In step 1, if the phone does not do dialplan interpretation, then what the
user entered is a dialstring, and not a telephone number. This could be
encoded, as you show, as a SIP uri, but might be better encoded as a
dialstring, per draft-rosen-iptel-dialstring-02.txt. A tel uri is
explicitly NOT used to carry a dialstring.
I think it would be better labeled as a dialstring, and not something that
could be confused as a telephone number. It remains true that the user-part
of sip:5056416 at my-voip-provider.at can only be interpreted by the
my-voip-provider.at domain, so your flow definitely can work.
Brian
-----Original Message-----
From: enum-bounces at ietf.org [mailto:enum-bounces at ietf.org] On Behalf Of
Otmar Lendl
Sent: Thursday, August 18, 2005 6:33 AM
To: voipeer at lists.uoregon.edu; geopriv at ietf.org; enum at ietf.org
Subject: [Enum] Re: [voipeer] Re: [Geopriv] Re: [Simple] tel URIs in
commonpolicy
On 2005/08/18 05:08, Jonathan Rosenberg <jdrosen at cisco.com> wrote:
My point is that I think it makes sense to consider the tel URI a URN,
and that it is merely an accident of history that it wasn't a URN more
properly. Now, as you and I both know phone numbers in the PSTN are
abused to represent lots of things, but there is no reason to carry
forward this confusion into voip. This is why I am proposing that when a
phone number is in a tel URI, it represents a name. We don't know where
it is on the network (indeed even if its on an IP network). To know
that, we translate to an address. That address is a SIP URI. That SIP
URI can contain a phone number, i.e.
sip:+19739525000 at provider.net;user=phone, however in this format the
phone number has been resolved to an address. The act of porting a
number is a change in the translation of the phone number as a name (the
tel URI) to the phone number as an address (the SIP URI).
This is a very sensible notion.
Based on this thinking the dialing of a number on a VoIP-phone
goes through the following conceptual stages:
1) User enters a (potentially partial) number on his phone.
The phone appends its default domain and sends the invite to its proxy:
e.g. sip:5056416 at my-voip-provider.at
2) The SIP proxy applies the local dialplan to translate the
SIP address to an E.164 number:
e.g. customer is in vienna, thus 5056416 maps to +43 1 5056416
-> We now have a URN: tel:+4315056416
3) The SIP proxy now tries to route the call. In this example,
user ENUM finds:
"E2U+sip" "!^.*$!sip:office at enum.at!"
or it could map to the local PSTN gateway with an URI like
sip:+4315056416 at AS5300.my-voip-provider.at
/ol