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Re: Where to attach a DTMF API

From: Neil Stratford <nstratford@voxeo.com>
Date: Tue, 29 Nov 2011 09:41:08 +0000
Message-ID: <4ED4A8B4.9050305@voxeo.com>
To: public-webrtc@w3.org
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?

Received on Tuesday, 29 November 2011 09:41:48 UTC

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