[asterisk-bugs] [JIRA] (ASTERISK-28728) Asterisk crash in RTP stack (segfault)

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28728?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-28728:
-------------------------------------

    Assignee: Asterisk Team  (was: Joseph Hayhoe)
      Status: Triage  (was: Waiting for Feedback)

> Asterisk crash in RTP stack (segfault)
> --------------------------------------
>
>                 Key: ASTERISK-28728
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28728
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.29.2
>            Reporter: Joseph Hayhoe
>            Assignee: Asterisk Team
>              Labels: webrtc
>         Attachments: core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-brief.txt, core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-full.txt, core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-locks.txt, core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-thread1.txt
>
>
> We experienced a segmentation fault in our server today. We recently upgraded our asterisk version in the last 24 hours from 13.27.1 to 13.29.2-1. Attaching the files from ast-coredumper to this report. We have experienced this issue in the past and it was related to pjsip which we use heavily with WebRTC for our endpoint clients.
> Asterisk logs around the time of the crash are listed below:
> [2020-02-06 20:12:39] VERBOSE[31512][C-00000232] app_dial.c: PJSIP/SAT-ATT-SIP-00000451 is making progress passing it to PJSIP/5159-00000450
> [2020-02-06 20:12:39] VERBOSE[31512][C-00000232] app_dial.c: PJSIP/SAT-ATT-SIP-00000451 is making progress passing it to PJSIP/5159-00000450
> [2020-02-06 20:12:39] ERROR[21202] pjproject:      icess0x7f449811db98 ...Error sending STUN request: Invalid argument
> [2020-02-06 20:12:39] VERBOSE[31512][C-00000232] app_dial.c: Call on PJSIP/SAT-ATT-SIP-00000451 placed on hold
> [2020-02-06 20:12:39] VERBOSE[31512][C-00000232] res_musiconhold.c: Started music on hold, class 'default', on channel 'PJSIP/SAT-ATT-SIP-00
> 000451'
> [2020-02-06 20:12:39] ERROR[22532] pjproject:      icess0x7f449811db98 ..Error sending STUN request: Invalid argument
> [2020-02-06 20:12:44] Asterisk 13.29.2 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2019-11-22 00:59:22 UTC



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