[asterisk-bugs] [Asterisk 0019173]: RTP timestamp skewed after call transfer or call unhold

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19173 
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Reported By:                xxot
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19173
Category:                   Core/RTP
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.3.3 
JIRA:                       SWP-3383 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-04-23 18:50 CDT
Last Modified:              2011-04-27 07:47 CDT
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Summary:                    RTP timestamp skewed after call transfer or call
unhold
Description: 
During incoming call from outside on asterisk after call transfer (which is
performed on asterisk) to another extension remote party lost incoming
audio channel (one way audio). We find out that the reason is connected
with RTP timestamp which is jumped to huge abnormal values after transfer
or putting call on hold. This issue was described here
https://issues.asterisk.org/view.php?id=11491
and I believe here: https://issues.asterisk.org/view.php?id=17007
But there was bug in version 1.4. In 1.8.3.3 we have the same: timestamps
skewed to crazy values. In documentation to RTP there is an information
that RTP Marker should be set if RTP source is changed. It is really set
here, but many phones use timestamps for counting jitter and don't pay
attention on RTP Marker. This is documented bug in phones Cisco 7960 and it
is still not fixed. Is there any chance to fix this issue in asterisk and
keep RTP timestamps stable even after call transfer?
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Relationships       ID      Summary
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related to          0011491 RTP timestamp skewed after return from ...
related to          0017007 [patch] RTP Timestamp changes after tra...
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 (0134176) xxot (reporter) - 2011-04-27 07:47
 https://issues.asterisk.org/view.php?id=19173#c134176 
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All SIP phones which are connected to asterisk behind the NAT and we have
no directmedia here. RTP source, actually, asterisk itself. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-27 07:47 xxot           Note Added: 0134176                          
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