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Re: [Bug 18485] Change DTMF API to be on PeerConnection

From: Roman Shpount <roman@telurix.com>
Date: Wed, 8 Aug 2012 10:33:36 -0400
Message-ID: <CAD5OKxv3uzPhztYccR+RYp3BAor4THJKEDEJ1SGjNh2r_6ECGA@mail.gmail.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
Cc: public-webrtc@w3.org
On Wed, Aug 8, 2012 at 10:20 AM, Stefan Hakansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 08/08/2012 04:00 PM, Randell Jesup wrote:
>> This causes problems for speakerphone situations: in many/most
>> implementations (including those based on the webrtc.org code),
>> <audio>/<video> elements not part of the core webrtc logic may not be
>> fed into the echo canceller.  This means the tone would echo
>> uncontrolled into the microphone and to the far end, but distorted and
>> out of phase with local generation of DTMF (or for IVR systems, possibly
>> cause confusion, though probably not).
> If this is a real issue (I guess it depends on the implementation in the
> browser and the underlying system) then we should avoid A. But that would
> also mean that no other sounds could be produced while using webrtc.
> Imagine that you get an email (in your web client) or chat message and you
> have audio notifications enabled, those shouldn't echo to the far end (or
> should they?).

You should suppress sending audio during the time RFC 4733 is being sent.
This causes interop problems anyway and would prevent potential echo
problems. I think this is the IETF group recommendation anyway.
Roman Shpount
Received on Wednesday, 8 August 2012 14:34:13 UTC

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