Summary:	ASTERISK-26553: pjsip: Cannot hear transcoded sound files in a g722 call
Reporter:	Daniel Heckl (DanielYK)	Labels:
Date Opened:	2016-11-04 10:21:29	Date Closed:	2016-11-18 12:54:34.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip_sdp_rtp
Versions:	13.12.1	Frequency of Occurrence
Related Issues:	duplicates ASTERISK-26603 [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no is related to ASTERISK-26423 res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
Environment:		Attachments:	( 0) codec_problem_log.txt ( 1) pjsip_debug_log_fix_4453.txt
Description:	If I offer g722 (first priority) and alaw (second priority) in a SDP, I do not hear sound files which are not in g722 and have to be transcoded. The log shows they are transcoded correctly (e.g. gsm -> alaw). If I only offer g722 OR only alaw OR alaw as first priority in the SDP, I hear the played sound files. In the log attached there are played some gsm files and one g722 file. I do only hear the g722 file.
Comments:	By: Asterisk Team (asteriskteam) 2016-11-04 10:21:30.765-0500 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 C. Colp (jcolp) 2016-11-08 05:37:49.312-0600 This appears to be one of the issues I fixed in ASTERISK-26423. If you can build Asterisk 13 from git you can confirm if it is fixed or not, otherwise you will need to wait for 13.13.0. By: Daniel Heckl (DanielYK) 2016-11-08 14:37:17.165-0600 I have build Asterisk branch 13 from git and tested the situation. The situation has not changed. By: Rusty Newton (rnewton) 2016-11-10 18:31:45.380-0600 Daniel can you provide an exact pjsip configuration and dialplan that would allow me to reproduce the issue simply? By: Alexei Gradinari (alexei gradinari) 2016-11-17 08:29:48.222-0600 Daniel, I think it's related to ASTERISK-26603. Could you, please, test my patch https://gerrit.asterisk.org/#/c/4453/ It should fixed this issue. By: Daniel Heckl (DanielYK) 2016-11-17 10:58:49.778-0600 Alexei, great, your patch has fixed my problem :) Good job! But it is not perfect. I have attached a debug log, there are a lot of strange debug notes. I sometimes read "Ooh, format changed". I think the changed format is expected, so there should not be those comments. By: Alexei Gradinari (alexei gradinari) 2016-11-17 11:31:40.348-0600 Daniel, The debug message "Oooh, got a frame with format.." should be present after any codec negotiation. This is normal. Turn off debug and you will not see this message. By: Daniel Heckl (DanielYK) 2016-11-17 11:36:23.008-0600 Great, then I confirm the patch as fix for the problem. By: Rusty Newton (rnewton) 2016-11-18 12:54:34.892-0600 Great! I'm going to go ahead and close this out as a duplicate of ASTERISK-26603.