[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:comment-tabpanel&focusedCommentId=215028#comment-215028 ] 

Rusty Newton commented on ASTERISK-23171:
-----------------------------------------

I see you are using the "opus" codec, which is not supported in Asterisk 11.7.0. Are you using a patched Asterisk 11.7.0, if so what all patches are you using?   Otherwise are you using Asterisk 12 and what version or SVN revision?
                
> 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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