[asterisk-bugs] [Asterisk 0018601]: No RTP port update when SIP RE-INVITE is received

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18601 
====================================================================== 
Reported By:                kondik
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18601
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.15 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-01-11 16:10 CST
Last Modified:              2011-01-19 06:23 CST
====================================================================== 
Summary:                    No RTP port update when SIP RE-INVITE is received
Description: 
Exactly the same issue as: https://issues.asterisk.org/view.php?id=15149
but those issue is closed, so I open new issue.

When RE-INVITE comes to asterisk with new media port, asterisk sends 200
OK, and send RTP packets further to the old port instead of new port which
received with RE-INVITE.

My asterisk version is: 1.6.2.15

Could you help me with this?
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0015149 No audio on SIP RE-INVITE connecting wi...
====================================================================== 

---------------------------------------------------------------------- 
 (0130663) davidw (reporter) - 2011-01-19 06:23
 https://issues.asterisk.org/view.php?id=18601#c130663 
---------------------------------------------------------------------- 
It is a perfectly legitimate thing for a UAC to do!  200 is the correct
response.  The problems lies with the client, if it is making changes to
the SDP without updating the SDP version. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-19 06:23 davidw         Note Added: 0130663                          
======================================================================