[asterisk-dev] [Code Review] Lower level SIP serialization
Joshua Colp
reviewboard at asterisk.org
Sat Mar 2 12:31:52 CST 2013
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2362/
-----------------------------------------------------------
Review request for Asterisk Developers and Mark Michelson.
Summary
-------
While writing testsuite tests for the new SIP work I came across some weird crash issues. After examining them I determined that the issue was pjsip code continuing to modify state after we queue tasks, potentially even deallocating some resources we need. The attached change moves serialization to a much lower level so that serialization occurs before invoking any pjsip modules. This means that for most cases we no longer need to do serialization ourselves in pjsip callbacks. This can be set on a per-dialog basis so it is not limited to just the session work.
Diffs
-----
/team/group/pimp_my_sip/include/asterisk/res_sip.h 382201
/team/group/pimp_my_sip/res/res_sip.exports.in 382201
/team/group/pimp_my_sip/res/res_sip/sip_distributor.c 382201
/team/group/pimp_my_sip/res/res_sip_session.c 382201
Diff: https://reviewboard.asterisk.org/r/2362/diff
Testing
-------
Ran normal calls through to confirm functionality works as it previously did. Ran against testsuite tests manually under heavy cps with obscure scenarios to confirm no issues.
Thanks,
Joshua
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130302/216fbc37/attachment.htm>
More information about the asterisk-dev
mailing list