Summary:	ASTERISK-26654: chan_sip: ILBC or Opus codec offer correlates with one-way audio
Reporter:	Luke Escude (lukeescude)	Labels:
Date Opened:	2016-12-09 13:48:57.000-0600	Date Closed:	2016-12-19 18:00:37.000-0600
Priority:	Critical	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/CodecHandling
Versions:	13.13.1	Frequency of Occurrence
Related Issues:
Environment:	CentOS x64 - No NAT, public IP	Attachments:	( 0) 2calls.pcap ( 1) asterisk_debug ( 2) extensions.conf ( 3) GOOD_AstDebug ( 4) GOODCALL.pcap ( 5) sip.conf
Description:	We have been struggling with random (very inconsistent) one-way audio phone calls for a couple of months now. Out of the 10,000 phone calls we process per day, this only happens to less than 20 of them. This issue ONLY affects a few of my customers on a raw Asterisk installation, none of my FreePBX customers are having any issues. The server is hosted in our datacenter with a public IP - no firewall, no NAT, and it trunks to Flowroute. The customer is behind NAT, however the system works just fine like this - we always have 2-way audio between customer and our server. Attached is a PCAP of 2 phone calls that experienced one-way audio, and the console debug messages from asterisk. Flowroute is pulling a packet capture of their side for me, as we speak. New info: Calling her works flawlessly on FreePBX.
Comments:	By: Asterisk Team (asteriskteam) 2016-12-09 13:48:57.592-0600 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: Luke Escude (lukeescude) 2016-12-09 13:49:41.152-0600 Attached: PCAP and debug output. By: Luke Escude (lukeescude) 2016-12-09 14:10:37.194-0600 Asterisk 13.13.1, x64 I'll post up configuration for the peer and trunk in just a bit By: Luke Escude (lukeescude) 2016-12-09 14:17:42.132-0600 Attached extensions.conf - just the context that particular extension uses to dial out. By: Luke Escude (lukeescude) 2016-12-09 15:02:49.024-0600 Attached: The same phone call made with FreePBX - Works Perfectly! By: Luke Escude (lukeescude) 2016-12-09 15:30:15.003-0600 We have been able to identify that this only happens with Sprint phones. However FreePBX still works with Sprint phones... By: Luke Escude (lukeescude) 2016-12-10 18:39:12.015-0600 FIGURED IT OUT! This is iLBC related. The iLBC codec is IGNORING rport,comedia settings. Even though none of these calls are even using iLBC, the simple fact that iLBC is being offered makes the call lose audio. A ticket with similar issues: ASTERISK-26593 By: Luke Escude (lukeescude) 2016-12-10 18:43:05.146-0600 It should be noted that this also occurs with opus. By: Rusty Newton (rnewton) 2016-12-13 08:52:58.779-0600 Looks goofy. Opening this up.   Also, a reminder that chan_sip is under extended support since Asterisk 12, so the best way to get this issue resolved quickly is to submit a patch yourself. Otherwise, extended support modules rely on the support of the broader community as they have no guarantee of support from the core team. If you can reproduce these issues with res_pjsip/chan_pjsip (core support) then open a new issue for that. Thanks! By: Luke Escude (lukeescude) 2016-12-13 08:59:06.624-0600 I'm surprised at that actually... the last time I tried PJSIP (over a year ago) it was having major BLF problems with several different model phones. I still think of it as unready for production, but I may be biased from that experience. By: Rusty Newton (rnewton) 2016-12-13 09:08:02.308-0600 It is definitely ready for production. We've had good reports from hundreds of users using it in major production environments. All core team SIP development focus is on PJSIP now. (and has been since Asterisk 12) By: Luke Escude (lukeescude) 2016-12-19 17:39:04.297-0600 I recompiled with PJSIP, and completely did away with chan_sip entirely - Some really good success with other features I was looking forward to utilizing, but Opus still poses a problem. I'll do some more testing and capture some logs for you guys to look at. By: Luke Escude (lukeescude) 2016-12-19 18:00:37.380-0600 Will start a new ticket regarding PJSIP and Opus codec offer.