[asterisk-bugs] [Asterisk 0017851]: SIp/SDP for Non-NAT phone not used during 183 Session Progress with Non-NAT peer

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=17851 
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Reported By:                whardier
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   17851
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                       SWP-2176 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2010-08-12 20:27 CDT
Last Modified:              2011-01-17 16:15 CST
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Summary:                    SIp/SDP for Non-NAT phone not used during 183
Session Progress with Non-NAT peer
Description: 
SIP phone sends initial SDP for media path and makes a call through
chan_sip to a peer that sends back a 183 Session Progress with media.  I
cannot force the peer not to send a 183 it seems.

As soon as a 183 is received from the peer chan_sip sends a new SDP to the
phone and the peer to including itself in the media path and proxies the
media for the session progress between the peer and the phone.  Once
connected a new SDP is sent to the phone and the peer referencing eachother
and a native bridge is done.

Is this the intentional behavior of the SIP channel driver - will it
always attempt to proxy media when a 183 is present?  I am currently
attempting to save bandwidth by using the 'r' flag when dialing this peer
to keep from sending the phone session audio.  Session audio is still
present of course between the peer and Asterisk system.

I am classifying this as major priority since the difference in bandwidth
usage is substantial when you plan on only handling SIP traffic without
media.  The servers I am using handle media for some peers that don't have
heavy bandwidth requirements - however when you plan on only handling
signalling for some heavier peers the session progress can eat all your
bandwidth pretty quickly.
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 (0130571) russell (administrator) - 2011-01-17 16:15
 https://issues.asterisk.org/view.php?id=17851#c130571 
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Yes, this is the expected behavior of Asterisk in this situation. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-17 16:15 russell        Note Added: 0130571                          
2011-01-17 16:15 russell        Status                   acknowledged =>
resolved
2011-01-17 16:15 russell        Resolution               open => no change
required
2011-01-17 16:15 russell        Assigned To               => russell         
2011-01-17 16:15 russell        Status                   resolved => closed  
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