[Home]

Summary:ASTERISK-08566: Last few daily builds of 1.4-svn have produced distorted audio in SIP
Reporter:Dan Moschuk (dnatural)Labels:
Date Opened:2007-01-12 11:12:26.000-0600Date Closed:2007-02-03 18:13:40.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debug.txt
Description:Outbound calls placed on the PSTN are fine, but internal calls have one way audio.  Alice calls Bob and can hear him but Bob cannot hear Alice.  Both Alice and Bob are using NAT, although not on the same local area network.

****** ADDITIONAL INFORMATION ******

1.4.0 does not have this issue.  It's creeped up only very recently.
Comments:By: Serge Vecher (serge-v) 2007-01-12 12:33:57.000-0600

As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Trim startup information and attach verbosedebug.txt to the issue.
By: Dan Moschuk (dnatural) 2007-01-12 13:32:42.000-0600

I just received a bit more accurate description of the reported effects -- apparently "one way audio" is a bit of a misnomer... something is making it to the other end, but it sounds like say... 1  audio frame out of every 100.  Note that it's 100% reproducible.
By: Anthony LaMantia (alamantia) 2007-01-12 17:33:45.000-0600

can you try to collect the information request by serge-v?
By: Dan Moschuk (dnatural) 2007-01-12 17:42:38.000-0600

yes it's already been attached to the bug
By: Serge Vecher (serge-v) 2007-01-15 10:00:47.000-0600

well, looks like two channels are successfully bridged via p2p method. What happens if you add rtp debug to the picture?
By: Olle Johansson (oej) 2007-02-01 15:36:39.000-0600

There's been a few changes to the RTP bridge lately. Please test with latest 1.4 from subversion. Thanks.
By: Dan Moschuk (dnatural) 2007-02-03 17:14:25.000-0600

All fixed!  Thanks.
By: Joshua C. Colp (jcolp) 2007-02-03 18:13:39.000-0600

Fixed all 'round.