[Rtcomm] [Rtcomm] [Bug 3652] DTMF doesn't work after SIP call is initiated

[From: bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=3652





------- Comment #6 from olivier.crete at collabora.co.uk  2008-09-04 01:34 GMT+3 -------
(In reply to comment #5)
> Please see this link at ITT:
> http://www.internettablettalk.com/forums/showthread.php?p=220623#post220623

This part is full of inaccuracies:

First, we always pick the codec preferred by the remote provider (on every
call, it sends a list of codec in its order the preference).

Second, if the voice codec is a high-compression codec like G.729 or iLBC, we
will send DTMF in low-compression codecs like PCMA or PCMU if the other party
accept those codecs.

Third, there are actually three methods in which DTMF can be sent with a
SIP/RTP call.
 - As sound in the call... we support that fully (including downgrading the
codec while sending DTMF to use a codec that wont mangle it) 
 - As RTP events (RFC 2833/4733), we also fully support that.. this is
sometimes called in-band and sometimes out-of-band
 - as SIP INFO (RFC 2976), this is not supported for the moment.


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