Subject: Re: [Sip] Re: [Enum] SIP AoR, tel and ENUM - Second Try

From: "Stastny Richard" <[email-masked]>
Date: Fri, 9 Dec 2005 17:15:56 +0100

My take on this:
 
A user enters a phone number, the UA sends it by default
to his provider A with
sip:[email-masked];user=phone
 
proxy A of Provider A interprets this as an E.164 number and
does whatever he is programmed to do with this:
 
1. forward it to his own gateway sip:[email-masked];user=phone
2. forward it to another proxy with a gateway sip:[email-masked];user=phone
3. query ENUM
in this case he may either get back sip:[email-masked]
or
sip:[email-masked];user=phone
or
sip:[email-masked];user=phone
 
Richard

________________________________

Von: [email-masked] im Auftrag von Dale R. Worley
Gesendet: Fr 09.12.2005 16:51
An: Sip
Betreff: Re: [Sip] Re: [Enum] SIP AoR, tel and ENUM - Second Try

On Fri, 2005-12-09 at 10:28 -0500, Paul Kyzivat wrote:
> > if UA is to send a request to sip:[email-masked], it MUST send it to
> > provA.com.  in case of sip:[email-masked];user=phone, it can (and
> > usually MUST) send it wherever it is sending requests to e.164 numbers,
> > i.e., where it would send tel:+4312345, because the UA may not have any
> > deal to use provA.com as its outbound proxy.
>
> I disagree with you Juha. AFAIK the routing of *either* form MUST be
> routed solely based on the provA.com domain. Unless you are responsible
> for the domain in question you are not to interpret the user part in any
> way. There is no dispensation to violate this when the user=phone
> parameter is present.

RFC 3273 seems to me to come down very strongly in favor of Paul's
position.  The only escape would be a later RFC that specifically
authorized an agent that is not authoritative for provA.com to interpret
the URI differently.

Dale

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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 [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip

---

From: Juha Heinanen <[email-masked]>
Date: Fri, 9 Dec 2005 18:58:30 +0200

> sip:[email-masked];user=phone

that is fine if the user is placing the call, but how about when the
user wants to refer the calling party with 302 to his number on pstn
network?  

last time i checked, phones from paul's company use sip uri format,
which according to paul's interpretation must be routed to whatever host
part is listed there, which for sure get blocked by the proxy if the
caller belongs to a different domain.

-- juha

---

From: "Dale R. Worley" <[email-masked]>
Date: Fri, 09 Dec 2005 12:57:35 -0500

> user wants to refer the calling party with 302 to his number on pstn

It seems to me that if the referrer wants the calling party to try to
get to a PSTN number with the best routing as seen from the calling
party's UA, it should send a tel: URI.  If a UA cannot send a tel: URI,
that is a deficiency in the UA.

This seems to be the same sort of problem as the dichotomy between a
host name and the IP address of one of its network interfaces -- if you
mean a specific interface (and hence specific network path), use the IP
address; if you mean to permit any route to the host, use its name.

Dale

---

From: Paul Kyzivat <[email-masked]>
Date: Fri, 09 Dec 2005 13:43:49 -0500

> sip:[email-masked];user=phone

The key here is that the phone is *creating* a uri in this domain. To do 
so it must be considered part of the domain in order to interpret the 
user part of the uri. It then sends it to a server that is also 
permitted to interpret the user part.

It *could* also just use tel:+4312345. In that case anybody can 
interpret the number because its meaning is defined by a standard.

> 2. forward it to another proxy with a gateway sip:[email-masked];user=phone

In other words, if it wants to pass this to another domain using the sip 
form, and wants the other domain to interpret it as a phone number, then 
it needs to rewrite the URI to a form that the receiving network uses to 
represent phone numbers. Knowing if/how/when this can be done requires 
some kind of agreement (implicit or explicit) between the source network 
and the destination network.

Again, this *could* also be done by rewriting as a tel uri before 
passing to the other network.

	Paul

---

From: Paul Kyzivat <[email-masked]>
Date: Fri, 09 Dec 2005 13:49:18 -0500

> last time i checked, phones from paul's company use sip uri format,

yes, I know they do. Maybe someday I will prevail. :-(

> caller belongs to a different domain.

There is no requirement for the UA to deliver the request *directly* to 
that address. It is perfectly free to put this request into the R-URI 
and then deliver the request to its own proxy. Then that proxy can deal 
with the routing.

	Paul

---

From: Klaus Darilion <[email-masked]>
Date: Mon, 12 Dec 2005 11:11:16 +0100

Hi!

As we all see, there are several problems which can be avoided using tel 
URIs instead of SIP URIs. The problem is: I've never seen any 
implementation yet using/supporting tel URIs.

Does somebody know SIP clients, proxies, gateways which do support tel URIs?

regards
klaus