[asterisk-bugs] [Asterisk 0014418]: If a SIP URI is resolved with SRV records, the port must no be in the Request-URI

[Asterisk Bug Tracker]

The following issue has been set as RELATED TO issue 0014419. 
====================================================================== 
https://issues.asterisk.org/view.php?id=14418 
====================================================================== 
Reported By:                klaus3000
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14418
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.23 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-02-06 04:04 CST
Last Modified:              2009-08-20 14:21 CDT
====================================================================== 
Summary:                    If a SIP URI is resolved with SRV records, the port
must no be in the Request-URI
Description: 
Hi!

If Asterisk resolves a SIP domain by SRV, it should use the resolved
IP:port for sending the transport only, not changing the RURI.

E.g. if Asterisk sends a request to sip:user at mydomain.com and resolves
using SRV, then the requests URI must not contain the port.






    -- Now forwarding SIP/u+437206200730153-b7058588 to
'SIP/2 at app.innofon.at' (thanks to SIP/u+437206200730151-08ef44e8)
    -- ast_get_srv: SRV lookup for '_sip._udp.app.innofon.at' mapped to
host app.innofon.at, port 5160
Audio is at 81.16.153.184 port 12012
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 81.16.153.184:5160:
INVITE sip:2 at app.innofon.at:5160 SIP/2.0
Via: SIP/2.0/UDP 81.16.153.184:5160;branch=z9hG4bK462e8ee7;rport
From: "3 (SNOM 200)" <sip:3 at 81.16.153.184:5160>;tag=as054e6763
To: <sip:2 at app.innofon.at:5160>
Contact: <sip:3 at 81.16.153.184:5160>
Call-ID: 455000696c1b0a0829b83b384670155e at 81.16.153.184
CSeq: 102 INVITE
User-Agent: InnoSIP-app
Max-Forwards: 70
Date: Fri, 06 Feb 2009 09:55:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014419 [patch] Asterisk must not perform SRV l...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-08-20 14:21 mnicholson     Relationship added       related to 0014419  
======================================================================