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

[Matt Jordan (JIRA)]

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

Matt Jordan closed ASTERISK-23171.
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    Resolution: Duplicate

I'm closing this out as a duplicate. Keeping the issues linked should allow them to be resolved at the same time, when someone from the community works the issue.

If you have any other additional information, please attach it to the other ASTERISK issue.
                
> 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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