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

[Rusty Newton (JIRA)]

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

Rusty Newton updated ASTERISK-23171:
------------------------------------

    Assignee: Beppo mazzucato
      Status: Waiting for Feedback  (was: Triage)

Sounds like you see the issue a few times a day. Can you provide the following?

1. Debug log capture up to the crash with all the options shown here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

2. Configuration files for the SIP peers and dialplan involved

3. Backtraces, following instruction here https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace  (with BETTER_BACKTRACES and DONT_OPTIMIZE)



                
> 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
>
>
> 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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