This is starting to smell like BCP material actually. 

> -----Original Message-----
> From: Paul Kyzivat [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, April 16, 2008 16:04
> To: Audet, Francois (SC100:3055)
> Cc: SIP IETF; Dan WING; Dean Willis
> Subject: Re: [Sip] E.164 - who owns it
> 
> replying to myself - another thought:
> 
> Paul Kyzivat wrote:
> > 
> > Francois Audet wrote:
> >> That's kind of what I was also saying.
> >>
> >> PS: what's the difference between 2 and 4? 
> > 
> > I think 4 is a typo and was presumably intended to be
> > 
> >     4) sip:+445588675309:a.com;user=phone
> > 
> > It is already the case that a.com may treat 3,4 & 5 alike 
> if it wishes.
> > And b.com may treate 1,2 & 5 alike if it wishes.
> > 
> > The question is whether a.com may treat 1&2 like 5, or if b.com may 
> > treat 3&4 like 5.
> > 
> > Apparently some are saying that user=phone is license to ignore the 
> > domain. But that just raises the question of why one would insert a 
> > domain along with a parameter that says to ignore it. Makes 
> no sense to me.
> 
> This would make *some* sense to me if the semantics were:
> 
> 1) you MAY route to this domain following 3263 procedures 
> with expectation that it will work.
> 
> 2) if you are unable to do that (say because you don't have a 
> sip path at all, or don't have a path to the domain in the 
> URI) then you may treat it as a tel URI and route it any way 
> that works for you.
> 
> But it seems that in many of the use cases that have been 
> presented in this thread it is actually the case that if the 
> recipient routes to the domain in the URI the request *won't* 
> work. That makes absolutely no sense to me. In that case TEL 
> seems clearly the right way.
> 
> Now Hadriel has made a case that it is the SBC's job to fix 
> this up. So when the URL exits the domain in which it will 
> work, then the SBC should fix it up with the domain it is 
> going into. I guess that could make sense if that SBC was 
> acting as an agent of *both* domains. But typically it is 
> only acting as an agent of one domain. An SBC acting for 
> b.com has no business changing the domain of the URL to 
> a.com, since doing so is (or isn't) a policy of a.com.
> 
> A way around that is to say that if the URI contains b.com, 
> then an SBC acting for b.com, when it knows it won't honor 
> that, can change it to a TEL URI when it exits the domain. It 
> then may well go through another SBC acting for a.com as it 
> enters the a.com domain. That SBC could change the TEL URI to 
> an a.com URI if that will be handled correctly within the 
> a.com domain.
> 
>       Paul
> 
> >     Paul
> > 
> >>> -----Original Message-----
> >>> From: Anders Kristensen [mailto:[EMAIL PROTECTED]
> >>> Sent: Wednesday, April 16, 2008 15:21
> >>> To: Dean Willis
> >>> Cc: Hadriel Kaplan; SIP IETF; Audet, Francois (SC100:3055); Paul 
> >>> Kyzivat; Dan WING
> >>> Subject: Re: [Sip] E.164 - who owns it
> >>>
> >>>
> >>>
> >>> Dean Willis wrote:
> >>>> What does Bob put into the Contact of a 302 he might 
> send to Alice?
> >>>>
> >>>> I've heard several alternatives:
> >>>>
> >>>> 1) sip:+445588675309:b.com
> >>>>
> >>>> 2) sip:+445588675309:b.com; user=phone
> >>>>
> >>>> 3) sip:+445588675309:a.com
> >>>>
> >>>> 4) sip:+445588675309:b.com;user=phone
> >>>>
> >>>> 5) tel:+445588675309
> >>> The presence of user=phone and a valid E.164 number in 
> the user part 
> >>> seems like a pretty strong hint so I wonder if the 
> following might 
> >>> work as a pragmatic, if not exactly elegant, way forward: allow 
> >>> proxies and the UAC to treat cases 2 and 4 as if the URI is 
> >>> equivalent to 5 in the sense that they can do ENUM query on the 
> >>> E.164 number and they can even rewrite the domain part and expect 
> >>> the URI to identify the same entity.
> >>>
> >>> This doesn't prevent a.com from recursing when really Alice's UA 
> >>> would have liked to use another provider for gateway services but 
> >>> then neither does using a tel: URI.
> >>>
> >>> We'd also want to start pushing support for tel:, of course.
> >>>
> >>> Thanks,
> >>> Anders
> >>>
> >>>> So let's step through what these mean and why they each 
> might not 
> >>>> work. Keep in mind also that there's a very real 
> possibility that 
> >>>> Alice might nod NEED a gateway to reach Jenny; it's 
> possible that 
> >>>> Jenny's phone number is already in ENUM and is directly
> >>> reachable via
> >>>> SIP. But this only works if Alice knows to look it up that way.
> >>>>
> >>>> 1) causes Alice to make a call to a user ID in the domain
> >>> of b.com.  
> >>>> Assuming that b figures out that this means a telephone
> >>> destination,
> >>>> Alice probably doesn't have an account at b.com to use for
> >>> placing the
> >>>> call. So unless B provides free SIP-to-PSTN calling, or at least 
> >>>> provides free translation service to ENUM, she's out of luck.
> >>>>
> >>>> Of course, maybe a.com knows that when it gets a 302 back with a 
> >>>> contact that looks like it might be a phone number, it
> >>> should discard
> >>>> the host part and do phone number routing on the user part. 
> >>> That works
> >>>> until a.com also has user IDs that look like phone numbers. 
> >>> Of course,
> >>>> this is a direct violation of RFC 3261, which bans a.com from 
> >>>> retargeting a request with a b.com host part.
> >>>>
> >>>> 2) is much the same as 1, except that "user=phone" 
> provides another 
> >>>> hint. B is no more likely to provide gateway services, 
> but at least 
> >>>> least if A is going to do an illegal foreign 
> retargeting, it has a 
> >>>> broader hint that phone number routing instead of user 
> routing is 
> >>>> required.
> >>>>
> >>>> 3) instructs alice to make a call in the domain of a.com. 
> >>> That's fine
> >>>> as long as sip:+445588675309:a.com routes to Jenny. Does
> >>> it? It might
> >>>> route to somebody with user ID +445588675309, which is a 
> perfectly 
> >>>> valid user ID.
> >>>>
> >>>> It might also be true that Alice doesn't use a.com for her
> >>> PSTN calls.  
> >>>> Perhaps instead, she uses c.com. So to use c.com 
> services, or do an 
> >>>> enum lookup, she has to do an illegal retargeting.
> >>>>
> >>>> 4) is much like 3, with the added hint that a telephone-number 
> >>>> destination is indicated. This might eliminate the
> >>> accidental calling
> >>>> of user +445588675309, but it does nothing to resolve the
> >>> question of
> >>>> Alice not using a.com services for PSTN gateways. There's
> >>> also an open
> >>>> issue as to whether PSTN routing (such as an ENUM lookup) can be 
> >>>> applied, vs requiring the call to traverse a gateway.
> >>>>
> >>>> At the very best, a.com (or Alice's UA) knows to discard
> >>> the host part
> >>>> and do telephone routing on the user part. If Alice's UA 
> does this, 
> >>>> it's probably a violation of RFC 3261, as the UA itself is not 
> >>>> responsible for the host-part of the contact. A proxy in
> >>> a.com could
> >>>> do this translation. But what happens if Alice doesn't use
> >>> a.com for
> >>>> routing PSTN calls? For example, I might place SIP calls using 
> >>>> "softarmor.com", but I make my PSTN calls using 
> "sipphone.com", so 
> >>>> a
> >>>> 302 sent to me for "sip: 
> >>> [EMAIL PROTECTED];user=phone" will
> >>>> most assuredly fail.  Now, if the a.com proxy knew to 
> translate the 
> >>>> call, that would be OK, but depending on the 
> relationship between 
> >>>> Alice and a.com, it might not.
> >>>>
> >>>> 5) doesn't work at all, because Alice's phone doesn't
> >>> understand tel:  
> >>>> URLs, since understanding of such was listed as a MAY in
> >>> RFC 3261. But
> >>>> it's the only suggestion from the set that doesn't require
> >>> violating a
> >>>> bunch of existing protocol rules.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> Really, in the long list of interop issues, this one's
> >>> pretty low on
> >>>>> the totem pole, imho.  People are having trouble getting
> >>> basic calls
> >>>>> to work without the aid of a middle-box.  That's a bit
> >>> higher on the
> >>>>> list to me. :)
> >>>> This is a very basic interdomain calling scenario. I'm
> >>> really hoping
> >>>> we get more and more interdomain calls.
> >>>>
> >>>>
> >>>>>>> Again, don't shoot the messenger: it makes sense to me
> >>> to use Tel
> >>>>>>> URI for this. I am just saying it may cause interop
> >>> problems. Maybe
> >>>>>>> that's ok, and maybe implementations will start 
> implementing tel 
> >>>>>>> URI.
> >>>>>> I've no doubt that there will be interop problems. That's what 
> >>>>>> happens when you specify something that has previously been 
> >>>>>> unspecified and left to whim or caprice.
> >>>>> OK, but failing calls will not give it a high chance of being 
> >>>>> adopted.  Maybe we can figure out how to get a sip: uri 
> to work as 
> >>>>> well - there's more than one way to skin a cat, even 
> with a potato 
> >>>>> peeler. ;)
> >>>> Failing calls that don't work now isn't much of a handicap. 
> >>>  Changing
> >>>> the rules of RFC 3261 to make case 3 or 4 generally valid
> >>> might also
> >>>> work.
> >>>>
> >>>> Fundamentally, we need a documented solution that works in
> >>> the general
> >>>> case and doesn't conflict with MUST level rules of other
> >>> RFCs we cite,
> >>>> unless it supercedes those RFCs.
> >>>>
> >>>>
> >>>> --
> >>>> Dean
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use 
> >>>> [EMAIL PROTECTED] for questions on 
> current sip Use 
> >>>> [EMAIL PROTECTED] for new developments on the application of sip
> >>>>
> > _______________________________________________
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> > This list is for NEW development of the core SIP Protocol Use 
> > [EMAIL PROTECTED] for questions on current sip Use 
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> > 
> 
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