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

<stephane.proust@orange.com> Fri, 11 January 2013 12:33 UTC

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From: stephane.proust@orange.com
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Call for Consensus Regarding Selecting Recommended Audio Codecs
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Date: Fri, 11 Jan 2013 12:33:02 +0000
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Subject: Re: [rtcweb] Call for Consensus Regarding Selecting Recommended Audio Codecs
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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 : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part de Magnus Westerlund
Envoyé : jeudi 20 décembre 2012 09:30
À : rtcweb@ietf.org
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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