[asterisk-bugs] [JIRA] (ASTERISK-29881) Segmentation fault when using PJSIP + TURN server + WebRTC
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Tue Feb 8 12:01:06 CST 2022
[ https://issues.asterisk.org/jira/browse/ASTERISK-29881?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258011#comment-258011 ]
Asterisk Team commented on ASTERISK-29881:
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Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
> Segmentation fault when using PJSIP + TURN server + WebRTC
> ----------------------------------------------------------
>
> Key: ASTERISK-29881
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29881
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: pjproject/pjsip
> Affects Versions: 18.9.0
> Environment: FreeBSD 13.0-RELEASE-p6
> Reporter: Martin Solciansky
> Assignee: Martin Solciansky
> Labels: webrtc
> Attachments: asterisk_bt_20220125.txt
>
>
> FreeBSD/asterisk server acts as a SIP gateway for WebRTC clients to a different asterisk server. All traffic to clients is routed via CoTURN server. Asterisk HTTP server is used to serve incoming WebRTC connections.
> Maximum number of SIP clients was 12 but problem occurs even with 1 client, usually occurs within few hours of server restart but we were unable to reproduce the segfault by simply loading the server with new sessions. It is somewhat random.
> Tried latest FreeBSD asterisk18-18.9.0 package, then tried recompiling from ports without optimization to produce a meaningful backtrace.
> Also tried different FreeBSD versions on different machines.
> This is a very minimalistic configuration and server only forwards traffic from SIP client to the other asterisk and vice-versa.
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