[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:              06-05-2008 17:55 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 - 06-05-08 17:55  
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I'm pretty sure this is an Asterisk bug related to RTP timestamps.

With Asterisk-1.2 Cisco<->Polycom, the timestamp of the first RTP packet
is set to 0.

With Asterisk-1.4 IAX<->Polycom, the timestamp of the first RTP packet is
set to 160.

With Asterisk-1.4 Cisco<->Polycom, the timestamp of the first RTP packet
is seemingly random. 

Issue History 
Date Modified   Username       Field                    Change               
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06-05-08 17:55  jolan          Note Added: 0087873                          
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