RE: [Sip] Record-Route-Header ftag-field and routing

"Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com> Tue, 18 December 2007 19:10 UTC

Return-path: <sip-bounces@ietf.org>
Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1J4hpX-0005gL-0r; Tue, 18 Dec 2007 14:10:15 -0500
Received: from sip by megatron.ietf.org with local (Exim 4.43) id 1J4hpV-0005g2-TT for sip-confirm+ok@megatron.ietf.org; Tue, 18 Dec 2007 14:10:13 -0500
Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1J4hpV-0005e8-G8 for sip@ietf.org; Tue, 18 Dec 2007 14:10:13 -0500
Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1J4hpU-0000CV-Nw for sip@ietf.org; Tue, 18 Dec 2007 14:10:13 -0500
Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 18 Dec 2007 14:10:12 -0500
Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11/8.12.11) with ESMTP id lBIJACcU022155; Tue, 18 Dec 2007 14:10:12 -0500
Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id lBIJABij019896; Tue, 18 Dec 2007 19:10:12 GMT
Received: from xmb-rtp-215.amer.cisco.com ([64.102.31.124]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 18 Dec 2007 14:10:11 -0500
X-MimeOLE: Produced By Microsoft Exchange V6.5
Content-class: urn:content-classes:message
MIME-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
Subject: RE: [Sip] Record-Route-Header ftag-field and routing
Date: Tue, 18 Dec 2007 14:10:11 -0500
Message-ID: <8983EC086A9D954BA74D9763E853CF3E04615EF9@xmb-rtp-215.amer.cisco.com>
In-Reply-To: <200712181903.50731.mf@dafuer.com>
X-MS-Has-Attach:
X-MS-TNEF-Correlator:
Thread-Topic: [Sip] Record-Route-Header ftag-field and routing
Thread-Index: AchBoG/KoYClhXYoTNu33pPK6iTxhgABqhIg
From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: mf@dafuer.com, sip@ietf.org
X-OriginalArrivalTime: 18 Dec 2007 19:10:11.0933 (UTC) FILETIME=[9CFF88D0:01C841A9]
DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; l=4700; t=1198005012; x=1198869012; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=sanjsinh@cisco.com; z=From:=20=22Sanjay=20Sinha=20(sanjsinh)=22=20<sanjsinh@cisc o.com> |Subject:=20RE=3A=20[Sip]=20Record-Route-Header=20ftag-fiel d=20and=20routing |Sender:=20 |To:=20<mf@dafuer.com>,=20<sip@ietf.org>; bh=0AXF7/7/q4kFnURe2DPeQuZHCW+IMKJlz0z3LgUQ9uE=; b=hkWx8rTDcRjg0EnoGNKLcvNk5eGeEfN97Sj5G1/I/q3xEVrPYof8Ht0bu6 TfZSBr0hzo+wSxrvwgaxg6n7dewohE0LPWSr1AzyQQmaxsFXfOwLfStkJEqJ Ld3DCBzfIE;
Authentication-Results: rtp-dkim-1; header.From=sanjsinh@cisco.com; dkim=pass ( sig from cisco.com/rtpdkim1001 verified; );
X-Spam-Score: -2.5 (--)
X-Scan-Signature: a1852b4f554b02e7e4548cc7928acc1f
Cc:
X-BeenThere: sip@ietf.org
X-Mailman-Version: 2.1.5
Precedence: list
List-Id: Session Initiation Protocol <sip.ietf.org>
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sip>, <mailto:sip-request@ietf.org?subject=unsubscribe>
List-Post: <mailto:sip@ietf.org>
List-Help: <mailto:sip-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sip>, <mailto:sip-request@ietf.org?subject=subscribe>
Errors-To: sip-bounces@ietf.org

This would've been appropriate on sip-implementers mailer.

The behavior below looks correct to me and has nothing to do with ftag
field in RR header. 

In the 1st case since UAS is getting a request with a RR header, next
request on that dialog is sent to the address in RR header and since the
proxy adding RR supports loose routing, request uri of BYE is contact
address from previous request.

In the 2nd case, since there are 2 RR headers, route set for UAS is:
<sip:213.218.28.202;lr=on>, <sip:213.218.28.102;lr=on>. 
And the next request on that dialog is correctly sent to 213.218.28.202.


Sanjay


>-----Original Message-----
>From: Frank, Marc [mailto:mf@dafuer.com] 
>Sent: Tuesday, December 18, 2007 1:04 PM
>To: sip@ietf.org
>Subject: [Sip] Record-Route-Header ftag-field and routing
>
>Hi guys,
>
>because of some problems with my SIP-Client, I have traced the 
>traffic of a snom SIP-phone. Now I'm a little confused about 
>the Record-Route header.
>
>I've made two incoming testcalls with different providers.
>
>At the first test the snom-phone received an invite-message 
>with record-route with ftag-field. When the phone hangup it 
>sends it's BYE-Request to the IP-address defined in the last 
>route-header-field.
>
>At the second test (2nd provider) the snom-phone received an 
>invite-message with record-route and NO ftag-field. When the 
>phone hangup it sends it's BYE-Request to the IP-address of 
>the registrar.
>
>Is it correct, that a SIP-application which receives a 
>record-route with ftag must send the traffic depending on this 
>call to that destination?
>
>Regards 
>
>Marc :-)
>
>
>Trace-Snapshot:
>
>
>Call 1 with re-routing
>---------------------------------------------------------------
>-----------------
>Received from udp:212.227.15.197:5060 at 18/12/2007 
>14:00:09:980 (1392 bytes):
>
>INVITE sip:aaaaa at xx.xxx.229.7:2054;line=j3eted9k SIP/2.0
>Record-Route: <sip:212.227.15.232;ftag=1937134676;lr=on>
>Via: SIP/2.0/UDP 
>212.227.15.197;branch=z9hG4bK59f8248d2c568739bc7dc8a7de9c4f82
>Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK589.5362a815.0
>Via: SIP/2.0/UDP 
>212.227.15.197;branch=z9hG4bK202ed21ce8126c48fd64f9f8458a705d
>Via: SIP/2.0/UDP sipgw01.bmcag.com:5060
>;received=62.206.6.140;branch=z9hG4bKterm-2ab0b6-496151951416-4
>96214255114
>From: caller <sip:caller at 
>sipgw01.bmcag.com;user=phone>;tag=1937134676
>To: called <sip:called at 212.227.15.197;user=phone>
>Contact: <sip:caller at sipgw01.bmcag.com:5060>
>
>...
>
>Sent to udp:212.227.15.232:5060 at 18/12/2007 14:00:32:380 (695 bytes):
>
>BYE sip:caller at sipgw01.bmcag.com:5060 SIP/2.0
>Via: SIP/2.0/UDP xx.xxx.229.7:2054;branch=z9hG4bK-54bqxvq49o3d;rport
>Route: <sip:212.227.15.232;ftag=1937134676;lr=on>
>From: "called" <sip:called at 212.227.15.197;user=phone>;tag=ckf1qvklvd
>To: "caller" <sip:caller at 
>sipgw01.bmcag.com;user=phone>;tag=1937134676
>Call-ID: 7e38a35-4274d41-7d136ce2-b115 at sipgw01.bmcag.com
>CSeq: 1 BYE
>---------------------------------------------------------------
>-----------------
>
>
>
>Call 2 without re-routing
>---------------------------------------------------------------
>-----------------
>
>Received from udp:213.218.28.202:5060 at 18/12/2007 
>18:02:31:170 (1422 bytes):
>
>INVITE sip:user at xx.xxx.229.7:2060;line=i8vjxde1 SIP/2.0
>Record-Route: <sip:213.218.28.202;lr=on>
>Record-Route: <sip:213.218.28.102;lr=on>
>Via: SIP/2.0/UDP 213.218.28.202;branch=z9hG4bKbd69.7040b7b5.0
>Via: SIP/2.0/UDP 213.218.28.102;branch=z9hG4bKbd69.b2ae7011.0
>Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F510055C3A
>From: "caller" <sip:caller at provider.net;user=phone>;tag=00E0F5100
>To: <sip:called at provider.net;user=phone>
>Call-ID: 00E0F510055C3A11C0C5000025B9@213.218.28.164
>CSeq: 34207 INVITE
>Contact: <sip:caller at ip>
>
>...
>
>Sent to udp:213.218.28.202:5060 at 18/12/2007 18:02:43:310 (731 bytes):
>
>BYE sip:called at ip SIP/2.0
>Via: SIP/2.0/UDP 80.152.229.7:2060;branch=z9hG4bK-t7jlwh3sb054;rport
>Route: <sip:213.218.28.202;lr=on>
>Route: <sip:213.218.28.102;lr=on>
>---------------------------------------------------------------
>-----------------
>
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip