Subject: [Sipping] RE: calling number blocking at sip/isup gateway

From: "Tom-PT Taylor"<[email-masked]>
Date: Tue, 29 Jan 2002 10:41:02 -0500

Mike, I'm following this discussion with great interest because I am heavily
involved in importing the SIP-T work into ITU-T Study Group 11.  I'm
concerned with one assumption you make below -- an assumption many of us
make much of the time.  That is that the UAC calling into the PSTN has a
telephone number which it places into the From: header.  What about UACs
which do not have telephone numbers assigned?  Will they be barred from PSTN
access, or should we mandate that the gateway assign some sort of number
which it alone can cross-correlate to the original user@domain if need be
for malicious call trace, etc.?

Tom Taylor
[email-masked]
Ph. +1 613 736 0961 (ESN 396 1490)

-----Original Message-----
From: [email-masked] [mailto:[email-masked]]
Sent: Tuesday, January 29, 2002 8:28 AM
To: [email-masked]; [email-masked]
Subject: Re: [Sip] calling number blocking at sip/isup gateway

In a message dated 1/27/02 4:56:48 PM Eastern Standard Time, 
[email-masked] writes:

> What a long thread.

> etc.

Jon,

Great summary of what is needed and what is not. There have been a lot of 
protocol hacks mentioned in the last week, including putting a "bogus url"
in 
the From header or having the gateway replace the real calling party number 
with some "gateway number". Also there have been suggestions that, somehow, 
encryption was part of the solution for privacy of calling party ID and that

a "coded secret and a decoder ring" could be used. But encryption would only

address the issue of preventing someone in the middle from seeing the ID, 
which is not the issue here. In the general case, we're talking about a 
calling line ID capability for which the calling and called party have 
nothing in common, certainly not a shared secret.

I agree, there is certainly the case in which there is simply no Calling 
party number in the signaling (e.g., From=anonymous in the INVITE) when one 
element doesn't trust the next to preserve privacy. But there is the other 
case in which it trusts the next network element and includes the ID (maybe 
because the law or the network policy requires it) and it also includes an 
indicator of what privacy procedures apply. If every network portion is 
trusted, the ID can be passed to the final "switch" which then doesn't 
deliver it to the called party. So just because you get "unavailable" on
your 
Caller-ID screen doesn't mean that it wasn't in the IAM, but it could. Of 
course, the problem for SIP is the possible absence of any trusted device 
that functions as the final "switch".

However, because of other security concerns, a real IP telephone service 
provider would require that it knows the identity of the calling party and 
the model for this service must be based on a presumption that the calling 
party can trust the service provider. Therefore, in this case, the From must

contain a correct number, and the gateway may remove it (after verifying
it).

The case of the calling party leaving out the From can not be used in a 
"tariffed" (if I can use that term) telephone service. While the SIP
protocol 
may allow it to be left out, the service provider may require it before 
allowing call setup.

I think this whole discussion should be taken to the SIPPING list until
there 
is an agreement on the functional requirements.

Mike

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Use [email-masked] for questions on current sip
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---

From: Juha Heinanen <[email-masked]>
Date: Tue, 29 Jan 2002 18:17:28 +0200

Tom-PT Taylor writes:

 > What about UACs

yes, either way.  in finland we are not allowed to gateway any call to
pstn unless it has a valid from number (either of the user or the
gateway) and if we place the gw's number there, we must be able to
answer the question, which of the gw's subscribers actually made the
call.  in addition, the law requires the gw to mark the presentation of
the from number of all calls from the subscriber as "restricted" if
asked so by the subscriber.

if sip doesn't have means to implement the above, then there is no
gatewaying of sip calls to pstn in finland and there is no business case
for us until sip starts to take over the world, which will take a few
years at least.  all i want is for a proxy a way to mark the from number
of the invite as "restricted" without touching the from header AND i
want the chosen way to be clearly documented in the sip-isup i-d so that
i don't have to tell each sip/ss7/proxy vendor separately, how it is
done.

what method you choose to do it, is not my business, since i'm just
a stupid user of the sip protocol.

-- juha

---

From: Jonathan Rosenberg <[email-masked]>
Date: Wed, 30 Jan 2002 02:45:40 -0500

> i don't have to tell each sip/ss7/proxy vendor separately, how it is

Well, we have a documented way for that in the privacy spec right now,
which is for the proxy to insert the RPID-Privacy header indicating
calling party, with any kind of privacy, and type of subscriber. 

Of course, if the call doesn't terminate on PSTN, you have no privacy,
since the UA inserted its address in the From field, which won't be
stripped. You can fix that too with a b2bua, but you have indicated
above you don't want to do that. Well, you cant have your cake and eat
it too for this case.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
[email-masked]                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

---

From: Juha Heinanen <[email-masked]>
Date: Wed, 30 Jan 2002 10:28:15 +0200

> calling party, with any kind of privacy, and type of subscriber. 

yes, and i'm prefectly happy with it.  the problem is that the sip-isup
i-d doesn't say a word about it and it is that document that the ss7/sip
gw vendors will read.

-- juha