| Summary: | ASTERISK-28728: Asterisk crash in RTP stack (segfault) | ||||
| Reporter: | Joseph Hayhoe (jhayhoe) | Labels: | webrtc | ||
| Date Opened: | 2020-02-06 22:41:09.000-0600 | Date Closed: | 2020-02-08 11:21:25.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | pjproject/pjsip | ||
| Versions: | 13.29.2 | Frequency of Occurrence | |||
| Related Issues: |
| ||||
| Environment: | Attachments: | ( 0) core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-brief.txt ( 1) core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-full.txt ( 2) core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-locks.txt ( 3) core.dc3-freepbx1.int.liquidweb.com-2020-02-06T20-12-40+0000-thread1.txt | |||
| 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 | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2020-02-06 22:41:11.109-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Joseph Hayhoe (jhayhoe) 2020-02-06 22:44:10.700-0600 Attached core dumps By: George Joseph (gjoseph) 2020-02-07 10:39:02.481-0600 The backtraces don't have any symbols in them which makes it hard to tell what's going on. Please follow the instructions to install the debuginfo packages at: https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-Backtraces(Segfaults/CoreDumps/AsteriskCrashing) then run ast_coredumper again against the original core file. By: Joseph Hayhoe (jhayhoe) 2020-02-07 11:29:25.836-0600 Uploading new backtraces after setting up debuginfo By: George Joseph (gjoseph) 2020-02-07 14:35:56.985-0600 that's much better, thanks! Hang on to that coredump in case we need it to examine the state of the process when it segfaulted. By: Sean Bright (seanbright) 2020-02-08 11:21:25.604-0600 Duplicate of ASTERISK-28576. Fixed in Asterisk 13.30 | ||||