[asterisk-bugs] [JIRA] (ASTERISK-26853) res_rtp_asterisk: Crash in pjnath when receiving packet

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26853?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236642#comment-236642 ] 

Richard Mudgett commented on ASTERISK-26853:
--------------------------------------------

"Voice cut" is so vague as to be meaningless.  Audio quality issues are unlikely to show up in the logs you provided.

* Does the voice stutter or sound choppy?  Do you have DEBUG_THREADS enabled?  DEBUG_THREADS will slow the systems performance so much that you could get a choppy or stuttering voice.  These patches add more locking to protect from reentrancy.
* Does the voice just stop even though the call is still connected?  Simply stopping could be a deadlock but you would also see channels hanging around after the calls are supposed to be gone.  With DEBUG_THREADS enabled you can get "core show locks" output and a backtrace [1].
* Does this happen to all calls?
* Are the calls in a conference (ConfBridge)?  You mentioned participants this time which implies a conference.
* I see in the logs that you are using both chan_sip and chan_pjsip on the same system.  There might be an interaction between these channel drivers.

Also the patch is currently going through the final automatic integration tests to merge.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

> res_rtp_asterisk: Crash in pjnath when receiving packet
> -------------------------------------------------------
>
>                 Key: ASTERISK-26853
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26853
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 13.14.0, 14.2.0
>         Environment: Debian jessie
>            Reporter: Studio ADAGIO
>            Assignee: Richard Mudgett
>         Attachments: backtrace_segfault_2.txt, backtrace_segfault.txt, backtrace-threads.txt, backtrace.txt, Config.tar.gz, core-show-locks.txt, debug_deadlock.zip, debug_segfault_2.txt, debug_segfault.txt, debug.txt, debug_voice_cut.txt, gdb.txt, messages_deadlock, messages.log, messages_segfault.txt, valgrind.txt, verbose_deadlock, verbose.log, verbose_segfault_2.txt, verbose_segfault.txt, verbose_voice_cut.txt
>
>
> Hi
> We have a business application that uses both conventional telephony and VoIP.
> We use the PJSIP library to make VoIP calls from mobile devices (Android & iOS). On server side we have Asterisk with PJSIP.
> Sometimes "Asterisk" process crash with "double free or corruption". This happens shortly after the INVITE transaction was finished (we hear about 0.5s of sound) and only if the call was started on Android device.
> We tried to reproduce the crash with other softphones (Zoiper, CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it doesn't crash when iOS app is used. So, it seems that, the problem is with our Android implementation, but we don't know where to search for the solution.
> We tried workarounds from here: ASTERISK-25274
> ASTERISK-25275
> But nothing worked.
> This crash occur once in about 200 calls.
> After using Valgrind (valgrind.org) to analyze Asterisk memory, we restart Asterisk and crash is happening more often. Is there a link ?
> You will find backtrace and debug in attachments.
> We tried Asterisk versions: 13.14 and 14.2
> PJSSIP versions: 2.5.5, 2.6
> (We tried to change audio codec but nothing changed)
> Thanks a lot



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