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

Roman Shpount <roman@telurix.com> Mon, 07 March 2016 18:35 UTC

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Date: Mon, 07 Mar 2016 13:35:52 -0500
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From: Roman Shpount <roman@telurix.com>
To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard
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On Mon, Mar 7, 2016 at 5:47 AM, Tim Panton <tim@phonefromhere.com> wrote:

> Actually they are all implementable with webRTC - if combined with
> webAudio.
> You just generate the DTMF tones with webAudio, mix (as appropriate)  with
> the mic signal and then
> inject the result into the outgoing using
> AudioContext.createMediaStreamDestination()
> and friends.
>
> We could add a caveat that all the DTMF apis are optional if the browser
> also implements
> webAudio. As it is possible to write a a javascript polyfill that
> implements the current api in
> terms of webaudio, which should be adequate for the legacy compatibility
> needs.
>
> Also we should add a note that _all_ these sorts of usages would be better
> served by
> carrying the data over the DataChannel, DTMF is for legacy interop only.
>
>
This would be really unwise. DTMF generated as inband media is not
delivered by all codecs. Inband DTMF has much higher error rate then RFC
4733 in case of packet loss. Not supporting RFC 4733 will negatively affect
interop.

P.S. We already spent more time discussing DTMF then it would take to
implement it.
_____________
Roman Shpount

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