AW: [Enum] RE: [Sipping] Associating SIP sessions with PSTN calls

Thank you Richard for forwarding this discussion to the ENUM list.

As you stated below, the topic has already been discussed (;-) sigh) at
the ENUM list. There is also existing my draft-stastny-enum-scenarios-00.txt
which beside other issues dealing with this problem.

Furthermore I want to point out that exactly the discribed scenario is
one of the main topics of our ENUM-trials, especially the VISIONng
ENUM-trial dealing with +87810 UPT. In the case of personal numbers
the calls are routed as described below on the PSTN in the normal
way to a gateway. In addition, a GW (either SIP or H323) exists on the
Internet. The ENUM entry contains a pointer to this gateway. Anybody 
querying ENUM may directly route the call to this gateway and the call
is terminating in the same way as if it would come from the PSTN. The 
termination could either be on an IP-based client or on the PSTN or a 
mobile terminal, if forwarded. Of course the same may happen for any
normal termination on the PSTN, ported or not, if a GW aka point-of-
interconnect (PoI) exists on the Internet, and this PoI is annonced e.g. via
ENUM.

In the draft-brandner-enumservices-compendium-00.txt the ivoice service
is introduced "as an convenient way of indicating that the resource indicated
by the associated URI is connected to an IP network ..."

The main discussion up to now is, if in addition to User-ENUM a second
tree called Infrastructure ENUM is existing, because not all numbers of
a given range may exist in user ENUM because of the opt-in principle and
the potential problems caused by a clash of responsibilities in the Tier 2.
This second tree is also proposed in draft-ietf-enum-e164-gstn-np-05.txt

Anyway, I am currently working on an I-D putting together the above
mentioned documents, draft-yu-sip-np, draft-yu-tel-url and draft-brandner-tel-svc
for a consistent application of ENUM on NP, Freephone and other routing issues.

Finally it should be possible to announce the destination (GW, PoI, PSTN route) of 
ANY phone number in ENUM, to by queried by the origination (user and/or network)
directly. 

HEALTH WARNING: If you are not aware already, this will obsolete carriers and 
transit networksand allow global number portability for any E.164 number, even for
geographic numbers.

And I fully agree to the statement below from Alan, that this is the real
application for ENUM: a globally available, scaleable, performant IN-Service.

regards
Richard Stastny

	-----Ursprüngliche Nachricht----- 
	Von: Richard Shockey [mailto:rich.shockey@NeuStar.com] 
	Gesendet: Do 19.09.2002 19:28 
	An: Alan Johnston; Michael O'Doherty; 'Tom_Gray@Mitel.COM' 
	Cc: 'sipping@ietf.org'; enum@ietf.org 
	Betreff: [Enum] RE: [Sipping] Associating SIP sessions with PSTN calls

	At 01:58 PM 9/18/2002 -0500, Alan Johnston wrote:
	>Michael,
	>
	>Tom is right - ENUM is the right solution to this.  Here is how it could work:
	>
	>If the PSTN number is dialed in the PSTN, it routes as normally and rings
	>the user's PSTN phone, which is what you want.
	>
	>If the PSTN number is dialed in a SIP network which does not query ENUM,
	>the call will be routed to a PSTN gateway and ring the user's PSTN phone,
	>which again is a success.
	>
	>If the PSTN number is dialed in a SIP network that does query ENUM, it
	>returns a SIP URI for the endpoint.  An INVITE is then sent to the proxy
	>in the domain of the SIP URI, which presumably is the service provider for
	>the user.  The proxy knows that this user wishes to receive calls on their
	>PSTN phone (for some reason), so it routes the call to a PSTN gateway and
	>rings the user's phone.
	>
	>Any SIP requests which are for other media sessions or any other purpose
	>are routed using ENUM to the proxy, which forwards them to the SIP User
	>Agent, which establishes the associated session you are after (whatever
	>that is).
	>
	>You are right, however, in noting that this sort of use is not described
	>in any of the ENUM documents - they mainly describe a whole set of
	>questionable uses for ENUM that simply don't work.  I would encourage you
	>to bring this up on the ENUM list, as this is exactly what ENUM was
	>designed to do.

	Yes please  .. as your friendly neighborhood ENUM WG -co chair I encourage
	this.

	I will cross post the document to my list ASAP.

	We've been over a lot of this ground already.

	>Thanks,
	>Alan Johnston
	>sip:alan@digitalari.com
	>
	>
	>At 07:16 PM 9/18/2002 +0100, Michael O'Doherty wrote:
	>
	>>Tom,
	>>
	>>The first option in the draft below looks at ENUM.
	>>
	>>The main problem is if the user wants to use their PSTN phone for voice
	>>calls (even from SIP phones) and their SIP device for other media
	>>sessions (like PSTN associated sessions).
	>>
	>>How do we indicate with an ENUM response that voice calls should go to
	>>the PSTN number and other media/mixed media calls to a SIP device (one
	>>suggestion here was that if ENUM returns a tel URL as one of the options
	>>for routing then that option MUST be used if the call is a voice call) ?
	>>
	>>We couldn't see any nice clean way to do this with ENUM as it stands
	>>today - option 4 in the draft (using a well know domain to do the CLI
	>>lookup) just seemed to be less messy and more obvious.
	>>
	>>Comments ?
	>>
	>>Cheers,
	>>
	>>Mick.
	>>
	>>
	>>-----Original Message-----
	>>From: Tom_Gray@Mitel.COM
	>>[<mailto:Tom_Gray@Mitel.COM>mailto:Tom_Gray@Mitel.COM]
	>>Sent: 18 September 2002 18:35
	>>To: O'Doherty, Michael [MDN05:MD01:EXCH]
	>>Cc: 'sipping@ietf.org'
	>>Subject: Re: [Sipping] Associating SIP sessions with PSTN calls
	>>
	>>
	>>
	>>Why can't the answer to this be ENUM?
	>>
	>>
	>>
	>>
	>>
	>>"Michael O'Doherty" <mdoherty@nortelnetworks.com>@ietf.org on 09/18/2002
	>>11:43:55 AM
	>>
	>>Sent by:  sipping-admin@ietf.org
	>>
	>>To:   "'sipping@ietf.org'" <sipping@ietf.org>
	>>cc:
	>>Subject:  [Sipping] Associating SIP sessions with PSTN calls
	>>
	>>
	>>
	>>Hi,
	>>
	>>The draft below describes a number of schemes to allow a SIP session be
	>>associated with a PSTN call, plus the authors choice of the best approach
	>>from the options briefly described.
	>>
	>>We'd be interested to see if anyone has any comments, in particular if
	>>there are any options we missed.
	>>
	>>Additionally, looking at this problem domain threw up a more general
	>>requirement also:
	>>
	>>  - the ability to route a SIP request differently depending on the media
	>>service that the request is requesting
	>>
	>>Do people feel this is a valid requirement in a  SIP networks and if so has
	>>anybody any thoughts on how it might be addressed ?
	>>
	>>Cheers,
	>>
	>>Mick.
	>>
	>>The draft can be retrieved form the link below until it shows up in the
	>>archives:
	>>
	>><http://ties.itu.int/~watsonm/draft-odoherty-sipping-pstn-number-association-01.txt>http://ties.itu.int/~watsonm/draft-odoherty-sipping-pstn-number-association-01.txt
	>>
	>>
	>
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	 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
	Richard Shockey, Senior Manager, Strategic Technology Initiatives
	NeuStar Inc.
	46000 Center Oak Plaza   Bldg 10    Sterling, VA  20166
	Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
	<mailto:richard@shockey.us> or <mailto:rich.shockey@neustar.biz>
	<http://shockey.us > ; <http://www.neustar.biz> ; <http://www.enum.org>
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