Summary:	ASTERISK-25446: Warnings with jitter buffer enabled and transcoding from G722 to ulaw
Reporter:	Eli Hunter (elihunter)	Labels:
Date Opened:	2015-10-05 13:25:45	Date Closed:	2020-01-14 11:13:38.000-0600
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Codecs/General
Versions:	11.19.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Centos 6.5 x64	Attachments:	( 0) codecs.conf ( 1) sample_sip.conf
Description:	This is the same as ASTERISK-24424. I'm seeing this issue with Polycom endpoints using G722 with ulaw at the other end after enabling the jitter buffer 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 sip.conf and codecs.conf are attached
Comments:	By: Asterisk Team (asteriskteam) 2015-10-05 13:25:48.053-0500 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: Eli Hunter (elihunter) 2015-10-05 13:26:43.746-0500 sip.conf and codecs.conf By: Rusty Newton (rnewton) 2015-10-07 17:58:49.987-0500 We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Rusty Newton (rnewton) 2015-10-07 17:59:59.326-0500 In addition to the debug logs please provide detailed information on the endpoints you are using (model and firmware). You don't describe an issue other than the WARNING messages. Is any issue presented to the user? (such as audio issues) By: Eli Hunter (elihunter) 2015-10-17 02:07:21.080-0500 I've found that removing the jbforce=1 line stops the warnings from showing.  I did have reports of call quality issues from multiple users while this was enabled however I'm not sure it was this. It's producing somewhere around 40-50 warnings per second for each active call which seems excessive. I'll work on getting a debug of this. By: Rusty Newton (rnewton) 2015-10-22 18:36:37.291-0500 Thanks, yes the debug I think will be helpful. That does sound like an excessive rate of warnings. By: Asterisk Team (asteriskteam) 2015-11-06 12:00:19.039-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Daniel Denson (dandenson) 2016-06-09 23:32:27.481-0500 not resolved as of v11.22.0 jbforce=yes definitely triggers issue, =no and it's fine.