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Re: Where to attach a DTMF API

From: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Date: Tue, 29 Nov 2011 10:48:37 +0100
Message-ID: <4ED4AA75.5000205@ericsson.com>
To: public-webrtc@w3.org
On 11/29/2011 10:41 AM, Neil Stratford wrote:
> On 29/11/2011 08:48, Stefan Håkansson LK wrote:
>> I the mail referenced, sendDTMF is a method on MediaStreamTrack. I
>> think the method should apply on PeerConnection because my
>> understanding is that the idea is to generate RTP-packets according to
>> RFC4733, not to insert tones in the audio. This means that "sendDTMF"
>> has really no meaning outside a PeerConnection.
>> I understand that this means that there are some other things that has
>> to be met:
>> * There must be an audio MediaStreamTrack in at least one of the
>> localStream's (that the DTMF RTP packets can share SSRC with)
>> * If there are several outgoing audio RTP streams (having different
>> SSRC's), it must be possible to understand (control?) which SSRC that
>> will be reused by DTMF.
>> My very simple proposal for this would be that the DTMF RTP packets
>> will share SSRC with the first audio track of the first MediaStream
>> that has at least one audio track. If there is no such MediaStream in
>> localStream's, then "sendDTMF" will fail.
> It is important that it is possible to send DTMF without any request for
> microphone access if the call is purely to an informational IVR where
> the caller is never expected to speak, but still needs to navigate that
> IVR. Similarly there are cases where DTMF may be required but the call
> is video only, with no audio component.
> How should we handle these cases? Can we create a null audio track using
> the current API?
How would you do it for a legacy client? My understanding was that 
RFC4733 required and existing (audio) RTP stream (identified by SSRC) to 
insert its DTMF payloads into.
> Neil
Received on Tuesday, 29 November 2011 09:49:16 UTC

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