[asterisk-dev] [Code Review] 3066: bridge_native_rtp: Deadlock during 4-way conference creation
Mark Michelson
reviewboard at asterisk.org
Wed Dec 11 18:32:05 CST 2013
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3066/#review10388
-----------------------------------------------------------
Ship it!
Heh, every time I thought I spotted something wrong, it turned out I was wrong instead :)
The only suggestion I have for this is that since RTP glue update_peer() method is now consistently called with the channel locked, I would update its documentation in rtp_engine.h to note this.
- Mark Michelson
On Dec. 11, 2013, 11:20 p.m., Kevin Harwell wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3066/
> -----------------------------------------------------------
>
> (Updated Dec. 11, 2013, 11:20 p.m.)
>
>
> Review request for Asterisk Developers and kmoore.
>
>
> Bugs: ASTERISK-22749
> https://issues.asterisk.org/jira/browse/ASTERISK-22749
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> This contains the patch on the issue (submitted by kmoore), as well as the fix for the adding in a bridge lock while calling bridge_start from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. The addition to the given patch adds in the necessary bridge locking in order to avoid a deadlock.
>
>
> Diffs
> -----
>
> branches/12/main/channel.c 403687
> branches/12/include/asterisk/channel.h 403687
> branches/12/channels/chan_sip.c 403687
> branches/12/channels/chan_pjsip.c 403687
> branches/12/bridges/bridge_native_rtp.c 403687
>
> Diff: https://reviewboard.asterisk.org/r/3066/diff/
>
>
> Testing
> -------
>
> Added channels via DTMF attended transfer to get to a 4-way bridge and then removed them and noted all channels and the bridge had been torn down.
>
>
> Thanks,
>
> Kevin Harwell
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131212/d4c08cf6/attachment-0001.html>
More information about the asterisk-dev
mailing list