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

Jacques Brooks (JIRA) noreply at issues.asterisk.org
Fri Mar 27 07:29:34 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24875?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225650#comment-225650 ] 

Jacques Brooks commented on ASTERISK-24875:
-------------------------------------------

We've been using app_konference since our product went live, 4-5 years ago.  Asterisk 13 did need a newer version of app_konference to work properly. With the possibility in mind that it could be app_konference we fell back to asterisk 11.16, which enabled us to use a prior version of app_konference, and that we knew had no crashing issues processing "regular" sip traffic but we still crashed when throwing WEBRTC into the mix  Of course the combination of app_konference and WEBRTC could be the culprit but I'm not expert enough to comment on that (-:

> 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, 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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