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hisham.khartabil | 1 Sep 2003 10:45
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RE: AW: AW: SIPIT Interop problem with ;user=phone

So, I'll ask the question again, this time for real:

Under the informative text in 2806bis labeled "A Use of "tel" URIs with SIP (Informative)", it says:

"
             2.   The outbound proxy does not use the same phone
                  context, but can route to a proxy that handles this
                  phone context. This routing can be done via a lookup
                  table or the domain name of the phone context might be
                  set up to reflect the SIP domain name of a suitable
                  proxy. For example, a proxy may always route calls
                  with tel URIs like

                  tel:1234;phone-context=munich.example.com

                  to the SIP proxy located at munich.example.com." 

So, how does the proxy route this message to munich.example.com? Or how does it discover the proxy at munich.example.com?

The reason I'm asking this again is due to my proposal in earlier emails to use a tel URI for dial strings and
sip URI for pure sip users.

Is there something wrong with mandating that if an entity placed a tel-URI in a sip request along with a
phone-context carrying a domain name, then that domain name must be the address of a sip proxy? 

/Hisham

> -----Original Message-----
> From: ext Cullen Jennings [mailto:fluffy <at> cisco.com]
> Sent: Saturday, August 30, 2003 2:48 AM
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hisham.khartabil | 1 Sep 2003 11:21
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RE: AW: AW: SIPIT Interop problem with ;user=phone

So, the answer in any case is that it is somehow routable. Therefore I conclude the allowing a sip URI to carry
a dial strings is not needed.

/Hisham

> -----Original Message-----
> From: ext Stastny Richard [mailto:Richard.Stastny <at> oefeg.at]
> Sent: Monday, September 01, 2003 12:20 PM
> To: Khartabil Hisham (NMP/Helsinki); fluffy <at> cisco.com;
> dean.willis <at> softarmor.com
> Cc: Brian.Rosen <at> marconi.com; sip <at> ietf.org
> Subject: AW: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
> 
> 
> You should not need to route this call anyway, because 
> normally you should
> not see such a context out of the context. The idea is: if 
> you know the context
> then you also know how to route the call, if you do not know 
> the context,
> you just say invalid number. In other contexts only global 
> understandable
> URIs should be used
>  
> Richard
> 
> 	-----Ursprüngliche Nachricht----- 
> 	Von: hisham.khartabil <at> nokia.com 
> [mailto:hisham.khartabil <at> nokia.com] 
> 	Gesendet: Mo 01.09.2003 10:45
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Elwell, John | 1 Sep 2003 17:38
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RE: Update of display name during a call

Cullen,

See [JRE] in-line.

John (john.elwell <at> siemens.com)

-----Original Message-----
From: Cullen Jennings [mailto:fluffy <at> cisco.com] 
Sent: 29 August 2003 16:26
To: Elwell, John; sip <at> ietf.org
Subject: Re: [Sip] Update of display name during a call

I did think about using the PAI header and it seemed to have some problems.
One is that it is only usable in walled gardens and it would be nice to have
a more generic solution than that but ignoring this, there are other
problems too. 
[JRE] I am not opposing the more generic solution. I would just like to see
PAI allowed for display update during a call in scenarios where PAI is used.

Consider a call that passes through two trust different trust domains. The
first trust domain MUST remove the PAI before it passes it to the second.
Even if it did not remove it, the second MUST ignore it. Even if both
domains failed to do this, the information being passed in the PAI is a
network assertion of identity inside a single trust domain and may not
correspond at all to the display information. For example, a trust domain
might decide to use their billing identifier for the subscriber in the PAI.
[JRE] You seem to be suggesting that PAI is not (necessarily) for an
identity that is to be displayed. However, section 8 of RFC3325 states
"Typically, a user agent renders the value of a P-Asserted-Identity header
field that it receives to its user." This suggests that PAI is indeed for
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Henry Chen | 2 Sep 2003 05:39
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RE: Update of display name during a call


Not really since a UA only expect to receive PAI, called party receives
it in a request and calling party receives it in the response. 

-----Original Message-----
From: Cullen Jennings [mailto:fluffy <at> cisco.com] 
Sent: Sunday, August 31, 2003 10:56 AM
To: Henry Chen; Elwell, John; sip <at> ietf.org
Subject: Re: [Sip] Update of display name during a call

Well the PAI is a network asserted From. You are talking about a network
asserted To. I suspect putting both in the same field with lead to
problems.

On 8/29/03 10:49, "Henry Chen" <hjlechen <at> cisco.com> wrote:

> 
> The caller's name can be authenticated and put in a PAI header by a 
> proxy in the initial Invite according to RFC3325.
> 
> The problem is how to pass called party name to the call originator 
> when the call is forwarded by the proxy.
> 
> Is there any problem to put called party id or display name in a PAI 
> header in the response to Invite by the proxy in the trusted domain 
> serving the call?
> 
> Thanks,
> 
> Henry
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Stastny Richard | 1 Sep 2003 11:19
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AW: AW: AW: SIPIT Interop problem with ;user=phone

You should not need to route this call anyway, because normally you should
not see such a context out of the context. The idea is: if you know the context
then you also know how to route the call, if you do not know the context,
you just say invalid number. In other contexts only global understandable
URIs should be used
 
Richard

	-----Ursprüngliche Nachricht----- 
	Von: hisham.khartabil <at> nokia.com [mailto:hisham.khartabil <at> nokia.com] 
	Gesendet: Mo 01.09.2003 10:45 
	An: fluffy <at> cisco.com; Stastny Richard; dean.willis <at> softarmor.com 
	Cc: Brian.Rosen <at> marconi.com; sip <at> ietf.org 
	Betreff: RE: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
	
	

	So, I'll ask the question again, this time for real:
	
	Under the informative text in 2806bis labeled "A Use of "tel" URIs with SIP (Informative)", it says:
	
	"
	             2.   The outbound proxy does not use the same phone
	                  context, but can route to a proxy that handles this
	                  phone context. This routing can be done via a lookup
	                  table or the domain name of the phone context might be
	                  set up to reflect the SIP domain name of a suitable
	                  proxy. For example, a proxy may always route calls
	                  with tel URIs like
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Rosen, Brian | 2 Sep 2003 15:38

RE: AW: AW: SIPIT Interop problem with ;user=phone

Getting more and more confused.

A "dial string" cannot go in a tel uri.
It can go in the user part of a sip uri.

So, if a sip UA does not have the ability to automatically
translate dial strings to routable telephone numbers, it has
to be in a sip uri user part.  I don't think that is
controversial.

The question is, does the UA need to mark the user part as
a dial string, or is it enough for the host mentioned in the
domain part of the sip uri to just know the difference?
I'd prefer to mark the userpart as a dial string, but I can
cope if we don't.

Brian

> -----Original Message-----
> From: hisham.khartabil <at> nokia.com [mailto:hisham.khartabil <at> nokia.com]
> Sent: Monday, September 01, 2003 5:21 AM
> To: Richard.Stastny <at> oefeg.at; fluffy <at> cisco.com;
> dean.willis <at> softarmor.com
> Cc: Brian.Rosen <at> marconi.com; sip <at> ietf.org
> Subject: RE: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
> 
> 
> So, the answer in any case is that it is somehow routable. 
> Therefore I conclude the allowing a sip URI to carry a dial 
> strings is not needed.
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xiao'tong liang | 2 Sep 2003 03:32
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Re: Re: the meaning of the user agent field in SIP

User Agent in SIP network always means terminals(call
originated or terminated entity). Terminals can appear
in many kinds: the inner UAs in gateway, the inner UAs
in server(such as Voice mail server), also can be a UA
in SIP Phone.

--
Liang xiaotong
newrocktech.com Shanghai

--- Anu Gupta <theanug <at> yahoo.com> wrote:
> Hi All,
> I am new to SIP and I need the meaning of the
> user_agent field in SIP. If anyone knows kindly
> reply.
> Thanks,
> Anu.
> 

__________________________________
Do you Yahoo!?
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors <at> cs.columbia.edu for questions on current sip
Use sipping <at> ietf.org for new developments on the application of sip
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Kolinowitz Bernd | 2 Sep 2003 15:49
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Sending an announcement!!


Hi!

We want to implement a typical PSTN feature in our SIP proxy. If caller A sends an INVITE (including audio in
the sdp) to the sip-proxy, the proxy should send an announcement (like a jingle) to caller A while it is
trying to reach the callee.

We are now searching for a way to implement the feature concerning RFC 3264 (offer/answer model of SDP).
First we thought to answer with early media, that means to send a 183 response including the SDP for the
announcement and demanding a PRACK request.
Next we have thought to forward the 200 response from the callee to the caller. 
Now there is our problem: What should the caller do with the SDP?

The good version: He accept the SDP from the callee as an answer of his SDP offer in the first INVITE-request
and sends an empty ACK.
The bad version: The SDP in the 200 response will be treated as a new offer and he sends an SDP answer in the ACK request.

How will the client react? Is there another way to realize this feature?

Thanks in advance
Bernd

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors <at> cs.columbia.edu for questions on current sip
Use sipping <at> ietf.org for new developments on the application of sip
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hisham.khartabil | 2 Sep 2003 16:37
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RE: AW: AW: SIPIT Interop problem with ;user=phone



> -----Original Message-----
> From: ext Rosen, Brian [mailto:Brian.Rosen <at> marconi.com]
> Sent: Tuesday, September 02, 2003 4:38 PM
> To: Khartabil Hisham (NMP/Helsinki); Richard.Stastny <at> oefeg.at;
> fluffy <at> cisco.com; dean.willis <at> softarmor.com
> Cc: sip <at> ietf.org
> Subject: RE: AW: AW: [Sip] SIPIT Interop problem with ;user=phone
> 
> 
> Getting more and more confused.
> 
> A "dial string" cannot go in a tel uri.
> It can go in the user part of a sip uri.
> 
> So, if a sip UA does not have the ability to automatically
> translate dial strings to routable telephone numbers, it has
> to be in a sip uri user part.  I don't think that is
> controversial.

I can hand a Nokia colleague a business card that has the following info:

Tel: 04012345
SIP: 04067891 <at> nokia.com

He dials +3584012345. How does the UA know if it should append  <at> nokia.com and make it a sip URI or just make it a
tel URI?

Options (if the user indicated its choice)
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Rosen, Brian | 2 Sep 2003 17:01

RE: AW: AW: SIPIT Interop problem with ;user=phone

> He dials +3584012345. How does the UA know if it should 
> append  <at> nokia.com and make it a sip URI or just make it a tel URI?
By definition, if the UA does not know how to translate
dial strings to phone numbers, it MUST use a sip uri.  There
is no option to use a tel uri.

I think your example not ideal only because a UA could
have a trivial rule that a number starting with + is known to
be a globally dialable number, and thus COULD be put in a tel
uri.  However, I would hope that phones did not do a lot of
guessing; if they have knowledge of the local dialing plan and
special dialing options, then they should interpret the dial
string and do the right thing.  If they don't they should put
the dial string in a sip uri and let a proxy figure it out.

> 
> Options (if the user indicated its choice)
> - The dialled digits placed in a tel URI with a phone-context.
This cannot work because the dial string may not be a phone number,
and thus this would not be in conformance with 2806bis.

> - If the UA does not have a routing table and cannot place a 
> phone-context, then it needs to place the dialled digits in a 
> sip URI. But I believe it should indicate the user as a dial 
> string (as you suggest). 
I agree.
....snip
> 
> - If the user dials county code and area code (3584012345), 
> the dialled digits placed in a tel URI with no phone-context.
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