[asterisk-bugs] [JIRA] (ASTERISK-26853) Asterisk crash with pjsip library

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26853?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235709#comment-235709 ] 

Joshua Colp commented on ASTERISK-26853:
----------------------------------------

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> Asterisk crash with pjsip library
> ---------------------------------
>
>                 Key: ASTERISK-26853
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26853
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.14.0, 14.2.0
>         Environment: Debian jessie
>            Reporter: Studio ADAGIO
>         Attachments: gdb.txt, verbose.log
>
>
> 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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