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

[Jacques Brooks (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24875?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jacques Brooks updated ASTERISK-24875:
--------------------------------------

    Attachment: refs.txt
                extensions.conf
                badmagiclog

So it does indeed look like it's an app_konference issue when using WEBRTC. We put together a very simple dialplan (as you can see) and we started getting the bad magic errors only when invoking app_konference with WEBRTC.  When we redid the same test using confbridge we didn't get bad magic errors; at least for the simple (and short) test that we did.

Not sure how I'm supposed to proceed now; assuming it's an app_konference issue does this leave your purview and I should try to contact app_konference originator(s)?

> 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: asterisklog.gz, backtrace2.txt, backtrace.txt, badmagiclog, extensions.conf, refs.txt, sip.conf
>
>
> 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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