| Summary: | ASTERISK-23171: Crash in res_rtp_asterisk on WebRTC incoming call | ||||
| Reporter: | Beppo Mazzucato (beppo.it) | Labels: | |||
| Date Opened: | 2014-01-21 11:53:58.000-0600 | Date Closed: | 2014-02-12 22:38:56.000-0600 | ||
| Priority: | Major | Regression? | No | ||
| Status: | Closed/Complete | Components: | Resources/res_rtp_asterisk | ||
| Versions: | 11.7.0 | Frequency of Occurrence | Frequent | ||
| Related Issues: |
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| Environment: | CentOS 6.5 64 bit Google Chrome Version 32.0.1700.77 jssip 0.3.0 | Attachments: | ( 0) backtrace.txt ( 1) backtrace2.txt ( 2) log-dialplan-sip.txt ( 3) log-dialplan-sip2.txt | ||
| Description: | Asterisk crash sometimes on WebRTC incoming calls. The frequency depends from the number of concurrent user.
Never seen with 1-2 users it happen 3-4 times a day with 10 users. I'm attaching the backtrace of the last two incidents (unfortunately optimized) | ||||
| Comments: | By: Beppo Mazzucato (beppo.it) 2014-01-21 11:55:01.220-0600 Bug Bugtrace By: Rusty Newton (rnewton) 2014-01-21 13:29:52.852-0600 Sounds like you see the issue a few times a day. Can you provide the following? 1. Debug log capture up to the crash with all the options shown here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information 2. Configuration files for the SIP peers and dialplan involved 3. Backtraces, following instruction here https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace (with BETTER_BACKTRACES and DONT_OPTIMIZE) By: Beppo Mazzucato (beppo.it) 2014-01-21 14:35:36.487-0600 I'm attaching for the two crashes the portion of the logfile with the call causing the crash and the relevant portion of dialplan and sip.conf. Recompiling with BETTER_BACKTRACES and DONT_OPTIMIZE produce an error in Chrome (Called with an SDP without ice-ufrag and ice-pwd) don't allowing to make calls and reproduce the crash. I'll enter a separate issue for this By: Beppo Mazzucato (beppo.it) 2014-01-21 14:36:33.652-0600 log dialplan and sip.conf By: Rusty Newton (rnewton) 2014-02-10 08:33:03.416-0600 I see you are using the "opus" codec, which is not supported in Asterisk 11.7.0. Are you using a patched Asterisk 11.7.0, if so what all patches are you using? Otherwise are you using Asterisk 12 and what version or SVN revision? By: Beppo Mazzucato (beppo.it) 2014-02-10 08:53:59.415-0600 I'm using the patch here https://github.com/meetecho/asterisk-opus in asterisk 11.7.0 By: Matt Jordan (mjordan) 2014-02-12 22:37:47.376-0600 The backtrace attached to this issue appears to be a duplicate of ASTERISK-22938. In both cases, an RTCP packet was received, which called {{pj_ice_sess_on_rx_pkt}}. By: Matt Jordan (mjordan) 2014-02-12 22:38:56.159-0600 I'm closing this out as a duplicate. Keeping the issues linked should allow them to be resolved at the same time, when someone from the community works the issue. If you have any other additional information, please attach it to the other ASTERISK issue. | ||||