[asterisk-dev] [Code Review] 3431: Fix channel staging assertion failure.
Matt Jordan
reviewboard at asterisk.org
Mon Apr 14 14:32:35 CDT 2014
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3431/#review11601
-----------------------------------------------------------
Ship it!
Please address the rtp_engine documentation finding before committing.
- Matt Jordan
On April 9, 2014, 2:19 p.m., rmudgett wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3431/
> -----------------------------------------------------------
>
> (Updated April 9, 2014, 2:19 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> The failing assertion ensures that the final snapshot gets generated so CDR records can get finalized. The only place where a channel staging snapshot flag could be left set is in the handle_request_bye(). The function could return before clearing the flag because the channel could dissappear while the function had to have the channel unlocked.
>
> * Fixed handle_request_bye() channel snapshot staging coverage area to not have a return in the middle of it and be unable to clear the staging flag.
>
> * Pushed the channel snapshot staging coverage area into ast_rtp_instance_set_stats_vars() to ensure that the staging is not interrutped.
>
> * Made callers of ast_rtp_instance_set_stats_vars() not call it with channels or channel driver private locks held to eliminate the deadlock potential. The callers must hold references to the passed in channel and rtp objects.
>
> * Eliminated sip_hangup() trying to get the bridge peer. It is futile at this point because the channel could never be in a bridge.
>
> * Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end of the function. The unref needs to happen after the last use of the pointer.
>
>
> Diffs
> -----
>
> /branches/12/main/rtp_engine.c 412047
> /branches/12/channels/chan_sip.c 412047
>
> Diff: https://reviewboard.asterisk.org/r/3431/diff/
>
>
> Testing
> -------
>
> I was unsuccessful in reproducing the testsuite channel staging assertion failure.
> However, SIP calls can still setup and teardown with the patch installed.
>
>
> Thanks,
>
> rmudgett
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140414/46919ed9/attachment-0001.html>
More information about the asterisk-dev
mailing list