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

[Joseph Hayhoe (JIRA)]

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

Joseph Hayhoe updated ASTERISK-28728:
-------------------------------------

    Description: 
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


  was:
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



> 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
>              Labels: webrtc
>
> 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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