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

Xavier Marjou <xavier.marjou@gmail.com> Wed, 13 March 2013 14:43 UTC

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Date: Wed, 13 Mar 2013 10:43:34 -0400
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From: Xavier Marjou <xavier.marjou@gmail.com>
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Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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---------- Forwarded message ----------
From: Xavier Marjou <xavier.marjou@gmail.com>
Date: Wed, Mar 13, 2013 at 10:43 AM
Subject: Re: [rtcweb] Agenda time request for
draft-marjou-rtcweb-audio-codecs-for-interop-01
To: Harald Alvestrand <harald@alvestrand.no>



On Wed, Mar 13, 2013 at 10:05 AM, Harald Alvestrand <harald@alvestrand.no>wrote;wrote:

>  High level thoughts from a browser implementor:
>
> - The first target of WebRTC is the browser to browser case. These codecs
> do not add any capability to the browser to browser case, because OPUS is
> available.
>

I agree with that browser to browser communication within a given calling
service is the first target of most players. However, there's the need to
have options for interworking with "legacy" VoIP equipments for players who
also want to open WebRTC communications towards "legacy" devices. Our draft
shows that OPUS and G.711 are really not sufficient for that.


> - In the most common browser distribution models, it is an advantage to
> include at least a fallback version of all available features in the
> binary, so that the set of features available to the user is constant. This
> means that the browser either incurs licensing costs or support costs (to
> support the variability of user scenarios).
>
> - The inclusion of royalty-required codecs severely crimps the
> distribution models available for browsers, which makes it even harder for
> new browsers to enter the market than it already is, and the WG charter we
> agreed on says that we "prefer non-encumbered codecs".
>

Yes, we understand that, and we think the text we propose tries to motivate
all players of the WebRTC ecosystem to open their technologies. I think our
text is really not such a big constraint as I read it as a conditional
statement like "if some conditions are met regarding AMR, or AMR-WB or
G.722, then the browser offers the possibility to use this codec". All
conditions may not be met today for the browser implementor, perhaps only
tomorrow, but at least writing down will provide incentives for all players
to make efforts and go in that direction.


>
> We can discuss the cost of transcoding separately, but the cost to the
> browsers of extra codecs, especially royalty-encumbered ones, should not be
> underestimated.
>
>
> On 03/13/2013 02:14 PM, 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> 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> 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
>> >
>> >
>> >
>> >
>> >
>> >
>>   > _______________________________________________
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>> >
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>
>
>
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