Summary: | ASTERISK-24424: Warnings when ulaw/SILK16 is being transcoded | ||
Reporter: | Samuel For (samfun) | Labels: | |
Date Opened: | 2014-10-15 12:09:05 | Date Closed: | 2014-12-03 09:02:32.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Codecs/General |
Versions: | 11.13.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Ubuntu 14.04 Linode VPS 1GB Ram 1 CPU Core SSD drive | Attachments: | ( 0) codecs.conf ( 1) sample_sip.conf |
Description: | When a call is ongoing for us the following log is being shown in the CLI up to 18 times per second:
{noformat} [Oct 15 09:53:47] WARNING[3634][C-00000001]: abstract_jb.c:284 ast_jb_put: SIP/XXXXX-00000004 received frame with invalid timing info: has_timing_info=0, len=20, ts=20560, src=slin 8000khz -> 16000khz {noformat} The above message is only present when one leg has ulaw and the other has SILK/16. g722 and SILK/16 works fine. {noformat} Configuration Inbound leg: ulaw Outbound leg: SILK/16 jbenable = yes jbforce = yes jbmaxsize = 400 jbresyncthreshold = 1000 jbimpl = adaptive jbtargetextra = 70 jblog = no {noformat} | ||
Comments: | By: Matt Jordan (mjordan) 2014-10-28 09:21:47.509-0500 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information The SILK codec displays a few extra things at Verbose 6+. If you can, please produce the logs at that level. By: Rusty Newton (rnewton) 2014-11-13 17:18:13.364-0600 [~samfun] we haven't heard back from you in a few weeks. Will you be able to provide the debug requested? By: Rusty Newton (rnewton) 2014-12-03 09:01:56.719-0600 I'm unable to reproduce - I need your sip.conf files, rtp.conf, codecs.conf, versions of the hard or softphones that you are using. However, I'm going to go ahead and close this out as we don't have anyone else reporting this, and the reporter has not responded since the 28th of October. By: Rusty Newton (rnewton) 2014-12-03 09:02:16.923-0600 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines By: Eli Hunter (elihunter) 2015-10-05 08:45:21.855-0500 I'm seeing this issue with Polycom endpoints using G722 with ulaw at the other end after enabling the jitter buffer. This is on Asterisk 11.19. WARNING[11984][C-0000076d]: abstract_jb.c:284 ast_jb_put: SIP/18-RST-000019c7 received frame with invalid timing info: has_timing_info=0, len=20, ts=17140, src=slin 8000khz -> 16000khz The only options in my rtp.conf are rtpstart=10000 rtpend=20000 By: Daniel Denson (dandenson) 2016-06-09 23:30:40.692-0500 This isn't resolved as of 11.22.0 |