Re: [rtcweb] Call for Consensus Regarding Selecting Recommended Audio Codecs

[<[e-mail]>]

Hello all

Considering all the discussions on the list, it is clear that a way must be found to address from codec perspective both the interoperability issue and the license cost issues with a good compromise spirit

Ensuring interoperability of Web RTC technology across various types of environment and devices (e.g. fixed/PC, mobile, tablets..) is a must.

Strictly sticking to the current situation with only OPUS and G.711 will not solve properly the issue especially because this would mean to fail to address correctly the biggest market of hundreds of millions of mobiles terminals that currently support neither G.711 nor OPUS. 
 
The lack of support & use of AMR and AMR-WB codecs  in Web-RTC will then require transcoding and degrade the user experience: lower end to end voice quality and increased latency. The lack of support of these codecs would also increase the costs on network side to implement codec transcoding functions.
 
Hence, it is needed to also consider the support & use of these additional codecs (AMR, AMR-WB) by Web-RTC.
 
It is recognized however that the support of these additional codecs by WebRTC would add some costs on browsers, part of these costs coming from codec license fees. However, from technical perspective, browser implementation of these codecs is not needed, when these codecs are already natively implemented by the hardware platform (e.g. by mobile phone).
 
Considering this very reduced set of additional codecs which has to be taken into account and considering, in addition, that these codecs are already supported by many devices, a way forward can surely be found like by ensuring that RTC Web give access to these codecs at reasonable costs whenever possible. And this could be especially the case when these codecs are already implemented on the devices.
 
Such way forward would require however more than just e-mail discussion between different options. We think that some more detailed inputs for next meeting would be needed to address properly this interoperability topic by considering priority use cases and compromise ways forward like proposed in this e-mail.

We intent to provide a draft on this issue at next meeting.

-----Message d'origine-----
De : PROUST Stephane OLNC/OLPS 
Envoyé : vendredi 11 janvier 2013 13:33
À : 'Magnus Westerlund'; [e-mail]
Objet : RE: [rtcweb] Call for Consensus Regarding Selecting Recommended Audio Codecs

It is clear from the discussions on the rtcweb e-mail reflector that the only additional codecs to be considered are limited to the following very small subset of 3 codecs: AMR, AMR-WB and G.722. 
In addition to G.711, these 3 codecs cover almost all legacy devices dedicated to voice services. They are consequently needed and sufficient to be supported by WebRTC to make it an attractive and future proof technology for usage in all environments including mobile and for interoperability use cases with most of all legacy voice terminals.

AMR and AMR-WB are indeed the most widely supported voice codecs in hundreds of millions of legacy mobile devices.
G.722 is royalty free (including a Packet Loss Concealment solution provided in ITU-T Software Tool Library) and is the codec used for HD Voice / DECT-Cat IQ fixed devices

Furthermore, considering that the reason for excluding AMR and AMR-WB from WebRTC was the licensing issue, there is no reason to NOT support AMR-WB and AMR at WebRTC level if these codecs are already implemented on the device.

Therefore I support option 1 and propose the following specification according this:
AMR-WB, AMR and G.722 are RECOMMENDED TO IMPLEMENT by WebRTC end-points
AMR and AMR-WB MUST BE supported at WebRTC level by WebRTC end-points on 3GPP mobile devices already implementing AMR and AMR-WB (*)

(*) note that the way these codecs are supported at RTC Web level is left open to implementors: either by a WebRTC specific software implementation of these codecs or by using APIs to access hardward implementation. 


-----Message d'origine-----
De : [e-mail] [mailto:[e-mail]] De la part de Magnus Westerlund
Envoyé : jeudi 20 décembre 2012 09:30
À : [e-mail]
Objet : [rtcweb] Call for Consensus Regarding Selecting Recommended Audio Codecs

WG,

As an outcome of the Vancouver IETF meeting codec discussions we did
promise to run a call for consensus regarding if the WG was interested
in specifying a small set of recommended audio codecs. We are sorry this
has been delayed until now.

The question for the call of consensus is between two options.

1) Run a process in the WG to select and specify a small set of
audio/speech codecs that would be RECOMMNEDED to implement by a WebRTC
end-points

2) Do nothing and let the already specified Mandatory to Implement Audio
codecs be the only audio codecs mentioned in the WebRTC specification.

Please indicate your position by January 16th 2013.

Regards

Magnus Westerlund

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