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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.
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@nokia.com [mailto:hisham.khartabil@nokia.com]
> Sent: Monday, September 01, 2003 5:21 AM
> To: Richard.Stastny@oefeg.at; fluffy@cisco.com;
> dean.willis@softarmor.com
> Cc: Brian.Rosen@marconi.com; sip@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.
>
> /Hisham
>
> > -----Original Message-----
> > From: ext Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Monday, September 01, 2003 12:20 PM
> > To: Khartabil Hisham (NMP/Helsinki); fluffy@cisco.com;
> > dean.willis@softarmor.com
> > Cc: Brian.Rosen@marconi.com; sip@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@nokia.com
> > [mailto:hisham.khartabil@nokia.com]
> > Gesendet: Mo 01.09.2003 10:45
> > An: fluffy@cisco.com; Stastny Richard;
> > dean.willis@softarmor.com
> > Cc: Brian.Rosen@marconi.com; sip@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
> >
> > 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@cisco.com]
> > > Sent: Saturday, August 30, 2003 2:48 AM
> > > To: Stastny Richard; Khartabil Hisham (NMP/Helsinki);
> > Dean Willis
> > > Cc: Brian Rosen; sip@ietf.org
> > > Subject: Re: AW: AW: [Sip] SIPIT Interop problem with
> > ;user=phone
> > >
> > >
> > >
> > > I don't know what I was thinking - I gave the wrong answer.
> > > The context in
> > > the tel does not at mean you should attempt to route the call
> > > to there - it
> > > is just an identifier of the namespace of who
> > understands the number.
> > >
> > > On 8/29/03 13:50, "Stastny Richard"
> > <Richard.Stastny@oefeg.at> wrote:
> > >
> > > >>> So, how does the proxy route this message to
> > munich.example.com?
> > > >>>
> > > >>> /Hisham
> > > >>>
> > > >>
> > > >> The same way it routes sip:fluffy@munich.example.com
> > > >
> > > > Tell that to Dean :) He thinks it is not routable (that's
> > > the point I was
> > > > trying to make).
> > > >
> > > > Warning: this needs NOT to be true and IMHO this part of
> > > > the informational section is misleading. In section 5.1.4
> > > Local numbers
> > > > it is clearly stated:
> > > >
> > > > "There are two ways to label the context: via a global
> > > number or any |
> > > > number of its leading digits (e.g., "+33") and via a
> > > domain name, |
> > > > e.g., "houston.example.com". The choice between the two is
> > > left to |
> > > > the "owner" of the local number and is governed by whether
> > > there is a |
> > > > global number or domain name that is a valid identifier
> > > for a |
> > > > particular local number.
> > > > |
> > > > The domain name does not have to resolve to any actual
> > > host, but MUST |
> > > > be under the administrative control of the entity managing
> > > the local |
> > > > phone context. "
> > > >
> > > > Since the domain name does not have to resolve to
> > an actual host,
> > > > one cannot rely on this, so your statement may be wrong.
> > > >
> > > > Richard
> > > >
> > > > -----Ursprüngliche Nachricht-----
> > > > Von: hisham.khartabil@nokia.com
> > [mailto:hisham.khartabil@nokia.com]
> > > > Gesendet: Fr 29.08.2003 17:31
> > > > An: fluffy@cisco.com; dean.willis@softarmor.com
> > > > Cc: Brian.Rosen@marconi.com; sip@ietf.org
> > > > Betreff: RE: AW: [Sip] SIPIT Interop problem with
> > ;user=phone
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >> -----Original Message-----
> > > >> From: ext Cullen Jennings [mailto:fluffy@cisco.com]
> > > >> Sent: Friday, August 29, 2003 5:51 PM
> > > >> To: Khartabil Hisham (NMP/Helsinki); Dean Willis
> > > >> Cc: Brian Rosen; sip@ietf.org
> > > >> Subject: Re: AW: [Sip] SIPIT Interop problem with
> > ;user=phone
> > > >>
> > > >>
> > > >> On 8/29/03 1:03, "hisham.khartabil@nokia.com"
> > > >> <hisham.khartabil@nokia.com>
> > > >> wrote:
> > > >>
> > > >>> 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?
> > > >>>
> > > >>> /Hisham
> > > >>>
> > > >>
> > > >> The same way it routes sip:fluffy@munich.example.com
> > > >
> > > > Tell that to Dean :) He thinks it is not routable (that's
> > > the point I was
> > > > trying to make).
> > > >
> > > > Regards,
> > > > Hisham
> > > >
> > > >
> > > >>
> > > >>
> > > >>
> > > >
> > > > _______________________________________________
> > > > 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
> > >
> >
> >
> >
> J*fj)b bm? >
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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@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip