> Is one correct and one not or does it not 
> matter as long as they both match?

RFC 3261 defines the user-param within a sip-uri.  It also defines how
to build responses; the response shown is not built correctly.

For completeness... it would also be valid to place user=phone as a
to-param within the INVITE; however the meaning and usage has not yet
been defined or registered with IANA.

For completeness... RFC 3699 deprecated RFC 2806.  The INVITE To's usage
of user=phone is valid per RFC 2806 which allowed "200" without a
phone-context.  It is not valid per RFC 3966; however many devices still
allow it for interoperability reasons.


> On 5 Jan 2009, at 22:46, "Brett Tate" <[email protected]> wrote:
> 
> > There is a user=phone issue within the 180's To header.  It 
> is inside 
> > the brackets for INVITE; it is outside for 180.
> >
> >> -----Original Message-----
> >> From: [email protected]
> >> [mailto:[email protected]] On 
> Behalf Of 
> >> Andrew Wood
> >> Sent: Monday, January 05, 2009 5:39 PM
> >> To: [email protected]
> >> Subject: [Sip-implementors] 180 being ignored by phone
> >>
> >> Im trying to implement a simple forking proxy server.
> >> In the example below the calling phone (200) is on
> >> 192.168.254.1 The called phone (201) is on 192.168.254.2 
> The proxy is 
> >> at 192.168.254.254
> >>
> >> Following is the Invite received by the proxy from the caller, and 
> >> the 180 which the proxy sends back to the caller.
> >> The phone rings, and the 180 is modified and forwarded 
> back but  the 
> >> caller doesnt get a ring tone.
> >> Ive checked them over and over and cant see whats wrong with them.
> >>
> >> Am I overlooking something here?
> >>
> >> Thanks for any help
> >>
> >> Andrew
> >>
> >>
> >> INVITE sip:[email protected];user=phone SIP/2.0
> >> Via: SIP/2.0/UDP 192.168.254.1:5060;branch=z9hG4bK1db88668c5d780ae
> >> Contact: <sip:[email protected]:5060;user=phone;transport=udp>
> >> To: <sip:[email protected];user=phone>
> >> From: <sip:[email protected];user=phone>;tag=4260414991
> >> Call-ID: [email protected]
> >> CSeq: 2 INVITE
> >> Expires: 300
> >> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, 
> >> PRACK, UPDATE
> >> Content-Type: application/sdp
> >> Proxy-Authorization: Digest
> >> username="userA",realm="",nonce="1231192060:64976ad5abf3f2b1e8
> >> f9d72b7c7e71b9",uri="sip:[email protected]",response="b6e1d9
> >> faa7b9dbb897250f531c5cc092",qop=auth-int,nc=00000001,cnonce="f
> >> 9919a53"
> >>
> >> Supported: replaces, 100rel
> >> User-Agent: Cisco-CP7912/8.0.0-060111A
> >> Content-Length: 286
> >>
> >> v=0
> >> o=200 10340144 10340144 IN IP4 192.168.254.1 s=Cisco 7912 SIP Call 
> >> c=IN IP4 192.168.254.1 t=0 0 m=audio 16384 RTP/AVP 0 18
> >> 8 101 a=rtpmap:0 PCMU/8000/1
> >> a=rtpmap:18 G729/8000/1
> >> a=fmtp:18 annexb=yes
> >> a=rtpmap:8 PCMA/8000/1
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-15
> >>
> >> ==============================================================
> >> ===================
> >>
> >>
> >> SIP/2.0 180 Ringing
> >> Via: SIP/2.0/UDP 192.168.254.1:5060;branch=z9hG4bK1db88668c5d780ae
> >> Contact: <sip:[email protected]:5060;user=phone;transport=udp>
> >> To: <sip:[email protected]:5060>;user=phone;tag=3440312132
> >> From: <sip:[email protected];user=phone>;tag=4260414991
> >> Call-ID: [email protected]
> >> CSeq: 2 INVITE
> >> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, 
> >> PRACK, UPDATE
> >> Server: Cisco-CP7912/8.0.0-060111A
> >> Content-Length: 0
> >>
> >> _______________________________________________
> >> Sip-implementors mailing list
> >> [email protected]
> >> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> >>
> >
> > 
> ______________________________________________________________________
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