[asterisk-dev] Update: Asterisk DTMF issues with Sonus

[Kristian Kielhofner]

Hello everyone,

  It seems we have finally found the cause of all of our DTMF issues
with Sonus equipment.

  When Asterisk is bridging two SIP channels with RFC 2833 DTMF it
appears to wait for the end of the event from the transmitting device
before re-transmitting the event to the other (bridged) channel.
While Asterisk is waiting for the event to end it isn't transmitting
RTP (or anything else) to the other device.  This can be seen as a gap
in RTP traffic to the bridged channel in between the last voice frame
and the start of the DTMF event.

  Sonus equipment appears to go haywire (that's the technical term) if
this gap in RTP traffic exceeds 100ms (although in practice the
"limit" seems to be much lower).  Don't ask me why, that's just what I
was told.  So as best as I can tell if the RFC 2833 event lasts longer
than 100ms (or anywhere near there - even 80ms) there isn't any RTP
traffic from Asterisk (voice or 2833 events) and RFC 2833 DTMF with
Sonus will never work.

  I know there are many people that claim to be using Asterisk 1.4
directly with Sonus equipment.  How, I don't know.  I'm running a
very, very vanilla Asterisk configuration and I've been battling large
carriers and Sonus on DTMF for over a month...

  It seems the best way to solve this is to have Asterisk forward 2833
events as it receives them instead of waiting till the end of the
event (or am I completely wrong about all of this)?

  Any other ideas?

  Thanks!

  PS - It's very clear (at this point) we are going to require some
modifications to RTP/RFC 2833 handling in Asterisk.  We are willing to
pay for these modifications and contribute the work (if appropriate)
back to Asterisk/Digium.  If you are interested, please e-mail me off
list.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com