[asterisk-bugs] [Asterisk 0012670]: some RTP packets sent to NAT IP instead of public IP; breaks built-in jitterbuffer on some phones

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12670 
====================================================================== 
Reported By:                jolan
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12670
Category:                   Core/RTP
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.19-rc3 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             05-16-2008 15:55 CDT
Last Modified:              05-16-2008 16:41 CDT
====================================================================== 
Summary:                    some RTP packets sent to NAT IP instead of public
IP; breaks built-in jitterbuffer on some phones
Description: 
I am experiencing the same symptoms as
http://bugs.digium.com/view.php?id=0012566 but with 1.4.20-rc3, not
trunk.  The bug was fixed in trunk, but not in 1.4.x.

To reiterate the problem:

Phone 100 calls phone 102.  Phone 102 answers and starts counting "1 2 3 4
5".  Phone 100 doesn't hear anything until "3" or "4".
====================================================================== 

---------------------------------------------------------------------- 
 file - 05-16-08 16:41  
---------------------------------------------------------------------- 
It sends audio to the NATted IP address and port of the phone you called.
That phone sent back a 200 OK with an RTP address of 10.0.0.96 port 2236.
Once that phone sent a packet to Asterisk and Asterisk saw it's source IP
address and port it switched over. Now for the phone that you placed the
call from Asterisk never actually changed the IP address and port it was
sending to. It was always 76.171.35.171 port 54729 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-16-08 16:41  file           Note Added: 0086962                          
======================================================================