[asterisk-dev] Request for Comments: RTP out-of-band dtmf signalling for connections between asterisk and cisco equipment

Andreas Brodmann andreas.brodmann at gmail.com
Wed Dec 19 02:58:05 CST 2007


The following issue arrose out of the bug report at
http://bugs.digium.com/view.php?id=11489.

I was asked by Eliel to post my thoughts to the developer list:

Beginning with asterisk-1.4.15 a bug concerning rfc2833 oob dtmf signalling
on sip trunks was closed.
Unfortunately this fix opened a new issue:

Cisco carrier components (bt) use x-nse in the sdp message rather than the
telephone-event (as defined in rfc2833) - although x-nse was meant to be a
complement.
Concerning the interoperability between asterisk and those components this
means
that rtp out-of-band signalling does not work anymore.

I wrote a patch (for chan_sip) that, when dtmfmode=rfc2833 was set in the
sip.conf, asterisk would use
rfc2833 even if the other side was not signalling this capability (resp.
only signalled x-nse).

Eliel didn't feel comfortable about this as it is more of a work-around than
a propper fix.

My question is:
Which way of making the two systems interoperable at this level would
be fine to digium/the developer community (and would be most likely to be
accepted into main stream code base):

1) modifying the code so that the existance of 'x-nse' in the sdp body is
accepted as if
'telephone-event' was there
2) adding a config switch to sip.conf that would allow the admin to force
using rfc2833 on
a sip trunk even if the other side doesn't signal rfc2833 capability

Thanks for you comments (and if I overlooked the fact that someone has
already
solved this, please let me know).

-Andreas
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