asterisk users - Oct 2012 - DTMF inband with telephone-event in SDP

Hello everyone!

We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).

Now we noticed there are some SIP clients that announce telephone-event
in their SDP, but send their DTMF inband. The problem with that is, that
Asterisk obviously does not try to detect inband DTMF after seeing the
telephone-event payload type in the SDP.

So we are in a kind of dilemma:
- dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for
the described ones.
- dtmfmode=inband would also work for most, but of course not for the
ones using g726 et al.

Is there any Asterisk setting to force inband DTMF detection (with
non-compressing codecs only, of course)? I browsed the code without result.

Does anybody have a hint how to handle this? Or if the SIP clients
behaviour is even RFC compliant?


Regards and TIA,
Jakob

Jakob Hirsch wrote:
> Hello everyone!
Hola,
> We use Asterisk for various services like voicemail. Our SIP clients
> usually use rtp events (rfc2833) for DTMF, which works just fine and
> independent from the codec (g711 vs. g726 etc.).
>
> Now we noticed there are some SIP clients that announce telephone-event
> in their SDP, but send their DTMF inband. The problem with that is, that
> Asterisk obviously does not try to detect inband DTMF after seeing the
> telephone-event payload type in the SDP.
Generally DTMF is something that has to be configured on both sides, you 
can't just configure it on one and have the negotiation force it to be that.
> So we are in a kind of dilemma:
> - dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for
> the described ones.
> - dtmfmode=inband would also work for most, but of course not for the
> ones using g726 et al.
>
> Is there any Asterisk setting to force inband DTMF detection (with
> non-compressing codecs only, of course)? I browsed the code without result.
Unfortunately there isn't a way to force this as you describe out of the 
box, you would have to make changes to chan_sip or explicitly have the 
clients configured properly.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org