[asterisk-bugs] [Asterisk 0013401]: Transport type not correctly displayed in "sip show peer" output

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13401 
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Reported By:                jtodd
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13401
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 140491 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-08-29 14:59 CDT
Last Modified:              2008-09-09 04:59 CDT
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Summary:                    Transport type not correctly displayed in "sip show
peer" output
Description: 

I have a SIP peer which is connecting with * using TCP.  I have verified
both with netstat as well as with tshark that the peer is connected with
TCP.  However, "sip show peer blah" always shows Transport: as UDP.  The
SIP object is set with "transport=udp,tcp" in the config file.  The
expected behavior would be that Transport: shows the current transport
method that the SIP entity is using for the registration.  If it was the
desire to have "Transport:" indicate the setting in the config file, then I
would suggest another name for that field (possibly "Transport-Ord." or
something like that?) and the options be printed in that new field.
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 (0092219) putnopvut (administrator) - 2008-09-09 04:59
 http://bugs.digium.com/view.php?id=13401#c92219 
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After looking at the code for a while, I think I know what's up here.
Basically what the output is showing is the "preferred" transport method as
indicated in sip.conf. In other words, it will always display the first
transport type configured for that peer.

The tricky part of what you propose deals with the following "...shows the
current transport method that the SIP entity is using for the
registration." The problem with this is that as far as I can tell, the
transport method used can vary between calls. For instance, if your peer is
configured with transport=udp,tcp as you have here, I would expect Asterisk
to make outbound calls to this peer using UDP instead of TCP (unless there
is an overriding factor like explicitly placing the TCP transport in the
Dial string).

If I'm correct in the above paragraph, then I think a good compromise
would be to list all the transport types configured for the peer instead of
just the first. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-09-09 04:59 putnopvut      Note Added: 0092219                          
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