| Summary: | ASTERISK-26810: codec_opus: Crash when translating frame with opus section | ||||
| Reporter: | Moritz Maisel (mmaisel) | Labels: | |||
| Date Opened: | 2017-02-22 01:49:41.000-0600 | Date Closed: | 2017-11-28 09:54:21.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Codecs/codec_opus | ||
| Versions: | 14.3.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | debian 8.7 on amd64 architecture with kernel version 3.16.0-4-amd64 | Attachments: | ( 0) asterisk-console.log ( 1) backtrace.txt ( 2) codecs.conf ( 3) opus_crash_sip_and_rtp.pcap.gz ( 4) pjsip.conf | ||
| Description: | We experience reproducable crashes of asterisk with codec_opus. While asterisks successfully processes a couple of calls (about 8-10) when bridging two OPUS/48000/2 channels before it crashes, it reproducably crashes on the first call bridging PCMA/8000 to OPUS/48000/2.
Asterisk does not crash with missing codecs.conf. The sample codecs.conf provided with asterisk (that has no active opus settings) does work, but as soon as we add the {code}[opus]{code} section asterisk crashes as described above. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2017-02-22 01:49:43.466-0600 The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process. By: Asterisk Team (asteriskteam) 2017-02-22 01:49:43.878-0600 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]. By: Moritz Maisel (mmaisel) 2017-02-22 01:55:30.293-0600 backtrace generated using {code}sudo gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core{code} By: Moritz Maisel (mmaisel) 2017-02-22 03:12:35.528-0600 dialplan and codecs.conf By: Moritz Maisel (mmaisel) 2017-02-22 03:19:53.136-0600 replaces codecs.conf with stripped down version By: Joshua C. Colp (jcolp) 2017-02-22 04:48:32.262-0600 We need additional information for this. What endpoints are in use? What is the SIP and RTP traffic when the crash occurs, as well as the Asterisk console output. By: Moritz Maisel (mmaisel) 2017-02-23 05:03:05.424-0600 Please find asterisk console log, sample codecs.conf with enabled opus section, pcap of SIP and RTP as well as used pjsip.conf attached. We were able to reproduce the issue in an ubuntu xenial64 vagrant VM using codec_opus-14.0_1.1.0-x86_64.tar.gz By: Sarbyn (sarbyn) 2017-03-07 03:36:38.272-0600 Same issue on Centos 7, kernel 3.10.0-514.el7.x86_64 with asterisk 14.3.0 and also asterisk LTS 13.13-cert1 By: Krischan Dickel (krischan) 2017-10-11 08:30:33.179-0500 The issue is still occurring in asterisk 15.0.0. | ||||