Summary:	ASTERISK-26781: bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
Reporter:	Sean Bright (seanbright)	Labels:
Date Opened:	2017-02-09 12:04:47.000-0600	Date Closed:	2017-02-28 14:43:43.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Features
Versions:	13.14.0	Frequency of Occurrence	Frequent
Related Issues:
Environment:		Attachments:	( 0) bridge-tone-audio.pcap ( 1) debug_log_26781.txt
Description:	When calling {{Bridge(somechan,p)}}, the tone that is played on {{somechan}} is often choppy/garbled or sometimes not heard at all. This is not often obvious with the default tone ({{beep}}), but with longer sound files it is very pronounced. Looking at the packet capture and discussing with Josh on IRC, we discovered that while the tone is playing, Asterisk is sending two separate streams of RTP packets. One with the SSRC ({{0x1D68A751}} in .pcap) from the original un-bridged call, and the second with the SSRC ({{0x41CB61CA}} in .pcap) provided by the bridge audio. Once the tone is finishing playing, Asterisk sends packets only for {{0x41CB61CA}}, until the channel leaves the bridge. At that time, the {{0x1D68A751}} packets resume. I've attached a capture file for reference. {{192.168.26.66}} is Asterisk GIT-13, and {{192.168.26.1}} is Bria.
Comments:	By: Asterisk Team (asteriskteam) 2017-02-09 12:04:47.735-0600 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: Sean Bright (seanbright) 2017-02-09 12:16:46.921-0600 Debug log attached.  The packet capture is from an earlier call not related to the debug log I attached. If you would like both of them from the same attempt, let me know. By: Joshua C. Colp (jcolp) 2017-02-09 12:24:14.112-0600 It seems as though the native bridge is not being broken when it should be, causing two internal media streams to go out when only one should. By: Friendly Automation (friendly-automation) 2017-02-28 14:43:44.463-0600 Change 5090 merged by zuul: bridge_native_rtp: Handle case where channel joins already suspended. [https://gerrit.asterisk.org/5090|https://gerrit.asterisk.org/5090] By: Friendly Automation (friendly-automation) 2017-02-28 14:51:29.632-0600 Change 5091 merged by Joshua Colp: bridge_native_rtp: Handle case where channel joins already suspended. [https://gerrit.asterisk.org/5091|https://gerrit.asterisk.org/5091] By: Friendly Automation (friendly-automation) 2017-02-28 17:15:30.697-0600 Change 5092 merged by zuul: bridge_native_rtp: Handle case where channel joins already suspended. [https://gerrit.asterisk.org/5092|https://gerrit.asterisk.org/5092]