| Summary: | ASTERISK-27194: jitterbuffer: Does not handle case where translator returns null frame. | ||
| Reporter: | Joshua Elson (joshelson) | Labels: | pjsip |
| Date Opened: | 2017-08-10 12:19:49 | Date Closed: | 2017-10-27 09:06:38 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Core/Jitterbuffer |
| Versions: | 14.6.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ( 0) adaptivejb_opus_crash.txt | |
| Description: | Asterisk 14.6 with adaptive jitterbuffer running and using Digium codec_opus can crash.
Last lines of dialplan output are: {noformat} [2017-08-10 10:54:59] VERBOSE[30202][C-00005d41] app_dial.c: Called PJSIP/7203986249@FC-DFW-PROXY [2017-08-10 10:54:59] VERBOSE[30202][C-00005d41] app_dial.c: PJSIP/FC-DFW-PROXY-00008950 is ringing [2017-08-10 10:54:59] VERBOSE[30202][C-00005d41] app_dial.c: PJSIP/FC-DFW-PROXY-00008950 answered PJSIP/6249-Company-0000894f [2017-08-10 10:54:59] VERBOSE[30202][C-00005d41] bridge_channel.c: Channel PJSIP/6249-Company-0000894f joined 'simple_bridge' basic-bridge <c1117a7f-a631-4fe0-91c4-cb81c041e179> [2017-08-10 10:54:59] VERBOSE[30202][C-00005d41] res_rtp_asterisk.c: 0x7f9e74810d40 -- Probation passed - setting RTP source address to 73.181.14.245:58905 [2017-08-10 10:54:59] ERROR[30202][C-00005d41] codec_opus.c: Opus: decoding - corrupted stream [2017-08-10 10:54:59] ERROR[30202][C-00005d41] codec_opus.c: Opus: decoding - corrupted stream [2017-08-10 10:54:59] ERROR[30202][C-00005d41] codec_opus.c: Opus: decoding - buffer too small {noformat} | ||
| Comments: | By: Asterisk Team (asteriskteam) 2017-08-10 12:19:50.508-0500 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: Joshua Elson (joshelson) 2017-08-10 12:22:11.354-0500 Backtrace attached. By: Joshua Elson (joshelson) 2017-08-10 12:29:23.461-0500 Backtrace attached. By: Friendly Automation (friendly-automation) 2017-10-27 09:06:39.589-0500 Change 6921 merged by Joshua Colp: codec.c: Defensively check the returned samples. [https://gerrit.asterisk.org/6921|https://gerrit.asterisk.org/6921] By: Friendly Automation (friendly-automation) 2017-10-27 09:19:56.139-0500 Change 6923 merged by Joshua Colp: codec.c: Defensively check the returned samples. [https://gerrit.asterisk.org/6923|https://gerrit.asterisk.org/6923] By: Friendly Automation (friendly-automation) 2017-10-27 09:40:21.466-0500 Change 6922 merged by Jenkins2: codec.c: Defensively check the returned samples. [https://gerrit.asterisk.org/6922|https://gerrit.asterisk.org/6922] | ||