[asterisk-dev] SIP sessions timers - please test!

[Johansson Olle E]

Friends,

Thanks to a group of sponsors and excellent work by Ray Jain, we now  
have a branch of Asterisk
that supports SIP Session Timers.

SIP session timers is an extension to SIP that makes sure a session is  
still alive by sending
out "keep-alive" messages now and then. This is a good addition to the  
RTP timeout timers
we already have. The RTP timers make sure we hangup a call when RTP  
media that was
supposed to flow through Asterisk disappears.

If the RTP media is re-invited away from Asterisk, the RTP timeout  
doesn't help much and
the SIP Session Timers kicks in. If the SIP signalling channel now  
times out, we will kill
the call and close the CDR. This makes sure that no SIP channels hang  
forever when
a device is killed or crashes.

Please test this. You need phones or servers that support SIP session  
timers.

Download the branch
  http://svn.digium.com/view/asterisk/team/group/sip_session_timers

Check the bug report (where you leave your test report)
http://bugs.digium.com/view.php?id=10665

And don't miss the good documentation in the bug report - the attached  
PDF!

A big thank you to all involved in this work! Let's move this forward  
quickly.

/Olle