[asterisk-bugs] [JIRA] (ASTERISK-24875) Randomly get segfaults processing WEBRTC calls

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24875?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225472#comment-225472 ] 

Rusty Newton edited comment on ASTERISK-24875 at 3/16/15 1:16 PM:
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Please attach both the sip.conf you are using successfully and the pjsip.conf that you used when observing the crash.

If you can reproduce the issue, please grab an Asterisk log captured during the time of the crash. 

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Use "Send Back" or "Enter Feedback" to send the issue back after you have this information. Thanks.


was (Author: rnewton):
Please attach both the sip.conf you are using successfully and the pjsip.conf that you used when observing the crash.

If you can reproduce the issue, please grab an Asterisk log captured during the time of the crash. 

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

> Randomly get segfaults processing WEBRTC calls
> ----------------------------------------------
>
>                 Key: ASTERISK-24875
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24875
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.16.0, 13.2.0
>         Environment: CentOS Linux
>            Reporter: Jacques Brooks
>            Assignee: Jacques Brooks
>            Severity: Critical
>         Attachments: backtrace.txt
>
>
> Asterisk randomly crashes when processing WEBRTC calls. Doesn't seem to be dependent on number of calls currently handling or how long Asterisk is running; have crashed with less than 10 calls and over 200 calls and have crashed with Asterisk running less than 20 minutes and at times when it's been running for hours  Using version 13.2 with sip.conf.  Tried to convert to pjsip.conf but was not very successful (would get only one or two calls up before crashing) so reverted back to sip.conf. 



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