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AW: [Partysip-dev] 481 after 183, CANCEL


From: Walter Schober
Subject: AW: [Partysip-dev] 481 after 183, CANCEL
Date: Mon, 29 Sep 2003 16:42:30 +0200

Hello Aymeric!

The INVITE and CANCEL are matched on the branch= tag in the Via headers
in pspm_sfp_cancel_match_invite(). But the branches differ.

Since branches are a hash of Request-URI, To, From, Call-ID and the
Request-URI is different per se, the question now is: Is the branch
calculation wrong or is branch the wrong method of matching call legs?

I will ask that the other proxy vendor as well (comercial proxy) :-)

Walter

------ logs ---- logs ----

These are the requests coming in from the other proxy and routed to the
destination:
(stuff deleted)
| ERROR | <sfp.c: 714> sfp.c: Hello World! (4)   -> in the /* compliant
UA   :) */ section :-)
'z9hG4bKQy4rx9DQ~63Nm3`'
'z9hG4bKqsh0.qG~VDJR.FZ'

Incoming INVITE:
INVITE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKQy4rx9DQ~63Nm3`
Via: SIP/2.0/UDP
192.168.3.230;branch=z9hG4bKnp701596867-43cf12fe192.168.3.230
To: sip:address@hidden
From: Walter Schober <sip:address@hidden>;tag=29d184cd
Call-ID: address@hidden
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13

Outgoing INVITE (does partysip add the <> here in the To:?):
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP
192.168.3.2:5060;branch=z9hG4bKf6de09f2d465eec40f531b245b8f094da.0
Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKQy4rx9DQ~63Nm3`
Via: SIP/2.0/UDP
192.168.3.230;branch=z9hG4bKnp701596867-43cf12fe192.168.3.230
From: Walter Schober <sip:address@hidden>;tag=29d184cd
To: <sip:address@hidden>
Call-ID: address@hidden
user-agent: Ahead SIPPS IP Phone Version 2.0.42.13

Incoming CANCEL:
CANCEL sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKqsh0.qG~VDJR.FZ
To: sip:address@hidden
From: Walter Schober <sip:address@hidden>;tag=29d184cd
Call-ID: address@hidden
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13





-----Ursprüngliche Nachricht-----
Von: Aymeric Moizard [mailto:address@hidden 
Gesendet: Freitag, 26. September 2003 11:46
An: Walter Schober
Cc: 'This is the partysip mailing list'
Betreff: RE: [Partysip-dev] 481 after 183, CANCEL




On Thu, 25 Sep 2003, Walter Schober wrote:

> Hi!
>
> Found the problem - sorry, I'm not being so long with SIP - CANCEL 
> sent To: without <>
>
> INVITE:
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > To: <sip:address@hidden>
> > Call-ID: address@hidden
>
> CANCEL:
> > To: sip:address@hidden
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > Call-ID: address@hidden
>
> And since a Call Leg = To: + From: + Call-ID:.

Strange behavior? I'm not sure such a CANCEL should fail!

Have you located the code that reject this?
Aymeric

> BTW: Good book IMHO: Alan B. Jhonston, "Understanding the Session 
> INitialted Protocol".
>
> W.
>
> -----Original Message-----
> From: address@hidden
> [mailto:address@hidden On 
> Behalf Of Walter Schober
> Sent: Thursday, September 25, 2003 6:48 PM
> To: 'Aymeric Moizard'
> Cc: 'This is the partysip mailing list'
> Subject: AW: [Partysip-dev] 481 after 183, CANCEL
>
>
> Hi Aymeric!
>
> This was already on. Double checked this. And since I have the same 
> problem now with my X-Ten Client on XP, this might be not Cisco 
> related. Which check for no tag is this?
>
> Can you give me a hint, which part of the signalling causes this 481? 
> E.g. why this reply is generated? How do you match INVITE-CANCEL?
>
> Thanks!
> Walter
>
> -----Ursprüngliche Nachricht-----
> Von: Aymeric Moizard [mailto:address@hidden
> Gesendet: Donnerstag, 25. September 2003 18:34
> An: Walter Schober
> Cc: 'This is the partysip mailing list'
> Betreff: Re: [Partysip-dev] 481 after 183, CANCEL
>
>
>
> If you look at the partysip configuration, it says:
>
> # With Cisco ATA (version v2.15 ata18x)
> # More generally, if you can't CANCEL calls, try to uncomment this:
>
> #disable_check_for_to_tag_in_cancel = on
>
> Try to enable this!
> Aymeric
>
> On Thu, 25 Sep 2003, Walter Schober wrote:
>
> > Hi there!
> >
> > Another thing that makes me wondering:
> > Setup: (A) SIPPS/X-Ten - ProxyA - PartySIP - Cisco 1700 (B)
> >
> > A invites B, B answers with 183 Session in progress (but is RINGING,
> > Grrr.)
> > A cancels B, PartySIP answers with 481 Call Leg down not exist.
> >
> > Any idea, why?
> >
> > Below the INVITE, 183, CANCEL, 481
> >
> > Walter
> >
> > --------------------------------------------------------------------
> > --
> > --
> > --
> >
> > INVITE sip:address@hidden SIP/2.0
> > Via: SIP/2.0/UDP 
> > 192.168.3.2:5060;branch=z9hG4bKf749f8bde9fb401ba6c4ba33e4b7b8f25.0
> > Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKx6OJcm9nOd8lsSs
> > Via: SIP/2.0/UDP 
> > 192.168.3.209;branch=z9hG4bKnp269796522-411b5598192.168.3.209
> > Record-Route: 
> > <sip:192.168.3.2:5060;lr;psp=f4242eaacf0b1348e1e49697805f77c4>
> > Record-Route: <sip:192.168.3.34:5060;lr>
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > To: <sip:address@hidden>
> > Call-ID: address@hidden
> > CSeq: 1 INVITE
> > Contact: <sip:address@hidden>
> > max-forwards: 69
> > expires: 180
> > user-agent: Ahead SIPPS IP Phone Version 2.0.42.13
> > Content-Type: application/sdp
> > Accept: application/sdp
> > Content-Length:   245
> >
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP 
> > 192.168.3.2:5060;branch=z9hG4bKf749f8bde9fb401ba6c4ba33e4b7b8f25.0,S
> > IP
> > /2
> > .0/UDP
> > 192.168.3.34:5060;branch=z9hG4bKx6OJcm9nOd8lsSs,SIP/2.0/UDP
> > 192.168.3.209;branch=z9hG4bKnp269796522-411b5598192.168.3.209
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > To: <sip:address@hidden>
> > Date: Mon, 01 Mar 1993 04:35:08 GMT
> > Call-ID: address@hidden
> > Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
> > CSeq: 1 INVITE
> > Content-Type: application/sdp
> > Session: Media
> > Content-Length: 134
> >
> > CANCEL sip:address@hidden:5060 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKNr2L_jXdCBFnvs!
> > To: sip:address@hidden
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > Call-ID: address@hidden
> > CSeq: 1 CANCEL
> > Max-Forwards: 70
> > Contact: sip:address@hidden
> > Content-Type: application/sdp
> > Expires: 180
> > Record-Route:  <sip:192.168.3.34:5060;lr>
> > User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
> > Accept: application/sdp
> >
> > |WARNING| <sfp.c: 753> module sfp: No valid INVITE found; answer to
> > request with 481.!
> > | INFO2 | <psp_osip.c: 70> osip module: wake up!
> > | INFO1 | <udp.c: 816>
> > SIP/2.0 481 Call Leg/Transaction Does Not Exist
> > Via: SIP/2.0/UDP 192.168.3.34:5060;branch=z9hG4bKNr2L_jXdCBFnvs!
> > From: petzi09 <sip:address@hidden>;tag=1014c496
> > To: <sip:address@hidden>
> > Call-ID: address@hidden
> > CSeq: 1 CANCEL
> > Server: partysip/0.6.0
> > Content-Length: 0
> >
> >
> >
>
>
>
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