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

"Olle E. Johansson" <oej@edvina.net> Wed, 13 March 2013 14:38 UTC

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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Wed, 13 Mar 2013 15:37:58 +0100
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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13 mar 2013 kl. 15:05 skrev Harald Alvestrand <harald@alvestrand.no>:

>> - 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.
>> 
I have a large amount of SIP phones on my desk, none which support AMR and/or AMR-WB. I had to pay extra for AMR in a SIP phone for my iPhone.

I don't agree with the standpoint that AMR is implemented on a majority of traditional non-WebRTC endpoints.

/O

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