[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. 
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http://bugs.digium.com/view.php?id=12670 
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Reported By:                jolan
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12670
Category:                   Core/RTP
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     feedback
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:              
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Date Submitted:             05-16-2008 15:55 CDT
Last Modified:              06-03-2008 19:00 CDT
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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".
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---------------------------------------------------------------------- 
 jolan - 06-03-08 19:00  
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I attached a captures of both a problem call (14badcall.pcap) and a call
that had no audio problems (14goodcall.pcap).

I don't see any obvious time gaps, different packet contents, different
packet ordering, etc. but it'd be nice if someone could confirm that.

I'm going to revert to Asterisk 1.2 and get a capture of a call there to
see what's different that is freaking these Polycoms out. 

Issue History 
Date Modified   Username       Field                    Change               
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06-03-08 19:00  jolan          Note Added: 0087766                          
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