[asterisk-bugs] [Asterisk 0012566]: Loss of audio after 2 seconds of P2P RTP bridging

noreply at bugs.digium.com noreply at bugs.digium.com
Fri May 16 08:38:25 CDT 2008


The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=12566 
====================================================================== 
Reported By:                falves11
Assigned To:                blitzrage
====================================================================== 
Project:                    Asterisk
Issue ID:                   12566
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 114912 
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             05-01-2008 13:23 CDT
Last Modified:              05-16-2008 08:38 CDT
====================================================================== 
Summary:                    Loss of audio after 2 seconds of P2P RTP bridging
Description: 
I have a peer like this

[federico]
type=peer
host=72.187.243.97
context=default
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=ulaw
nat=yes
insecure=port,invite

but when I type "sip show peers"
ame/username              Host            Dyn Nat ACL Port     Status
federico                   72.187.243.97        N      5060    
Unmonitored

and in fact the calls connect and there is second of audio and then the
audio stops. I am behind a nat.
====================================================================== 

---------------------------------------------------------------------- 
 blitzrage - 05-16-08 08:38  
---------------------------------------------------------------------- 
jpeeler found a locking issue in chan_sip, which appears to have resolved
the issue for at least 2 people.

Thanks to everyone who helped to resolve this issue! 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-16-08 08:38  blitzrage      Assigned To               => blitzrage       
05-16-08 08:38  blitzrage      Note Added: 0086940                          
======================================================================




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