[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=225558#comment-225558 ] 

Rusty Newton commented on ASTERISK-24875:
-----------------------------------------

If you are not getting a segfault immediately, but you are getting the 'bad magic number' messages, then you may first want to gather some reference count debugging output.

Please read through this guide: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging and follow the steps under Enabling Reference Count Logs. You can stop Asterisk and process/provide the output of the refs log right after you begin seeing the 'bad magic number' messages.

If you are doing this in Asterisk 13, and not using res_pjsip, then please disable res_pjsip by unloading the relevant modules before following those steps. [How to disable res_pjsip|https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-Disablingres_pjsipandchan_pjsip]



> 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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