Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt

As I said last time this came up, we need to include language that ensures that these non-mandatory codecs don't 
get used unless the mandatory ones are unavailable, or the web developer explicitly selected them.

I don't want to find that I'm getting amr-nb between a pair of phone based browsers and getting narrow band because
both the phones happen to have that codec available.

How about: 

To ensure a baseline level of interoperability between WebRTC clients, 
a minimum set of required codecs are specified below.
If other suitable audio codecs are available to the browser to use, it 
is advised that they are also included in the offer at a lower priority than the required codecs in order to 
maximize the possibility to establish the session without the need for 
audio transcoding when the required codecs are unavailable to one party.

- Of course this whole discussion highlights the need for a set of constraints (or an API) that let the web developer 
express her preferences wrt codec selection.

Tim.

On 11 Sep 2013, at 10:19, <stephane.proust@orange.com> <stephane.proust@orange.com> wrote:

>> These are the codecs that would be recommended under such text. None of them is really ideal for interactive audio
> 
> That is the reason why the proposed text only applies to "suitable" audio codecs
> " If other suitable audio codecs..."
> 
> That's also why it cannot be interpreted as a strictly normative text and why I also suggested (as it was previously suggested by the Chairs) to have some additional informative (and separate) text about what are those suitable codecs to be considered and what is the rationale behind this (http://www.ietf.org/mail-archive/web/rtcweb/current/msg08501.html)
> 
> Stéphane
> 
> 
> -----Message d'origine-----
> De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part de Jean-Marc Valin
> Envoyé : mardi 10 septembre 2013 18:27
> À : Mo Zanaty (mzanaty)
> Cc : rtcweb@ietf.org
> Objet : Re: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-02.txt
> 
>> To ensure a baseline level of interoperability between WebRTC clients, 
>> a minimum set of required codecs are specified below.
>> If other suitable audio codecs are available to the browser to use, it 
>> is RECOMMENDED that they are also included in the offer in order to 
>> maximize the possibility to establish the session without the need for 
>> audio transcoding.
> 
> Let's look at the practical effect here, assuming people were to actually follow such a recommendation. The only audio codecs that are implemented in browsers but not available for WebRTC are the ones used for HTML5. According to http://html5test.com/ these codecs are Vorbis, MP3, AAC, and uncompressed PCM. These are the codecs that would be recommended under such text. None of them is really ideal for interactive audio, none of them is implemented in old phones AFAIK, but all of them would require time and effort to add to a browser's RTP stack. I really don't see the point here.
> 
> 	Jean-Marc
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