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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Aug 12 20:27:36 CDT 2010


The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=17851 
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Reported By:                whardier
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17851
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
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:             2010-08-12 20:27 CDT
Last Modified:              2010-08-12 20:27 CDT
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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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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-12 20:27 whardier       New Issue                                    
2010-08-12 20:27 whardier       Asterisk Version          => Older 1.6.2 -
please test a newer version
2010-08-12 20:27 whardier       Regression                => No              
2010-08-12 20:27 whardier       SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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