[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

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
	>
	

f)+-i
bzj)fjb?*S+%b+4Ez U,zȩjezgr(雉+笶*'ܺޞ"K*ix"zwuޖfz{l{ayi'*'"