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Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

"Asveren, Tolga" <tasveren@sonusnet.com> Tue, 01 March 2016 18:40 UTC

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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
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Date: Tue, 01 Mar 2016 18:40:08 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
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Negotiated codec may reveal information about whether transcoding will happen (and it may be known a priori that all transcoded scenarios use a certain codec on the “legacy” side” etc…) or not depending on application/deployment model, e.g. AMR negotiated => no transcoding. It really depends on the use case/deployment model. I am not arguing this would be a generic/universal capability but may play a role. And if not, again nothing is really changing in my main argument: App. Developer can either supply the values or use the browser defaults.

Thanks,
Tolga

From: Roman Shpount [mailto:roman@telurix.com]
Sent: Tuesday, March 01, 2016 1:27 PM
To: Asveren, Tolga <tasveren@sonusnet.com>
Cc: Ted Hardie <ted.ietf@gmail.com>; Harald Alvestrand <harald@alvestrand.no>; rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

On Tue, Mar 1, 2016 at 3:04 AM, Asveren, Tolga <tasveren@sonusnet.com<mailto:tasveren@sonusnet.com>> wrote:
App. Developer may use different values based on the negotiated codec.


Can you please explain how negotiated codec will affect DTMF tone generation? WebRTC end points will only send DTMF tones using RFC 4733 telephone events, so codec negotiation will not affect DTMF tone duration.
_____________
Roman Shpount

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