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Summary:ASTERISK-20050: intermittent one way audio
Reporter:Jonn Taylor (jonnt)Labels:
Date Opened:2012-06-26 10:20:37Date Closed:2012-06-28 18:42:04
Priority:CriticalRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:SVN Frequency of
Occurrence		Frequent
Related
Issues:
Environment:CentOS 5.8 i686Attachments:
Description:Getting intermittent one way audio on SIP to IAX and SIP to USTM. I have also had this happen on USTM->IAX<->IAX->USTM. There were no error messages in the logs so I am unsure what addition info to provide. This started after revision 362497.
Comments:By: Rusty Newton (rnewton) 2012-06-28 18:41:48.247-0500

Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question to the community:  http://www.asterisk.org/support.


99% of the time one-way audio is caused by networking troubles and configuration changes. You'll want to find someone in the community to help you look over packet traces of the SIP/IAX traffic to examine whats going on. Look into using TCPDUMP to gather a log of the traffic, plus you'll want to correlate it with VERBOSE and DEBUG logs from Asterisk https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: Jonn Taylor (jonnt) 2012-06-30 10:07:02.080-0500

It is not a network issue! I have this two different systems. Both systems have 2 nic cards. One with a public IP and one with a private IP. RTP audio is passed from nic to the other as most of the phone on both systems are on the private side.

This started after the hangupcause and rtp changes to chan_sip. I will do some packet traces when I get back from vacation. I reveted both system back to 362497 and everything is working perfect!!!