[asterisk-bugs] [Asterisk 0013672]: Additional codecs are added to the SDP after a "Moved Temporarily" mesage - SIP TCP

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13672 
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Reported By:                mattdarnell
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13672
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.0.5
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-10-10 20:25 CDT
Last Modified:              2009-03-17 13:04 CDT
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Summary:                    Additional codecs are added to the SDP after a
"Moved Temporarily" mesage - SIP TCP
Description: 
After receiving a 302 SIP response Asterisk adds additional codecs to the
SPD when they are not allowed in sip.conf.  The only codec allowed in this
example is ulaw.

This is a 1.6.0 system with "0013523: Trouble with Temporarily Moved"
applied

    -- Got SIP response 302 "Moved Temporarily" back from 10.10.20.31
Transmitting (no NAT) to 10.10.20.31:5060:
ACK sip:3451 at 10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
Max-Forwards: 70
From: "4000" <sip:3451 at wikitelcom.com>;tag=as43e0f2af
To: <sip:3451 at 10.10.20.31>;tag=f4177796d8
Contact: <sip:3451 at 10.10.20.50:5060;transport=TCP>
Call-ID: 3deb3e4b3f6dc02c709391dd7d73f566 at wikitelcom.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0


---
    -- Now forwarding SIP/4000-b7d5ac38 to
'SIP/3451::::TCP at 10.10.20.31:5065' (thanks to SIP/sip-tcp-0831ed70)
  == Using SIP RTP CoS mark 5
Audio is at 10.10.20.50 port 15016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
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 (0101883) qwell (administrator) - 2009-03-17 13:04
 http://bugs.digium.com/view.php?id=13672#c101883 
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Have you verified that this is using the same peer to go out after the 302?
 It looks as though the dst port is changing from 5060 to 5065.

I'm not too well versed in SIP, but would setting insecure=port on the
peer perhaps fix this?  If no peer is matched, I believe it will use the
codec options specified in [general] (or whatever the default is). 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-17 13:04 qwell          Note Added: 0101883                          
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