[asterisk-bugs] [JIRA] (ASTERISK-23171) Crash in res_rtp_asterisk on WebRTC incoming call

[Beppo mazzucato (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23171?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Beppo mazzucato updated ASTERISK-23171:
---------------------------------------

    Status: Waiting for Feedback  (was: Waiting for Feedback)

I'm attaching for the two crashes the portion of the logfile with the call causing the crash and the relevant portion of dialplan and sip.conf.

Recompiling with BETTER_BACKTRACES and DONT_OPTIMIZE produce an error in Chrome (Called with an SDP without ice-ufrag and ice-pwd) don't allowing to make calls and reproduce the crash.

I'll enter a separate issue for this
                
> Crash in res_rtp_asterisk on WebRTC incoming call 
> --------------------------------------------------
>
>                 Key: ASTERISK-23171
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23171
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.7.0
>         Environment: CentOS 6.5 64 bit
> Google Chrome Version 32.0.1700.77
> jssip 0.3.0
>            Reporter: Beppo mazzucato
>            Assignee: Beppo mazzucato
>         Attachments: backtrace2.txt, backtrace.txt, log-dialplan-sip2.txt, log-dialplan-sip.txt
>
>
> Asterisk crash sometimes on WebRTC incoming calls. The frequency depends from the number of concurrent user.
> Never seen with 1-2 users it happen 3-4 times a day with 10 users.
> I'm attaching the backtrace of the last two incidents (unfortunately optimized)

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