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

[Studio ADAGIO (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26853?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Studio ADAGIO updated ASTERISK-26853:
-------------------------------------

    Attachment: valgrind.txt
                backtrace.txt

> 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: Unassigned
>         Attachments: backtrace.txt, Config.tar.gz, debug.txt, gdb.txt, messages.log, valgrind.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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