:
>> Hello,
>> 
>> 
>> 
>> I would like to request agenda time for:
>> 
>> 
>> 
>> draft-marjou-rtcweb-audio-codecs-for-interop-01
>> 
>> 
>> 
>> The document  presents use-cases underlining why WebRTC needs
> AMR-WB,  AMR
>> and G.722 as additional relevant voice codecs to satisfactorily 
>> ensure interoperability with existing systems.
>> 
>> 
>> 
>> A 10-minute time slot should be sufficient for presentation and
> discussion.
>> 
>> 
>> 
>> Regards
>> 
>> 
>> 
>> -Espen
>> 
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________ rtcweb mailing list 
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org> 
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org <mailto:rtcweb@ietf.org> 
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 
> 
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 

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