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Thread: Record-Route-Header ftag-field and routing
Record-Route-Header ftag-field and routing
country flaguser name
Germany		 2007-12-18 12:03:50 
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=z9hG4bK59f8248d2c568739bc7dc8a7de9c4f8
2
Via: SIP/2.0/UDP
212.227.15.232;branch=z9hG4bK589.5362a815.0
Via: SIP/2.0/UDP
212.227.15.197;branch=z9hG4bK202ed21ce8126c48fd64f9f8458a705
d
Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 
;received=62.206.6.140;branch=z9hG4bKterm-2ab0b6-49615195141
6-496214255114
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: 00E0F510055C3A11C0C5000025B9213.218.28.164
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>
------------------------------------------------------------
--------------------




_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementorscs.columbia.edu for questions on current
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sip
RE: Record-Route-Header ftag-field and routing
country flaguser name
United States		 2007-12-18 13:10:11 
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:mfdafuer.com] 
>Sent: Tuesday, December 18, 2007 1:04 PM
>To: sipietf.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=z9hG4bK59f8248d2c568739bc7dc8a7de9
c4f82
>Via: SIP/2.0/UDP
212.227.15.232;branch=z9hG4bK589.5362a815.0
>Via: SIP/2.0/UDP 
>212.227.15.197;branch=z9hG4bK202ed21ce8126c48fd64f9f8458
a705d
>Via: SIP/2.0/UDP sipgw01.bmcag.com:5060
>;received=62.206.6.140;branch=z9hG4bKterm-2ab0b6-4961519
51416-4
>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: 00E0F510055C3A11C0C5000025B9213.218.28.164
>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://ww
w1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP
Protocol Use 
>sip-implementorscs.columbia.edu for questions on current
sip 
>Use sippingietf.org for new developments on the
application of sip
>


_______________________________________________
Sip mailing list  https://ww
w1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementorscs.columbia.edu for questions on current
sip
Use sippingietf.org for new developments on the application of
sip
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