[asterisk-dev] [Code Review] 3562: chan_sip: Start session timer at 200, not at INVITE.

[wdoekes]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3562/
-----------------------------------------------------------

Review request for Asterisk Developers.


Bugs: ASTERISK-22551
    https://issues.asterisk.org/jira/browse/ASTERISK-22551


Repository: Asterisk


Description
-------

RFC says Asterisk should start counting session timers at 200. Asterisk starts at INVITE.

For short intervals (e.g. 90 secs) and long ringing times (e.g. 30 secs), this means that
a caller with refresher=uac will get disconnected by Asterisk before it has a chance to
send a refreshing reINVITE.

Reproduced using:
https://github.com/ossobv/sipp-scenarios/blob/master/INVITE-test-session-refresher-uac.xml


Diffs
-----

  /branches/1.8/channels/chan_sip.c 414344 

Diff: https://reviewboard.asterisk.org/r/3562/diff/


Testing
-------

Tested against the scenario mentioned above.

Before the patch, Asterisk aborts the call during <pause milliseconds="45000"/>
After the patch, the scenario completes succesfully.

The scenario tests both:
- asterisk not killing the dialog too early, and
- killing it when expected

After the scenario and a reasonable time, no excess 'sip show objects' were seen.


Thanks,

wdoekes

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140523/614bdf51/attachment.html>