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Date: Mon, 01 Apr 2002 14:03:22 +0550
From: vipin agrawal <poorva_agrawal@usa.net>
To: <sip@ietf.org>
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Subject: [Sip] Problems in compiling the osip source code
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hi firends,
i have  pproblems in compliling the source code of osip .

i am using windows platform ...
can anyone help in fixing this problem...

c:\documents and
settings\vipin\desktop\libosip-0.7.8.tar\libosip-0.7.8\example\msg_register.c(7)
:

 fatal error C1083: Cannot open include file: 'strings.h': No such file or
directory


please help

thanx

Regards

Vipin


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'William Marshall'" <wtm@research.att.com>
Cc: sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
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A few more comments below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: Friday, March 29, 2002 9:23 PM
> To: jon.peterson@neustar.biz
> Cc: sip@ietf.org
> Subject: RE: [Sip] Comment, SIP Privacy draft
> 
> 
> > Specifically, I am concerned with RPID usage as a replacement for the To
and
> > From headers, in architectures which have proposed that To and From
should
> > always contain random/anonymous information for all calls (not just when
> > privacy is needed) and RPID alone should be used to assert originating
and
> > destination identities. 
> 
> privacy-04 already says that UA's don't insert RPID headers.  If a UA
> is going to do it anyway, I don't see how a MUST NOT will stop it.
> At the proxy, it still needs to handle INVITEs from both trusted
> and untrusted sources, so considering the UA an untrusted source
> simplifies the number of cases to specify and handles some errors.
> I'd tend to agree that such architectures may cause interoperability
> problems, but have no basis in facts for that belief.
> 

Until sip-privacy removes the 'screen=' parameter, sip-privacy still has
sunny-day handling for the insertion of RPID headers by untrusted UAs. I'm
not suggesting that we add text saying that untrusted UAs MUST NOT add the
RPID. I'm suggesting that the concept of an 'untrusted' RPID be expunged (by
removing the 'screen' concept and the associated protocol apparatus), and
that proxies MUST remove RPIDs they receive from untrusted sources. I think
would be much more effective.

> >					If you would like to see
> > the changes I suggested, they are available in the ML archives here
> > 
> http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04258.html
> 
> Flemming already responded to this message, point by point, 
> msg04277.html.
> 

Yes, Flemming did reply to my comments, but I don't think that exchange
necessarily resolved the open issues. Since that time (especially during the
WG meeting in Minneapolis) I've been focusing primarily my efforts on just
the first issue in my mail, the 'screen=' parameter. If we get past this
I'll move on to further issues -  I feel 'screen' has the highest priority.
Suffice it to say that I'm not backing away from any of my other arguments -
I'm just trying to handle one matter at a time.

> > There are also some issues with the mechanism in so far as a trusted
entity
> > sends out a chunk of encrypted data in an RPID that only it can decrypt,
and
> > then expects entities that want access to this information to 'get back'
to
> > them via the given hostname and port. First of all, while it may be
> > specified elsewhere, nothing that I can see in sip-privacy-04 tells me
how I
> > can 'get back' to the host that generated the encrypted data. 
> 
> Section 7.2 clearly states that it generates a sip: or sips: URI,
> and has requirements for the content of the hostport, and for the
> contents of the username, and for a user-param.  Section 6.1 states
> that such a URI can be used as a Request-URI by the UA for certain
> call control functions or subsequent calls.  While the mechanisms you
> described can also be done, a simple INVITE to the enrypted URI
> is simpler.

So, then, you mean that I'm supposed to send an INVITE to the hostname and
port identified in the encrypted URI when I want that URI translated? I had
gotten the impression from the text that I could use the hostname and port
to query for a translation of the encrypted URI to its original form. I
guess I got that impression from the line in 6.1:

   ... whereas the "hostport" 
   identifies the entity that can decrypt the information

But now I can see how this admits of either reading. However, I'm a little
confused about how this is then supplying a valuable privacy service. When I
request privacy for a call, I don't exactly expect that the person I'm
calling will be able to call me back. But 7.2 suggests that encrypting the
URI in this way makes it suitable for transmission to untrusted entities. Is
this the desired behavior? Or does the server decrypting the URI make some
further decision to disallow certain requests to that URI? If so, this
should at least be mentioned.

As a means of malicious call trace (which I assume is the motivating
requirement) this also seem a little counterintuitive, especially if, as the
tracing network, I don't intend to start a session with the entity
identified by the URI. Is there some other request (maybe part of the
'certain call control functions') you mention above that I can send that
will just result in the decrypting of the URI? Or should the tracing network
requests a translation out-of-band somehow?

I must also point out that the other concerns I raised in my previous mail
are quite material to whether or not this is encryption mechanism is
genuinely implementable. There really are a lot of points that would need to
be resolved.

> > 		....  whereas the Call-Info means "this is
> > unimportant information that the end user should display if they can,
> > otherwise, don't worry about it."
> 
> My old rotary phone does exactly this with the Caller-ID 
> info.  Perhaps I need to upgrade my UA.

When I read things like this I'm really not sure if you're being serious.
Are you honestly suggesting that it doesn't matter whether or not the
network-verified identity for a request is presented to the UAS and end
user? That it doesn't matter if trusted network elements communicate such
identities to untrusted elements? I'd like to think it is clear from my
wording above that Call-Info is 'optional' in the sense that users can
display it if they like, whereas RPID has strict rules concerning whether or
not a UA is eligible to receive the network-verified identity - this is
unrelated to whether or not the UA can display it. Hence the distinction I
attempted to draw between selectively rendering the content (RPID) and
rendering optional content (Call-Info). There are restrictions on where RPID
is passed that do not currently exist for Call-Info.

> 
> Bill Marshall
> wtm@research.att.com
> 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr  1 09:13:02 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>
Cc: sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
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You raise a reasonable point on how a malicious call trace would be
triggered.  In the PSTN, one dials a service code and the network does the
rest.  "The rest" consists of a printout of the call trace on a teletype in
a telco office, or the more modern equivalent -- the called party gets no
access to its contents.  Thinking in SIP terms, the called party would
presumably also do an INVITE to a service code and include the encrypted
token which encapsulates the necessary trace information.

-----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: Monday, April 01, 2002 7:59 AM
To: 'William Marshall'
Cc: sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft

[snip]

So, then, you mean that I'm supposed to send an INVITE to the hostname and
port identified in the encrypted URI when I want that URI translated? I had
gotten the impression from the text that I could use the hostname and port
to query for a translation of the encrypted URI to its original form. I
guess I got that impression from the line in 6.1:

   ... whereas the "hostport" 
   identifies the entity that can decrypt the information

But now I can see how this admits of either reading. However, I'm a little
confused about how this is then supplying a valuable privacy service. When I
request privacy for a call, I don't exactly expect that the person I'm
calling will be able to call me back. But 7.2 suggests that encrypting the
URI in this way makes it suitable for transmission to untrusted entities. Is
this the desired behavior? Or does the server decrypting the URI make some
further decision to disallow certain requests to that URI? If so, this
should at least be mentioned.

As a means of malicious call trace (which I assume is the motivating
requirement) this also seem a little counterintuitive, especially if, as the
tracing network, I don't intend to start a session with the entity
identified by the URI. Is there some other request (maybe part of the
'certain call control functions') you mention above that I can send that
will just result in the decrypting of the URI? Or should the tracing network
requests a translation out-of-band somehow?

I must also point out that the other concerns I raised in my previous mail
are quite material to whether or not this is encryption mechanism is
genuinely implementable. There really are a lot of points that would need to
be resolved.

> > 		....  whereas the Call-Info means "this is
> > unimportant information that the end user should display if they can,
> > otherwise, don't worry about it."
> 
> My old rotary phone does exactly this with the Caller-ID 
> info.  Perhaps I need to upgrade my UA.

When I read things like this I'm really not sure if you're being serious.
Are you honestly suggesting that it doesn't matter whether or not the
network-verified identity for a request is presented to the UAS and end
user? That it doesn't matter if trusted network elements communicate such
identities to untrusted elements? I'd like to think it is clear from my
wording above that Call-Info is 'optional' in the sense that users can
display it if they like, whereas RPID has strict rules concerning whether or
not a UA is eligible to receive the network-verified identity - this is
unrelated to whether or not the UA can display it. Hence the distinction I
attempted to draw between selectively rendering the content (RPID) and
rendering optional content (Call-Info). There are restrictions on where RPID
is passed that do not currently exist for Call-Info.

> 
> Bill Marshall
> wtm@research.att.com
> 
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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> > Just because your network cannot verify the authenticity of the
> > information does not mean the information is useless. With the present
> > semantics, some people (I'm one of them) still find it useful from a pure
> > caller-id service perspective to have a clear-text caller-id being
> presented
> > although it was untrusted as opposed to having nothing at all, as long as
> I know
> > it's untrusted.
>
> This is the issue that I was trying to zoom in on. Who generates that
> caller-id that will be displayed to the UAS in your Iraqi case? The UAC?

No - the Iraqi network.


> My
> point is that the From header already provides a clear-text caller-id that
> was generated by the originating user but not necessarily verified by any
> intermediary, right? With or without sip-privacy-04, the From header already
> provides an (untrusted) means for an end-user to assert an identity that
> could be displayed by the destination endpoint, which the destination
> endpoint does not know to be verified by the network. So what does untrusted
> RPID add to this?

It allows for a network-asserted identity provided by network A to be delivered
via network B, even though network B does not trust A.


>
> > The longer term problem with your statement is that it assumes
> > that the "screen" mechanism is the only authenticity mechanism forever. As
> I
> > have stated many times, we expect a generic security solution that can
> > authenticate various headers in the future. Once we get that, you can now
> have a
> > Remote-Party-Id signed, sealed and delivered from an "untrusted" network
> and
> > actually still be able to verify it even though your network provider
> can't
> > vouch for it. Pretty useful - no ?
> >
>
> Since I am (at the moment) only arguing for the removal of 'screen', I have
> a hard time seeing how I would have a problem with assuming that 'screen'
> goes away. Moreover I've argued that we can dispense with the 'screen'
> mechanism without solving the more difficult problem of signing RPIDs.

Yes you have, and you have also said that we must remove any untrusted
Remote-Party-Ids. Well, if there in the future is a signature covering the
Remote-Party-Id (among other things), the end-user can now actually see who is
vouching for the Remote-Party-Id, regardless of whether any intermediary proxies
trust the previous hop and/or verifies the signature. This makes the
Remote-Party-Id more generally usable in the future as well, which seems to be
something you want to prevent.

-- Flemming

--
Flemming Andreasen
Cisco Systems





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CC: "Peterson, Jon" <jon.peterson@neustar.biz>,
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Subject: Re: [Sip] Comment, SIP Privacy draft
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Henning Schulzrinne wrote:

> I think we should be careful of trying to over-engineer what's primarily
> a social or legal problem.
>

I agree completely.

-- Flemming


--
Flemming Andreasen
Cisco Systems





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From daemon@ns.ietf.org  Mon Apr  1 10:24:49 2002
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"Peterson, Jon" wrote:

>
> Until sip-privacy removes the 'screen=' parameter, sip-privacy still has
> sunny-day handling for the insertion of RPID headers by untrusted UAs. I'm
> not suggesting that we add text saying that untrusted UAs MUST NOT add the
> RPID. I'm suggesting that the concept of an 'untrusted' RPID be expunged (by
> removing the 'screen' concept and the associated protocol apparatus), and
> that proxies MUST remove RPIDs they receive from untrusted sources. I think
> would be much more effective.

You are concerned about somebody doing something the draft does not define, and
in fact explicitly states it is not intended for. You have not provided any
technical problems with this, but are only concerned with the potential for
misuse. There is no point in either of us continuing to repeat the same
arguments on this point. If somebody else feels strongly about this point and
has something technical to offer, they should speak up.


> But now I can see how this admits of either reading. However, I'm a little
> confused about how this is then supplying a valuable privacy service. When I
> request privacy for a call, I don't exactly expect that the person I'm
> calling will be able to call me back.

Bill already explained how malicious call trace works - you will not be able to
call that party back.

-- Flemming

--
Flemming Andreasen
Cisco Systems




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"Peterson, Jon" wrote:

> Specifically, I am concerned with RPID usage as a replacement for the To and
> >From headers, in architectures which have proposed that To and From should
> always contain random/anonymous information for all calls (not just when
> privacy is needed) and RPID alone should be used to assert originating and
> destination identities. I feel that these architectures pose tangible
> threats to interoperability.

Well the draft doesn't say this (except for From in the anonymous case which I
have already agreed to change to "anonymous@localhost" instead), so this is
completely bogus.


> Were the SIP WG to pass sip-privacy-04
> as it stands it would be unnecessarily open to these sorts of abuses (though
> I would be the first to note that it is better than -03, which had no
> applicability statement and did not require IANA registration for
> extensions). If the remaining loopholes were closed, I would go back to
> ignoring sip-privacy-* as I have in the past; it certainly was not until I
> heard about the RPID To/From replacement architectures that I became, as I
> am now, interested is seeing changes to the draft.

Well, this is big part of the problem. The draft has been in the works for three
years and there has been plenty of opportunity for you to participate and shape
the draft. What is in the privacy-04 draft is the result of the discussions and
consensus that have formed among the active participants in this process.
However, as you admit, you have not been interested in contributing to this
draft but instead chose to ignore it until the very last minute, where you then
show up with a laundry list of items you want changed, and in most cases not
because there is anything technically wrong, but simply because you personally
disagree with them and fear that some people may use them in ways you do not
like. You claim to not be interested in killing the privacy draft, but your
timing could hardly have been more obstructive. I will furthermore also note,
that you have not managed to produce a lot of support for your positions. I
think this goes a long way towards deciding what to do with some of your
comments.


> If you would like to see
> the changes I suggested, they are available in the ML archives here:
>
> http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04258.html
>

To which I responded, but you didn't follow up (in fact I will note that WG Last
Call for this draft ended before you decided to revive just one of your
objections).

>
> This is not to say that all SIP headers must be overspecified to prevent
> such abuses. Again, this happened to come up for RPID and not for anything

> else, and the squeaky wheel gets the grease.

Can it get any more hypocritical than this ? It's OK for a bunch of headers to
be underspecified and open to abuse, but it's not OK for Remote-Party-Id to be
open to abuse, even though there are explicit statements in there saying what it
is *not* intended for and furthermore extensibility is under the control of
designated expert review (all of which were added in an attempt to accommodaty
you and address your concerns, but it is still not good enough for you). I'm
becoming less and less convinced that you are interested in anything but killing
this draft.

-- Flemming

--
Flemming Andreasen
Cisco Systems





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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Private SIP Extension for Service Route Discovery in 
                          Some Networks
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-scvrtdisco-00.txt
	Pages		: 12
	Date		: 28-Mar-02
	
This document proposes a private SIP extension header used in
conjunction with responses to REGISTER messages to provide a
mechanism by which the registrar may inform the UA of a service route
that the UA may use to request outbound services from the registrar's
domain.

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	Title		: SIP Extension for Registering Non-Adjacent Contacts
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-path-02.txt
	Pages		: 10
	Date		: 28-Mar-02
	
The REGISTER function is used in a SIP system primarily to associate
a temporary contact address with an address-of-record.  This contact
is generally in the form of a URI, such as Contact:
&ltsip:alice@pc33.atlanta.com> and is generally dynamic and
associated with the IP address or hostname of the SIP UA.  The
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from the user's home network to the registered UA must traverse these
proxies.  The REGISTER method itself does not give us a mechanism to
discover and record this sequence of proxies in the registrar for
future use.  This document defines an extension header, 'Path' which
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Hi everybody,
          I am student of SUNY at Stonybrook. I have to
do a project on Formal Verification of a protocol. I
wanted to work on something which has not been done
before. I was searching the net looking for a protocol
for few weeks now. I came across this SIP protocol.
Can you please guide me if anybody has done
Verification/model checking using any
tool(murphy,spin,XML..) on this protocol?
And it will be nice of you if you could guide me about
aspects/properties i can concentrate for verification of this 
protocol. I will
highly appreciate if you could guide this student.



thanks,
Nitin Khosla



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From daemon@ns.ietf.org  Mon Apr  1 10:43:28 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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To: Rohan Mahy <rohan@cisco.com>
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Subject: Re: [Sip] Summarization of Reason header discussion
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Rohan Mahy wrote:

> Hi Folks,
> 
> I haven't heard anything more about the Reason header discussion, so I'm
> going to summarize what I think we have consensus on.
[...]


Rohan:

I listed this before as well, but once again, the I-D says that the
header can appear in any request or response.  IMHO, it can appear
in any request, but should be limited to some responses.  There was
some email exchange between me, Sean Olson, and Tom Taylor, which
resulted in 155 and 183 being possible responses where the header
adds some value.

 
> 3) From a syntax perspective, I think we got agreement that the Reason
> should always begin with a SIP response code and a textual phrase, but may
> also include other responses/code families as extra parameters (for
> example, CPIM error messages, Q.850 error codes, etc.).  The motivation is
> that a SIP device only needs to know about one set of response codes (SIP)
> to do something appropriate with the Reason, but additional information is
> available if you care.
> 
> So, I'd like to find out if folks agree that my summary is accurate, and
> if we can move forward on defining a Reason header in SIP.


One more question, if the Reason header always begins with a response
code, can it appear in a request that initiates a dialog -- since there
is no response to pass through in this case.  This is more of case 5
Gonzalo identified in
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04410.html
I re-read that again and am not quite sure if Reason is applicable in
instances where a dialog has not been established.  For example, when a
B2BUA invites the second leg of a 2-party call, what Reason header
will it carry in the initial INVITE towards that leg?

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@ns.ietf.org  Mon Apr  1 12:21:42 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Date: Mon, 1 Apr 2002 10:43:48 -0600
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Hi all,

Note that in Minneapolis (after some airing of this draft), I received
several strong sugestions to change the proposed header name from "Path"
to "RegisterRecordRoute, short form "RRR". This appears to be related to
thefact that the main thing the draft does is define RecordRoute for
REGISTER operations.

My current plan is to include this change along with any other nits in
the next rev of the draft, and hopefully to proceed to WGLC from there
in time to include this material in the "Bundle 3" IETF last call set.

So please get any comments in ASAP -- I'd like to edit the next rev next
week.

--
Dean


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Internet-Drafts@ietf.org
> Sent: Friday, March 29, 2002 6:03 AM
> To: IETF-Announce:
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-willis-sip-path-02.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> 
> 
> 	Title		: SIP Extension for Registering 
> Non-Adjacent Contacts
> 	Author(s)	: D. Willis
> 	Filename	: draft-willis-sip-path-02.txt
> 	Pages		: 10
> 	Date		: 28-Mar-02
> 	
> The REGISTER function is used in a SIP system primarily to 
> associate a temporary contact address with an 
> address-of-record.  This contact is generally in the form of 
> a URI, such as Contact: &ltsip:alice@pc33.atlanta.com> and is 
> generally dynamic and associated with the IP address or 
> hostname of the SIP UA.  The problem is that network topology 
> may be that there are one or more SIP proxies between the UA 
> and the registrar, such that any message from the user's home 
> network to the registered UA must traverse these proxies.  
> The REGISTER method itself does not give us a mechanism to 
> discover and record this sequence of proxies in the registrar 
> for future use.  This document defines an extension header, 
> 'Path' which provides such a mechanism.
> 
> A URL for this Internet-Draft is: 
> http://www.ietf.org/internet-drafts/draft-willis-sip-path-02.t
xt

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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Flemming Andreasen" <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: "'William Marshall'" <wtm@research.att.com>, <sip@ietf.org>
Subject: RE: [Sip] Comment, SIP Privacy draft
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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Flemming Andreasen
>
> "Peterson, Jon" wrote:
>
> >
> > Until sip-privacy removes the 'screen=' parameter, sip-privacy still has
> > sunny-day handling for the insertion of RPID headers by
> untrusted UAs. I'm
> > not suggesting that we add text saying that untrusted UAs MUST
> NOT add the
> > RPID. I'm suggesting that the concept of an 'untrusted' RPID be
> expunged (by
> > removing the 'screen' concept and the associated protocol
> apparatus), and
> > that proxies MUST remove RPIDs they receive from untrusted
> sources. I think
> > would be much more effective.
>
> You are concerned about somebody doing something the draft does
> not define, and
> in fact explicitly states it is not intended for. You have not
> provided any
> technical problems with this, but are only concerned with the
> potential for
> misuse. There is no point in either of us continuing to repeat the same
> arguments on this point. If somebody else feels strongly about
> this point and
> has something technical to offer, they should speak up.
>
>

The issue Jon talks about is more philisophical than technical, so it is
hard to offer something technical. The problem is, what sort of errors can
occur? If an untrusted RPID is present in a message, then a UA should not
trust it. It is rather unclear to me what useful thing a UA can do with an
RPID _without_ trusting it, other than perhaps displaying it to the user
with some sort of "don't trust this" disclaimer.

Now, that behavior would be similar to today's use of email, where your
email reader reports this message is from me simply due to an untrusted from
header. Unfortunately, less sophisticated users do not understand what it
means when I say that From: is untrusted. They will assume the email is from
me, unless there is some evidence to the contrary. Indeed, they will tend to
doubt me if I were to say that I did _not_ send the email. I bring up email
only as an existance proof that the general public will often screw up when
handed an untrusted identity--and to suggest that we do not repeat that
design mistake.

So, an untrusted RPID allows a mistake on the part of the UA (perhaps doing
some sort of automatic processing) or by the user (by not really
understanding what trust means.) This creates a fail-unsafe condition (that
is, a failure on the part of the UA does damage)

If there were no RPID at all, both of these mistakes are impossible. This
creates a fail-safe situation, where the UA (and user) are unable to hurt
themselves by improperly handling the information.

Now, the only real _advantage_ I can think of keeping untrusted RPID headers
is the situation where one hop cannot verify the identity, but the next hop
can. But if I read the screen token section correctly, a "no" always wins if
there are multiple screen tokens. So, if P(1) says no, and P(2) says yes,
the UA should still treat the value as "no." Is this the intended
interpretation?



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        "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: Re: [Sip] Comment, SIP Privacy draft
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Ben Campbell wrote:

> The issue Jon talks about is more philisophical than technical, so it is

> hard to offer something technical. The problem is, what sort of errors can
> occur? If an untrusted RPID is present in a message, then a UA should not
> trust it. It is rather unclear to me what useful thing a UA can do with an
> RPID _without_ trusting it, other than perhaps displaying it to the user
> with some sort of "don't trust this" disclaimer.
>

That's the hard part about discussing philosophy which is why we really
shouldn't, because we are not all going to agree. I do believe the above is
actually useful, inasmuch as it is better than not getting anything at all.


> Now, that behavior would be similar to today's use of email, where your
> email reader reports this message is from me simply due to an untrusted from
> header. Unfortunately, less sophisticated users do not understand what it
> means when I say that From: is untrusted. They will assume the email is from
> me, unless there is some evidence to the contrary. Indeed, they will tend to
> doubt me if I were to say that I did _not_ send the email. I bring up email
> only as an existance proof that the general public will often screw up when
> handed an untrusted identity--and to suggest that we do not repeat that
> design mistake.

I would argue the other way and actually see e-mail as an excellent existence
proofs that this is both workable and useful. People that are not content with
that level of security can for example PGP sign their e-mails and get better
security. Similarly, if people are not happy with the screen parameter, they can
define a more comprehensive and general security solution and start applying
that on top. What is the problem here ?

>
> So, an untrusted RPID allows a mistake on the part of the UA (perhaps doing
> some sort of automatic processing) or by the user (by not really
> understanding what trust means.) This creates a fail-unsafe condition (that
> is, a failure on the part of the UA does damage)
>

There is no end to what kind of mistakes can be made by a UA. Using this kind of
logic, we should then note that bad things can happen if you blindly trust the
information provided in the From header field, and hence we should make sure
this can't happen by obfuscating the From header field. However, nobody is
seriously arguing this way for good reason.

> If there were no RPID at all, both of these mistakes are impossible. This
> creates a fail-safe situation, where the UA (and user) are unable to hurt
> themselves by improperly handling the information.
>

If you really want to apply that philosophy to the SIP protocol, then we have a
lot of unfinished work ahead of us. It certainly hasn't been a design principle
so far.

>
> Now, the only real _advantage_ I can think of keeping untrusted RPID headers
> is the situation where one hop cannot verify the identity, but the next hop
> can.

That is one advantage. Also, the ability to have a stronger security solution
applied on top is another (and I still think there is value in an untrusted
Remote-Party-Id as opposed to no Remote-Party-Id at all).


> But if I read the screen token section correctly, a "no" always wins if
> there are multiple screen tokens. So, if P(1) says no, and P(2) says yes,
> the UA should still treat the value as "no." Is this the intended
> interpretation?

Strictly speaking yes, but that part doesn't really apply here (there's only
supposed to be one token). If P(2) can actually authenticate the
Remote-Party-Id, it should set the screen parameter to "yes" (if there is a "no"
parameter, it should be removed).

-- Flemming

--
Flemming Andreasen
Cisco Systems



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Date: Mon, 01 Apr 2002 12:54:58 -0500
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Organization: Columbia University
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To: Ben Campbell <bcampbell@dynamicsoft.com>
CC: Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: Re: [Sip] Comment, SIP Privacy draft
References: <HNEOJECGFHIABDLENMMCEEPECAAA.bcampbell@dynamicsoft.com>
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> handed an untrusted identity--and to suggest that we do not repeat that
> design mistake.

In the absence of a PKI or only receiving email from parties that you
have a shared key with, would you suggest we remove or don't display the
From header in email? The problem is that it works for 99+% of the cases
without authentication, so removing this capability without a credible
replacement isn't too helpful, even if it makes the purists feel better.

> Now, the only real _advantage_ I can think of keeping untrusted RPID headers
> is the situation where one hop cannot verify the identity, but the next hop
> can. But if I read the screen token section correctly, a "no" always wins if
> there are multiple screen tokens. So, if P(1) says no, and P(2) says yes,
> the UA should still treat the value as "no." Is this the intended
> interpretation?

Or if you treat this as yet another indication of what the sender may
be. If I recognize the name, there's a high likelihood that it is
correct, just like From headers in email, but before I send $100,000 to
the account listed at a bank in Nigeria, trust that this is indeed BMW's
Director of North American operations (to cite another semi-public
version of identity fraud) or open that binary attachment, I'd rather
confirm that this is safe. As the latter case shows, the major damage
that identity impersonation does today would not be cured by a trusted
identity at all.

Unless it can be shown that the From header in SIP-bis subsumes all of
the RPID functionality, I fail to see what the harm is of providing
additional information, as long as it is labeled appropriately.

I can see one minor advantage: It does add the equivalent of the
X-Authentication header in email, namely an indication that From and
identity as seen by a proxy doesn't match, which would presumably be a
clue that something might (or might not) be amiss. Yes, it can be faked,
but it makes life a tad harder for SIP spammers using relays.

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From daemon@optimus.ietf.org  Mon Apr  1 16:08:24 2002
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Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
To: "Dean Willis" <dean.willis@softarmor.com>
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Some quick questions:


*) Question 1
============
Section 4.1 quotes:
"It has been suggested that the UA MAY choose to store the contents of    the Path header for future use as a preloaded
Route for use when the  UA wishes to send a SIP message that, for reasons of acquiring  services in the home network,
need to transit the service proxies in    the home network.  Such usage is explicitly outside the scope of this
 document."


Section 4.4: Procedures at the Home Proxy

"With the addition of Path, the home service proxy also copies  (inverted) the route set associated with the specific
contact in the  registrar database into the Route header of the outgoing message as a preloaded route.
 This causes the outoing message to transit the set   of proxies that indicated that they were to be used in future
   messages to that contact by including themseleves in the Path header"


ARC> Is it correct to assume that in such a n/w the homeproxy will surely be in the path of all SIP requests originating
from this UA ? If so, then Path orders are already being converted to Route Sets by the outgoing proxy as per above
qoute. So if the UA were to 'pre load' this same route (earlier quote) for whatever reason,
 what would happen when this pre-loaded route arrives at the Home Proxy who will attempt the same operation ?



*) Question 2
============
Also, the path header is being introduced because doing the same in RR is
not backward compatible.
Just wondering: The RR has progressed in such a way that we now have the
old R-R (no pre-loaded routes)
and the new R-R (pre-loaded routes). The current RFC states that
record-routes should be ignored in REGISTERs
and not followed in its 200OK.  To allow this case, cant this wording be
changed in the RFC saying the REGISTRar
may choose to ignore Route sets in REG/OK ? Even as of today, putting RR in
REGISTER wont break impl - it would
be ignored. Anyway backwards compatibility hardly seems to be the concern
specifically with RR looking at how
it has changed over time. So why not relax its usage and avoid another
header that does exactly the same thing ?

I am in favour of not overloading one header to mean a hundred things, but
to me this case looks like it means the same
thing really. Just that now we are saying RR is possible for all requests.
What the entities do with RR in non-dialog
related transactions is upto it.

Regds
Arjun
--
Arjun Roychowdhury @ Hughes
11717  Exploration Lane,
Germantown, MD, 20876
(O): 301-212-7860    (M):240-475-2139







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From daemon@optimus.ietf.org  Mon Apr  1 16:09:41 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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Cc: "'William Marshall'" <wtm@research.att.com>,
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Subject: RE: FW: [Sip] Comment, SIP Privacy draft
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A pointed question and a few other comments below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Monday, April 01, 2002 5:56 AM
> To: Peterson, Jon
> Cc: 'William Marshall'; 'sip@ietf.org'; Dean Willis; Joerg Ott; Rosen,
> Brian; Rohan Mahy
> Subject: Re: FW: [Sip] Comment, SIP Privacy draft
> 
> 
> 
> 
> 
[snip]
> > > With the present
> > > semantics, some people (I'm one of them) still find it useful from a
pure
> > > caller-id service perspective to have a clear-text caller-id being
presented
> > > although it was untrusted as opposed to having nothing at 

[snip]
> 
> It allows for a network-asserted identity provided by network A to be
delivered
> via network B, even though network B does not trust A.
> 

Yes, you seem to have skipped the question I was building up to... Who is
supposed to be the consumer of the network-asserted identity here, then? You
say above that you want 'screen' for Caller-ID. But in this Iraqi case,
surely if the originating user requested anonymity, no Caller-ID is going to
be displayed (and the From field would be just as good as RPID if anonymity
weren't requested). Since its untrusted, I don't think the RPID could be
used by the network either (for call-trace or billing). So who will be using
the untrusted RPID? I see no reason why the answer isn't 'no one', and that
seems to suggest to me that untrusted RPIDs can be removed at the trust
boundary.

IIRC this is the only example for which you have argued, to date, that
'screen' is necessary. If we can't substantiate a use for 'screen' in this
case, I'm left with the impression that 'screen' is unmotivated.

> >
> > Moreover I've argued that we can dispense with the 'screen'
> > mechanism without solving the more difficult problem of 
> > signing RPIDs.
> 
> Yes you have, and you have also said that we must remove any untrusted
> Remote-Party-Ids. Well, if there in the future is a signature covering the
> Remote-Party-Id (among other things), the end-user can now actually see
who is
> vouching for the Remote-Party-Id, regardless of whether any intermediary
proxies
> trust the previous hop and/or verifies the signature. This makes the
> Remote-Party-Id more generally usable in the future as well, which seems
to be
> something you want to prevent.

Well, fair enough - I'm not averse to discussing this point too, but first
discuss the matter at hand: How would the 'screen' parameter help the
end-user if they could just read the signature of messages? What, if
anything, does this argue in favor of keeping 'screen'?

Now on this other matter, I suspect that the introduction of signatures just
does away with the concept of an intrinsically 'untrusted' RPID, since every
element in the network would be empowered to have its own opinion about
whose RPIDs they trust. In that sense, there would no longer be any border
elements that make trust decisions on behalf of the entire federation. I
have in fact been arguing that the concept of an 'untrusted' RPID isn't
useful and that it should go away, so yes, I agree that signatures would be
an improvement in the mechanism. Indeed, if the mechanism supported per-RPID
signatures you and I would probably be one step closer to agreement, but the
devil is in the details with all this signing/encryption stuff.

> 
> -- Flemming
> 
> --
> Flemming Andreasen
> Cisco Systems
> 
> 
> 
> 

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From daemon@optimus.ietf.org  Mon Apr  1 16:20:36 2002
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From: "Couillaud, Pierre" <pierre.couillaud@polycom.com>
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Hello,

Here is my problem: given a small density gateway that support only up to a
fully channelized T1 interface,
that does not have any pretention to do PSTN routing, however that can be
used to provide PSTN access to 
SIP phones sitting behind it.... How do I register my PSTN interface to my
proxy ?

Can I register a route ?  And if so how ?  Can I just tell my proxy to
forward to the gateway any E.164 address 
that it cannot resolve by itself ?  I do not find anything in SIP that could
allow me to do something like this...  
Unless I am missing something here...

Thanks,

Pierre   

--- Cheers,
Pierre Couillaud, Polycom Canada Ltd.
Phone: (604) 697 9346 or (604) 990 5415 x146
"Anything one man can imagine, other men can make real", Jules Verne.


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From daemon@optimus.ietf.org  Mon Apr  1 16:42:57 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Francois Lessing'" <francoisx.lessing@intel.com>,
        SIP mailing list
	 <sip@ietf.org>
Subject: RE: [Sip] Can NOTIFY request fork ?
Date: Mon, 1 Apr 2002 16:19:30 -0500 
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From a purely theoretical point of view, the
question, "can I form a NOTIFY such that an
appropriately configured forking proxy will
be induced to fork it," the answer is "yes."
And, if you do so, you get what you deserve.

I contend that any network in which a NOTIFY
*does* fork has at least one misbehaving node.

More to the point: any mechanism that creates a
subscription without using SIP signalling is
completely responsible to make sure that the NOTIFY
requests are sent to one, and exactly one, subscriber.
I strongly suspect that any event package that
does not adequately address this issue will be laughed
out of the room if presented to any SIP-related
working group (as, indeed, it must for IANA
registration of a package name).

Look over the loose-routing stuff in the most recent
SIP spec if you intend to implement such a mechanism.
Note that this is not a trivial problem to solve, which
is why the sip-events document does not attempt to
specify how to address it.

/a

> -----Original Message-----
> From: Francois Lessing [mailto:francoisx.lessing@intel.com]
> Sent: Saturday, March 30, 2002 14:49
> To: SIP mailing list
> Subject: [Sip] Can NOTIFY request fork ?
> 
> 
> Hi,
> 
> I have been looking at non-SUBSCRIBE  mechanisms to create
> subscriptions and it seems that the use of such mechanisms mandates
> the possibility of forking behavior for NOTIFY requests.
> 
> My reasoning is as follows:
> The draft allows for non-SUBSCRIBE mechanisms to create
> subscriptions.
> 
> This means that a NOTIFY request has to be sent without a
> dialog being estblished (or at least one cannot assume that
> the user defined mechnism will create a dialog).
> 
> If the NOTIFY has to be sent without a dialog being established,
> a proxy may fork the request.
> 
> Is there something I missed ?
> 
> --
> Francois Lessing
> Trillium Digital Systems, a division of Intel Corporation
> Los Angeles, USA
> Tel: +1 310 481 5937
> Fax: +1 310 481 5558
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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"Peterson, Jon" wrote:

> >
> > It allows for a network-asserted identity provided by network A to be
> delivered
> > via network B, even though network B does not trust A.
> >
>
> Yes, you seem to have skipped the question I was building up to... Who is
> supposed to be the consumer of the network-asserted identity here, then? You
> say above that you want 'screen' for Caller-ID. But in this Iraqi case,
> surely if the originating user requested anonymity, no Caller-ID is going to
> be displayed

Certainly, but call trace may still work. Just becuase the information was
untrusted does not mean it is false. If the Iraqi phone company keeps sending us
false information, we can always put appropriate procedures in place to deal
with that.

> (and the From field would be just as good as RPID if anonymity
> weren't requested).

Not if you want network asserted identity information which is what
Remote-Party-Id gives you.


> Since its untrusted, I don't think the RPID could be
> used by the network either (for call-trace or billing).

It is correct that Remote-Party-Id does not provide for non-repudiation, but if
you really want that, you have a hard problem on your hand. Present day
telephony and data service provide trace solutions without any resemblance of
such guarantees and I suspect they will continue that way for a while.

> So who will be using
> the untrusted RPID? I see no reason why the answer isn't 'no one', and that
> seems to suggest to me that untrusted RPIDs can be removed at the trust
> boundary.

As I have explained many times already, untrusted does not equal useless in my
philosophy book. Furthermore, if you remove it, you can't provide a more secure
solution (read: crypto signature) in the future for this mechanism without
having everybody being able to verify such signatures. Any intermediary not able
to do that will will impose it's ugly policy on you and remove it even though
the signature could perhaps be verified by somebody else.


>
> > >
> > > Moreover I've argued that we can dispense with the 'screen'
> > > mechanism without solving the more difficult problem of
> > > signing RPIDs.
> >
> > Yes you have, and you have also said that we must remove any untrusted
> > Remote-Party-Ids. Well, if there in the future is a signature covering the
> > Remote-Party-Id (among other things), the end-user can now actually see
> who is
> > vouching for the Remote-Party-Id, regardless of whether any intermediary
> proxies
> > trust the previous hop and/or verifies the signature. This makes the
> > Remote-Party-Id more generally usable in the future as well, which seems
> to be
> > something you want to prevent.
>
> Well, fair enough - I'm not averse to discussing this point too, but first
> discuss the matter at hand: How would the 'screen' parameter help the
> end-user if they could just read the signature of messages? What, if
> anything, does this argue in favor of keeping 'screen'?
>

If you don't have screen, you need to specify that you remove
unverified/untrusted Remote-Party-Id. If you remove untrusted Remote-Party-Id,
you require all intermediaries to be able to either trust or verify a given
Remote-Party-Id. If you have a signature on a Remote-Party-Id, you don't want to
remove an untrusted Remote-Party-Id since a downstream entity can verify the
signature to determine authenticity.


> Indeed, if the mechanism supported per-RPID
> signatures you and I would probably be one step closer to agreement, but the
> devil is in the details with all this signing/encryption stuff.
>

If you had paid any attention to previous (and even fairly recent) discussions
about this draft, you would know that it used to have a signature mechanism, but
it was noted that it was a special case of a general security problem. General
problems should have general solutions, and hence we are waiting for the
security gurus to come up with that general security solution which could, among
other things, sign Remote-Party-Ids. This consensus has been formed long time
ago and was repeated recently.

-- Flemming


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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
Cc: "Flemming Andreasen" <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
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Subject: RE: [Sip] Comment, SIP Privacy draft
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> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Monday, April 01, 2002 11:55 AM
> To: Ben Campbell
> Cc: Flemming Andreasen; Peterson, Jon; 'William Marshall'; sip@ietf.org
> Subject: Re: [Sip] Comment, SIP Privacy draft
>
>
> > handed an untrusted identity--and to suggest that we do not repeat that
> > design mistake.
>
> In the absence of a PKI or only receiving email from parties that you
> have a shared key with, would you suggest we remove or don't display the
> From header in email? The problem is that it works for 99+% of the cases
> without authentication, so removing this capability without a credible
> replacement isn't too helpful, even if it makes the purists feel better.

The problem is that untrusted From headers in email has led to a general
population of users that believe they _can_ trust the from header.

>
> > Now, the only real _advantage_ I can think of keeping untrusted
> RPID headers
> > is the situation where one hop cannot verify the identity, but
> the next hop
> > can. But if I read the screen token section correctly, a "no"
> always wins if
> > there are multiple screen tokens. So, if P(1) says no, and P(2)
> says yes,
> > the UA should still treat the value as "no." Is this the intended
> > interpretation?
>
> Or if you treat this as yet another indication of what the sender may
> be. If I recognize the name, there's a high likelihood that it is
> correct, just like From headers in email, but before I send $100,000 to
> the account listed at a bank in Nigeria, trust that this is indeed BMW's
> Director of North American operations (to cite another semi-public
> version of identity fraud) or open that binary attachment, I'd rather
> confirm that this is safe. As the latter case shows, the major damage
> that identity impersonation does today would not be cured by a trusted
> identity at all.

I assume you are referring to the explosion of email worms? I think that is
another problem entirely and only tangentally related to the discussion.

>
> Unless it can be shown that the From header in SIP-bis subsumes all of
> the RPID functionality, I fail to see what the harm is of providing
> additional information, as long as it is labeled appropriately.
>
> I can see one minor advantage: It does add the equivalent of the
> X-Authentication header in email, namely an indication that From and
> identity as seen by a proxy doesn't match, which would presumably be a
> clue that something might (or might not) be amiss. Yes, it can be faked,
> but it makes life a tad harder for SIP spammers using relays.
>


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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Flemming Andreasen" <fandreas@cisco.com>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>,
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> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Monday, April 01, 2002 11:37 AM
> To: Ben Campbell
> Cc: Peterson, Jon; 'William Marshall'; sip@ietf.org
> Subject: Re: [Sip] Comment, SIP Privacy draft
>
>
>
>
> Ben Campbell wrote:
>
> > The issue Jon talks about is more philisophical than technical, so it is
>
> > hard to offer something technical. The problem is, what sort of
> errors can
> > occur? If an untrusted RPID is present in a message, then a UA
> should not
> > trust it. It is rather unclear to me what useful thing a UA can
> do with an
> > RPID _without_ trusting it, other than perhaps displaying it to the user
> > with some sort of "don't trust this" disclaimer.
> >
>
> That's the hard part about discussing philosophy which is why we really
> shouldn't, because we are not all going to agree. I do believe
> the above is
> actually useful, inasmuch as it is better than not getting
> anything at all.
>
>
> > Now, that behavior would be similar to today's use of email, where your
> > email reader reports this message is from me simply due to an
> untrusted from
> > header. Unfortunately, less sophisticated users do not
> understand what it
> > means when I say that From: is untrusted. They will assume the
> email is from
> > me, unless there is some evidence to the contrary. Indeed, they
> will tend to
> > doubt me if I were to say that I did _not_ send the email. I
> bring up email
> > only as an existance proof that the general public will often
> screw up when
> > handed an untrusted identity--and to suggest that we do not repeat that
> > design mistake.
>
> I would argue the other way and actually see e-mail as an
> excellent existence
> proofs that this is both workable and useful. People that are not
> content with
> that level of security can for example PGP sign their e-mails and
> get better
> security. Similarly, if people are not happy with the screen
> parameter, they can
> define a more comprehensive and general security solution and
> start applying
> that on top. What is the problem here ?

The problem here is that email clients _don't_ make it clear that you cannot
trust the from information. You seem to think the general populace of email
users understand that they cannot trust the identity for unsigned messages,
or even understand what it means to trust or not trust an identity. Maybe
people just don't care--PGP must not have been a big seller to die off as a
commercial product. Or more likely they just don't understand. The point is,
if you provide untrusted information here, people will trust it anyway.


>
> >
> > So, an untrusted RPID allows a mistake on the part of the UA
> (perhaps doing
> > some sort of automatic processing) or by the user (by not really
> > understanding what trust means.) This creates a fail-unsafe
> condition (that
> > is, a failure on the part of the UA does damage)
> >
>
> There is no end to what kind of mistakes can be made by a UA.
> Using this kind of
> logic, we should then note that bad things can happen if you
> blindly trust the
> information provided in the From header field, and hence we
> should make sure
> this can't happen by obfuscating the From header field. However, nobody is
> seriously arguing this way for good reason.
>
> > If there were no RPID at all, both of these mistakes are
> impossible. This
> > creates a fail-safe situation, where the UA (and user) are
> unable to hurt
> > themselves by improperly handling the information.
> >
>
> If you really want to apply that philosophy to the SIP protocol,
> then we have a
> lot of unfinished work ahead of us. It certainly hasn't been a
> design principle
> so far.

"We should stop repeating a mistake" and "We should go back and fix every
occurance of a mistake" are not equivalant assertions.

>
> >
> > Now, the only real _advantage_ I can think of keeping untrusted
> RPID headers
> > is the situation where one hop cannot verify the identity, but
> the next hop
> > can.
>
> That is one advantage. Also, the ability to have a stronger
> security solution
> applied on top is another (and I still think there is value in an
> untrusted
> Remote-Party-Id as opposed to no Remote-Party-Id at all).

Are you thinking that someday the RPID might have some sort of signature
that could be verified by the UA, but not by a proxy? Well, when we have
such I would be perfectly happy to have the proxy send on an RPID it cannot
verify, as long as the proxy can see that it contains something that might
be verifiable by the client. (In general, that is the same as the multihop
scenario below, I think.)

>
>
> > But if I read the screen token section correctly, a "no" always wins if
> > there are multiple screen tokens. So, if P(1) says no, and P(2)
> says yes,
> > the UA should still treat the value as "no." Is this the intended
> > interpretation?
>
> Strictly speaking yes, but that part doesn't really apply here
> (there's only
> supposed to be one token). If P(2) can actually authenticate the
> Remote-Party-Id, it should set the screen parameter to "yes" (if
> there is a "no"
> parameter, it should be removed).

On a re-read of the proxy behavior section, I can see how that is the case.
I suggest if this behavior stays in spec, that the proxy behavior wording be
tightened a little bit. Someone not privy to these conversations might not
interpret the statement that there must be no more than one screen token to
mean the same as "the proxy must remove all previous screen tokens."


>
> -- Flemming
>
> --
> Flemming Andreasen
> Cisco Systems
>
>


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> 
> The problem is that untrusted From headers in email has led to a general
> population of users that believe they _can_ trust the from header.

This seems to be a user interface problem as much as a protocol problem.
Even regular users seem to have learned to distinguish the open/closed
lock icon on web browsers indicating trust levels. Since there doesn't
seem to be a viable alternative in the foreseeable future, given ample
opportunity in the email space to develop one, I don't see how this
discussion is particularly helpful.

> I assume you are referring to the explosion of email worms? I think that is
> another problem entirely and only tangentally related to the discussion.

I was trying to highlight three different aspects of
(mis)identification:

1) In some cases, even a certified identity doesn't help any. The
various Outlook attachments are an example where being 100% sure that
you sent the email is probably harmful as it may lead me to trust the
attachment more than I should.

2) See http://slate.msn.com/?id=2063114 for a recent "interesting" case
of email spoofing. In that case, the name of the author probably was
less important than his (faked) corporate affiliation. In that case, a
user agent that detected a set of incongruous routing headers and
flashed a warning probably would have been more helpful.

3) For most people, impersonation is not a major problem, as far as I
can tell. I don't get spam from known people (yet) and I doubt that most
of the email scams rely on impersonation. Having a certified identity
for shady.character@coldmail.com isn't particularly helpful.

Unlike email, with SIP it should be relatively easy to check whether an
address is valid in real time. (Indeed, having an identity that can be
verified if reachable, but is not callable, could be a useful service.)
True impersonation is also fairly hard to scale to the levels needed to
make spam work, as the spammer has to have access to my address book or
impersonate some well-known person. Most people would be suspicious if
they got email from Britney Spears.

Given the lack of real incentives for most people, I doubt that
certified identity is high on people's list of things to spend money on.
Certificates would have to be easily trusted (small number of global
CAs) and easily movable (as otherwise most calls from hotel and
payphones would remain unauthenticated).

Thus, I find it somewhat unfair and not particularly helpful to make
principled objections on "holier-than-email" grounds, particularly
absent a credible proposal that addresses these issues and has any real
chance of being deployable in years measured in single digits.

I would hope that RPID is going to be the exception, as S/MIME-signed
messages might offer a reasonable alternative, but not something we can
mandate as the only solution.

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From daemon@optimus.ietf.org  Tue Apr  2 07:44:25 2002
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D A
	CTION:draft-illis...
Date: Tue, 2 Apr 2002 13:25:57 +0100 
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I would prefer to keep Path, or at least something shorter that the
mouthfull that is currently proposed.

If you really must change it, then it should at least be hyphenated (twice)
for consistency with the other headers.

Note also that it no longer does the same as Record-Route, as Record-Route
is a bidirectional capability, and it was the same strong suggestions in
Minneapolis that precluded it having that one and same functionality.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com

> ----------
> From: 	Dean Willis[SMTP:dean.willis@softarmor.com]
> Sent: 	01 April 2002 17:43
> To: 	sip@ietf.org
> Subject: 	[Sip] Header name in draft-willis-sip-path-02.txt, was: I-D
> ACTION:draft-illis...
> 
> 
> Hi all,
> 
> Note that in Minneapolis (after some airing of this draft), I received
> several strong sugestions to change the proposed header name from "Path"
> to "RegisterRecordRoute, short form "RRR". This appears to be related to
> thefact that the main thing the draft does is define RecordRoute for
> REGISTER operations.
> 
> My current plan is to include this change along with any other nits in
> the next rev of the draft, and hopefully to proceed to WGLC from there
> in time to include this material in the "Bundle 3" IETF last call set.
> 
> So please get any comments in ASAP -- I'd like to edit the next rev next
> week.
> 
> --
> Dean
> 
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> > Behalf Of Internet-Drafts@ietf.org
> > Sent: Friday, March 29, 2002 6:03 AM
> > To: IETF-Announce:
> > Cc: sip@ietf.org
> > Subject: [Sip] I-D ACTION:draft-willis-sip-path-02.txt
> > 
> > 
> > A New Internet-Draft is available from the on-line 
> > Internet-Drafts directories.
> > 
> > 
> > 	Title		: SIP Extension for Registering 
> > Non-Adjacent Contacts
> > 	Author(s)	: D. Willis
> > 	Filename	: draft-willis-sip-path-02.txt
> > 	Pages		: 10
> > 	Date		: 28-Mar-02
> > 	
> > The REGISTER function is used in a SIP system primarily to 
> > associate a temporary contact address with an 
> > address-of-record.  This contact is generally in the form of 
> > a URI, such as Contact: &ltsip:alice@pc33.atlanta.com> and is 
> > generally dynamic and associated with the IP address or 
> > hostname of the SIP UA.  The problem is that network topology 
> > may be that there are one or more SIP proxies between the UA 
> > and the registrar, such that any message from the user's home 
> > network to the registered UA must traverse these proxies.  
> > The REGISTER method itself does not give us a mechanism to 
> > discover and record this sequence of proxies in the registrar 
> > for future use.  This document defines an extension header, 
> > 'Path' which provides such a mechanism.
> > 
> > A URL for this Internet-Draft is: 
> > http://www.ietf.org/internet-drafts/draft-willis-sip-path-02.t
> xt
> 
> To remove yourself from the IETF Announcement list, send a message to 
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> 
> Internet-Drafts are also available by anonymous FTP. Login with the
> username "anonymous" and a password of your e-mail address. After
> logging in, type "cd internet-drafts" and then
> 	"get draft-willis-sip-path-02.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
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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 sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Summary of RE: [Sip] Comment, SIP Privacy draft
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All,

I feel profoundly lucky that yesterday and Friday were public holidays in
the UK :-)

Perhaps I can take advantage of this vantage point to offer a summary of the
thread. Protagonists please speak up if I have mis-represented you below,
but please don't re-open old fronts.

1) Call-Info

Long arguments about the alledged similarities/differences between Call-Info
and RPID. Suffice to say that the point has been made that Call-Info (and
perhaps other things) could be abused to do the things that RPID does, and
in particular to do the 'Bad Things' that some are worried about with RPID.
The point of disagreement was whether in the case of Call-Info, this would
be 'major abuse', with RPID was set up to make these 'Bad Things' easy, or
whether there was some equivalence.

The key point is that this kind of abuse of Call-Info may happen if we do
not provide an alternative.

2) Abuse of From/To headers

Jon is concerned that 'untrusted RPID' provides a standardised alternative
to From/To, allowing Bad Things, like never putting the real From/To
information in the From/To fields.

Others have pointed out that the draft explicitly states that untrusted UAs
MUST NOT add RPID. However the draft explicitly described proxy handling of
RPID headers from untrusted sources (proxies OR UAs), and allows these to be
propogated, marked as untrusted.

Therefore, a UA which DID add RPID, although non-compliant to the draft,
would have its RPID propogated through the network.

3) 'Untrustworthy RPID'

Flemming, Bill et al argued that RPID information is still useful when it is
'untrustworthy'. This can occur legitimately within the draft when an RPID
which is not 'private' (and so not encrypted) crosses a trust boundary. This
would represent an identity asserted by Network A and passed to Network B,
when B has no explicit trust relationship with A.

The contention was that since this information still has some value, it
should be passed to the UAS for display, and might also have some
application for call trace, although obviously not as useful as a
'trustworthy' identity.

Jon argued that the From field provides well enough for display, and the
value of untrustworthy information for call trace is debatable (and has been
debated at length).

There is agreement that some form of 'signed RPID' would be better.
Previously we have agreed on the need for a general mechanism to sign
information in SIP headers. Flemming has pointed out that if you agree to
have 'Signed RPID' then as a subset you have 'Untrustworthy RPID' (when you
don't recognise the signature). So removing 'Untrustworthy RPID' now would
be inconsistent, since there is already agreement to re-introduce it later.
That is unless you feel that the time interval during which Untrusted RPID
is not standardised is important.


Many other points were made, but the above seemed to me to be the key
issues.

And, now, my opinions:

1) Call-Info: I think the point has been made here - we're into judgement
calls on how bad an abuse X is compared to Y - I would rather there was no
need for people to abuse the protocol at all.

2) From/To: RPID is not the problem here, is it a solution to an underlying
problem with From/To. Blocking RPID will not solve the underlying problem,
and people will just route around to find another, probably worse, solution.
Better to address the real problem which has led people to propose
obfuscating From/To in the first place. If this problem is solved, then
there is no reason for people to want untrusted UAs to insert RPID. See
separate mail on this.

3) Untrustworthy RPID: I think Flemming and Bill have demonstrated the
utility of 'untrustworthy identity', and indeed this is what the From field
already provides. But From just provides the users assertion. If a network
has asserted additional identity information, why should the user should be
denied this ?

Having said that, the propogation of 'untrustworthy' RPID is strictly out of
scope according to the (retro-fitted) Applicability Statement. But then we
have agreement to add it 'in future' by virtue of the generic signing
mechanism - I guess this inconsistency is one for the ADs.

Regards,

Mark


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<P><FONT SIZE=3D2>All,</FONT>
</P>

<P><FONT SIZE=3D2>I feel profoundly lucky that yesterday and Friday =
were public holidays in the UK :-)</FONT>
</P>

<P><FONT SIZE=3D2>Perhaps I can take advantage of this vantage point to =
offer a summary of the thread. Protagonists please speak up if I have =
mis-represented you below, but please don't re-open old =
fronts.</FONT></P>

<P><FONT SIZE=3D2>1) Call-Info</FONT>
</P>

<P><FONT SIZE=3D2>Long arguments about the alledged =
similarities/differences between Call-Info and RPID. Suffice to say =
that the point has been made that Call-Info (and perhaps other things) =
could be abused to do the things that RPID does, and in particular to =
do the 'Bad Things' that some are worried about with RPID. The point of =
disagreement was whether in the case of Call-Info, this would be 'major =
abuse', with RPID was set up to make these 'Bad Things' easy, or =
whether there was some equivalence.</FONT></P>

<P><FONT SIZE=3D2>The key point is that this kind of abuse of Call-Info =
may happen if we do not provide an alternative.</FONT>
</P>

<P><FONT SIZE=3D2>2) Abuse of From/To headers</FONT>
</P>

<P><FONT SIZE=3D2>Jon is concerned that 'untrusted RPID' provides a =
standardised alternative to From/To, allowing Bad Things, like never =
putting the real From/To information in the From/To fields.</FONT></P>

<P><FONT SIZE=3D2>Others have pointed out that the draft explicitly =
states that untrusted UAs MUST NOT add RPID. However the draft =
explicitly described proxy handling of RPID headers from untrusted =
sources (proxies OR UAs), and allows these to be propogated, marked as =
untrusted.</FONT></P>

<P><FONT SIZE=3D2>Therefore, a UA which DID add RPID, although =
non-compliant to the draft, would have its RPID propogated through the =
network.</FONT></P>

<P><FONT SIZE=3D2>3) 'Untrustworthy RPID'</FONT>
</P>

<P><FONT SIZE=3D2>Flemming, Bill et al argued that RPID information is =
still useful when it is 'untrustworthy'. This can occur legitimately =
within the draft when an RPID which is not 'private' (and so not =
encrypted) crosses a trust boundary. This would represent an identity =
asserted by Network A and passed to Network B, when B has no explicit =
trust relationship with A.</FONT></P>

<P><FONT SIZE=3D2>The contention was that since this information still =
has some value, it should be passed to the UAS for display, and might =
also have some application for call trace, although obviously not as =
useful as a 'trustworthy' identity.</FONT></P>

<P><FONT SIZE=3D2>Jon argued that the From field provides well enough =
for display, and the value of untrustworthy information for call trace =
is debatable (and has been debated at length).</FONT></P>

<P><FONT SIZE=3D2>There is agreement that some form of 'signed RPID' =
would be better. Previously we have agreed on the need for a general =
mechanism to sign information in SIP headers. Flemming has pointed out =
that if you agree to have 'Signed RPID' then as a subset you have =
'Untrustworthy RPID' (when you don't recognise the signature). So =
removing 'Untrustworthy RPID' now would be inconsistent, since there is =
already agreement to re-introduce it later. That is unless you feel =
that the time interval during which Untrusted RPID is not standardised =
is important.</FONT></P>
<BR>

<P><FONT SIZE=3D2>Many other points were made, but the above seemed to =
me to be the key issues.</FONT>
</P>

<P><FONT SIZE=3D2>And, now, my opinions:</FONT>
</P>

<P><FONT SIZE=3D2>1) Call-Info: I think the point has been made here - =
we're into judgement calls on how bad an abuse X is compared to Y - I =
would rather there was no need for people to abuse the protocol at all.<=
/FONT></P>

<P><FONT SIZE=3D2>2) From/To: RPID is not the problem here, is it a =
solution to an underlying problem with From/To. Blocking RPID will not =
solve the underlying problem, and people will just route around to find =
another, probably worse, solution. Better to address the real problem =
which has led people to propose obfuscating From/To in the first place. =
If this problem is solved, then there is no reason for people to want =
untrusted UAs to insert RPID. See separate mail on this.</FONT></P>

<P><FONT SIZE=3D2>3) Untrustworthy RPID: I think Flemming and Bill have =
demonstrated the utility of 'untrustworthy identity', and indeed this =
is what the From field already provides. But From just provides the =
users assertion. If a network has asserted additional identity =
information, why should the user should be denied this ?</FONT></P>

<P><FONT SIZE=3D2>Having said that, the propogation of 'untrustworthy' =
RPID is strictly out of scope according to the (retro-fitted) =
Applicability Statement. But then we have agreement to add it 'in =
future' by virtue of the generic signing mechanism - I guess this =
inconsistency is one for the ADs.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Flemming Andreasen'"
	 <fandreas@cisco.com>
Cc: "'William Marshall'" <wtm@research.att.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: FW: [Sip] Comment, SIP Privacy draft
Date: Tue, 2 Apr 2002 15:06:28 +0100 
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Jon Peterson wrote:

> I only entered into this 
> fray because of
> the ways that RPID was proposed to be used, because a 
> concrete case was
> raised in which a major customer of SIP wanted to abuse RPID 
> and create
> serious interoperability concerns. 

This seems to be a key issue here, and one where I do share your concern.

To solve this we do need to solve the underlying problem which led to this
'abuse' of RPID. If we solve this problem, then I think all the rest of our
arguments drop out.

As I said before, I believe the arguments which led to this will lead to the
same place in other contexts - there is nothing special about this
particular 'customer' of SIP in this regard, so we really do need to solve
the root problem.

The root problem is this: A UA cannot place 'private' or 'potentially
private' information in the From or To fields.

This is because there is no mechanism defined to ensure that such
information is not passed to other UAs. In the absence of such a mechanism,
no requirements can be placed on what a UA does with the From/To fields. If
we do not place any requirements on these fields, then we cannot reliably
use them for anything.

We could argue about whether the very naming of the fields, examples in the
SIP spec, etc., define an implicit 'soft' requirement that they contain the
originator and desired destination (unless these are known to be private by
the UAC). The more we engineer this assumption into services, interworkings,
clients etc., then the more the implicit 'soft' requirement becomes in
practice an implicit 'hard' requirement.

We can see this already by the fact that a proposal in a particular system
to always include garbage in these fields causes such concern. People quite
reasonably want these fields to contain the real originator/destination.

In the case of a UA obtaining service from a public service provider, there
may be a requirement for the user to indicate their preferred originating
identity. This would naturally go in the From field, but it may be private
(to other UAs).

Equally, information which might usually be placed in From/To by the UA may
become private due to operation of services within the network. It would be
useful if this could be indicated in a simple way within a trust domain,
since modifying the From/To fields carries some cost.

My suggestions for what to do about this are:
1) We need to accept (and document) the fact that there will be many kinds
of privacy service, and many of these will require the modification of the
From/To fields. This cannot be done by Proxies (with 2543 compatibility), so
we must accept that any SIP-based public service offering (necessarily
incorporating privacy) cannot be based on proxies alone AND have 2543
compatibility.

2) Define a means for UAs to request privacy in respect of the From/To
fields (e.g. using a Generic URI parameter as proposed by Jon)

This would obviate the motivation for untrusted UAs to ever insert RPID. We
could change the privacy draft to differentiate between RPIDs received from
untrusted UAs and from untrusted proxies.

Regards...Mark


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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
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<TITLE>RE: FW: [Sip] Comment, SIP Privacy draft</TITLE>
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<BR>

<P><FONT SIZE=3D2>Jon Peterson wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; I only entered into this </FONT>
<BR><FONT SIZE=3D2>&gt; fray because of</FONT>
<BR><FONT SIZE=3D2>&gt; the ways that RPID was proposed to be used, =
because a </FONT>
<BR><FONT SIZE=3D2>&gt; concrete case was</FONT>
<BR><FONT SIZE=3D2>&gt; raised in which a major customer of SIP wanted =
to abuse RPID </FONT>
<BR><FONT SIZE=3D2>&gt; and create</FONT>
<BR><FONT SIZE=3D2>&gt; serious interoperability concerns. </FONT>
</P>

<P><FONT SIZE=3D2>This seems to be a key issue here, and one where I do =
share your concern.</FONT>
</P>

<P><FONT SIZE=3D2>To solve this we do need to solve the underlying =
problem which led to this 'abuse' of RPID. If we solve this problem, =
then I think all the rest of our arguments drop out.</FONT></P>

<P><FONT SIZE=3D2>As I said before, I believe the arguments which led =
to this will lead to the same place in other contexts - there is =
nothing special about this particular 'customer' of SIP in this regard, =
so we really do need to solve the root problem.</FONT></P>

<P><FONT SIZE=3D2>The root problem is this: A UA cannot place 'private' =
or 'potentially private' information in the From or To fields.</FONT>
</P>

<P><FONT SIZE=3D2>This is because there is no mechanism defined to =
ensure that such information is not passed to other UAs. In the absence =
of such a mechanism, no requirements can be placed on what a UA does =
with the From/To fields. If we do not place any requirements on these =
fields, then we cannot reliably use them for anything.</FONT></P>

<P><FONT SIZE=3D2>We could argue about whether the very naming of the =
fields, examples in the SIP spec, etc., define an implicit 'soft' =
requirement that they contain the originator and desired destination =
(unless these are known to be private by the UAC). The more we engineer =
this assumption into services, interworkings, clients etc., then the =
more the implicit 'soft' requirement becomes in practice an implicit =
'hard' requirement.</FONT></P>

<P><FONT SIZE=3D2>We can see this already by the fact that a proposal =
in a particular system to always include garbage in these fields causes =
such concern. People quite reasonably want these fields to contain the =
real originator/destination.</FONT></P>

<P><FONT SIZE=3D2>In the case of a UA obtaining service from a public =
service provider, there may be a requirement for the user to indicate =
their preferred originating identity. This would naturally go in the =
From field, but it may be private (to other UAs).</FONT></P>

<P><FONT SIZE=3D2>Equally, information which might usually be placed in =
From/To by the UA may become private due to operation of services =
within the network. It would be useful if this could be indicated in a =
simple way within a trust domain, since modifying the From/To fields =
carries some cost.</FONT></P>

<P><FONT SIZE=3D2>My suggestions for what to do about this are:</FONT>
<BR><FONT SIZE=3D2>1) We need to accept (and document) the fact that =
there will be many kinds of privacy service, and many of these will =
require the modification of the From/To fields. This cannot be done by =
Proxies (with 2543 compatibility), so we must accept that any SIP-based =
public service offering (necessarily incorporating privacy) cannot be =
based on proxies alone AND have 2543 compatibility.</FONT></P>

<P><FONT SIZE=3D2>2) Define a means for UAs to request privacy in =
respect of the From/To fields (e.g. using a Generic URI parameter as =
proposed by Jon)</FONT></P>

<P><FONT SIZE=3D2>This would obviate the motivation for untrusted UAs =
to ever insert RPID. We could change the privacy draft to differentiate =
between RPIDs received from untrusted UAs and from untrusted proxies.</F=
ONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>

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From daemon@optimus.ietf.org  Tue Apr  2 09:26:02 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Christian Huitema'" <huitema@windows.microsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>
Cc: Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
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Christian wrote:
> 
> > The reference to "Public telephony server" above is very misplaced.
> 3GPP &
> > DCS SIP networks are not telephony networks. We need less of this 
> > invective and more focus on the real problems.
> 
> The reference was not an invective, but rather an acknowledgement of
> legal implications of providing a "public service". The 
> requirements to
> be able to provide call tracing applies to public services, and result
> in a requirement to have something like the remote party ID. 

Apologies - the term 'public telephony' often has rather loaded meaning
round here and I mis-interpreted your comment. If you mean 'public service
provider' then I understand the distinction your are making.


> It is quite
> unclear that such a requirement applies to a private service, e.g.
> direct calls over the Internet between two corporations. 

What makes you think this ? At present a user receiving malicious calls at
work (from the PSTN) can ask for these to be traced. Why should they lose
this ability if their employer switches to a SIP-based telecommunications
service using the Internet ? (as well as getting worse QoS :-)

I think if it became widespread that corporations used Internet-based
telecommunications which did not support malicious call trace, then
regulators, not to say consumer groups and groups representing employees
might become concerned.

I would think we could make the SIP-based service at least as good as the
PSTN in this regard.

The 'trusted third party' in the form of the public service provider
performs a valuable service in mediating between the legitimate desire or
the caller to remain anonymous, and the equally legitimate desire for called
parties to be protected from malicious calls. A third party of some kind is
required to guard the privacy of the caller, whilst ensuring the tracebility
of the call.

Now there is no reason why this 'third party' needs to be a public service
provider. The same 'third party' function could be provided by proxies owned
by the corporations between which calls are being made.

We should be clear that if users choose to receive calls directly, without
such a 'trusted third party', then they will not receive the protection from
malicious calls that they are used to on the PSTN. It may be fine for
individual users to make this choice, but it is not something that anyone
providing service to many users (be that a commercial service, or an
employer providing telecommunications to their employees) could choose to
do.

So, we need mechanisms to enable this in as many scenarios as possible.

...Mark

OTOH, the
> existence of technologies such as remote-party ID have to be 
> taken into
> account when devising a privacy solution: we must assume that whatever
> information is given to the network will be disclosed.
> 
> -- Christian Huitema 
> 

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<P><FONT SIZE=3D2>Christian wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; The reference to &quot;Public telephony =
server&quot; above is very misplaced.</FONT>
<BR><FONT SIZE=3D2>&gt; 3GPP &amp;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; DCS SIP networks are not telephony =
networks. We need less of this </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; invective and more focus on the real =
problems.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The reference was not an invective, but rather =
an acknowledgement of</FONT>
<BR><FONT SIZE=3D2>&gt; legal implications of providing a &quot;public =
service&quot;. The </FONT>
<BR><FONT SIZE=3D2>&gt; requirements to</FONT>
<BR><FONT SIZE=3D2>&gt; be able to provide call tracing applies to =
public services, and result</FONT>
<BR><FONT SIZE=3D2>&gt; in a requirement to have something like the =
remote party ID. </FONT>
</P>

<P><FONT SIZE=3D2>Apologies - the term 'public telephony' often has =
rather loaded meaning round here and I mis-interpreted your comment. If =
you mean 'public service provider' then I understand the distinction =
your are making.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; It is quite</FONT>
<BR><FONT SIZE=3D2>&gt; unclear that such a requirement applies to a =
private service, e.g.</FONT>
<BR><FONT SIZE=3D2>&gt; direct calls over the Internet between two =
corporations. </FONT>
</P>

<P><FONT SIZE=3D2>What makes you think this ? At present a user =
receiving malicious calls at work (from the PSTN) can ask for these to =
be traced. Why should they lose this ability if their employer switches =
to a SIP-based telecommunications service using the Internet ? (as well =
as getting worse QoS :-)</FONT></P>

<P><FONT SIZE=3D2>I think if it became widespread that corporations =
used Internet-based telecommunications which did not support malicious =
call trace, then regulators, not to say consumer groups and groups =
representing employees might become concerned.</FONT></P>

<P><FONT SIZE=3D2>I would think we could make the SIP-based service at =
least as good as the PSTN in this regard.</FONT>
</P>

<P><FONT SIZE=3D2>The 'trusted third party' in the form of the public =
service provider performs a valuable service in mediating between the =
legitimate desire or the caller to remain anonymous, and the equally =
legitimate desire for called parties to be protected from malicious =
calls. A third party of some kind is required to guard the privacy of =
the caller, whilst ensuring the tracebility of the call.</FONT></P>

<P><FONT SIZE=3D2>Now there is no reason why this 'third party' needs =
to be a public service provider. The same 'third party' function could =
be provided by proxies owned by the corporations between which calls =
are being made.</FONT></P>

<P><FONT SIZE=3D2>We should be clear that if users choose to receive =
calls directly, without such a 'trusted third party', then they will =
not receive the protection from malicious calls that they are used to =
on the PSTN. It may be fine for individual users to make this choice, =
but it is not something that anyone providing service to many users (be =
that a commercial service, or an employer providing telecommunications =
to their employees) could choose to do.</FONT></P>

<P><FONT SIZE=3D2>So, we need mechanisms to enable this in as many =
scenarios as possible.</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>OTOH, the</FONT>
<BR><FONT SIZE=3D2>&gt; existence of technologies such as remote-party =
ID have to be </FONT>
<BR><FONT SIZE=3D2>&gt; taken into</FONT>
<BR><FONT SIZE=3D2>&gt; account when devising a privacy solution: we =
must assume that whatever</FONT>
<BR><FONT SIZE=3D2>&gt; information is given to the network will be =
disclosed.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- Christian Huitema </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'William Marshall'"
	 <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 2 Apr 2002 15:06:26 +0100 
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Jon Peterson wrote:
> 
> 
> A few more things about this Call-Info business.
<snip> 
> Call-Info
> clearly exists for a purpose unrelated to privacy/identity management,
> although I concede that its intended use may have its own privacy
> implications. But do your concerns about the legality of publically
> available services extend to instant messaging, Internet 
> gaming, and the
> like?

My concerns would extend to any service which inserted the real identity of
the user on their behalf, and did not provide a way for the user to turn
this off on a session by session basis.

I think in most (if not all) services on the Internet today, any identities
inserted by the network are user provided names which may or may not be
related to the users real identity.

Also, in practice, if a service always inserts the users real identity, this
is not mitigated by arguments along the lines of 'The header's semantics
mean that you should not always trust this identity' - the user may still
feel their privacy is compromised if their identity is given out without
their control, even if the header it is sent in is somehow marked as
'unreliable'.

So, if you wanted to build a service where an outbound home proxy inserted,
say, the user's photo for them, I think you would need a mechanism for the
user to prevent this happening on a per-session basis before the service
would be acceptable to the users, never mind the regulators. I guess you
could give the user a separate (virtual) outbound home proxy which did not
do this.

Anyway, your other comments suggested you did not see the need to proxies to
insert Call-Info at all, and if this were the case the problem goes away.

...Mark




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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] Comment, SIP Privacy draft</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Jon Peterson wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A few more things about this Call-Info =
business.</FONT>
<BR><FONT SIZE=3D2>&lt;snip&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Call-Info</FONT>
<BR><FONT SIZE=3D2>&gt; clearly exists for a purpose unrelated to =
privacy/identity management,</FONT>
<BR><FONT SIZE=3D2>&gt; although I concede that its intended use may =
have its own privacy</FONT>
<BR><FONT SIZE=3D2>&gt; implications. But do your concerns about the =
legality of publically</FONT>
<BR><FONT SIZE=3D2>&gt; available services extend to instant messaging, =
Internet </FONT>
<BR><FONT SIZE=3D2>&gt; gaming, and the</FONT>
<BR><FONT SIZE=3D2>&gt; like?</FONT>
</P>

<P><FONT SIZE=3D2>My concerns would extend to any service which =
inserted the real identity of the user on their behalf, and did not =
provide a way for the user to turn this off on a session by session =
basis.</FONT></P>

<P><FONT SIZE=3D2>I think in most (if not all) services on the Internet =
today, any identities inserted by the network are user provided names =
which may or may not be related to the users real identity.</FONT></P>

<P><FONT SIZE=3D2>Also, in practice, if a service always inserts the =
users real identity, this is not mitigated by arguments along the lines =
of 'The header's semantics mean that you should not always trust this =
identity' - the user may still feel their privacy is compromised if =
their identity is given out without their control, even if the header =
it is sent in is somehow marked as 'unreliable'.</FONT></P>

<P><FONT SIZE=3D2>So, if you wanted to build a service where an =
outbound home proxy inserted, say, the user's photo for them, I think =
you would need a mechanism for the user to prevent this happening on a =
per-session basis before the service would be acceptable to the users, =
never mind the regulators. I guess you could give the user a separate =
(virtual) outbound home proxy which did not do this.</FONT></P>

<P><FONT SIZE=3D2>Anyway, your other comments suggested you did not see =
the need to proxies to insert Call-Info at all, and if this were the =
case the problem goes away.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>
<BR>

</BODY>
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Supported header has lost its compact form 'k'.
Is this a known problem?


Christian Jansson
Hotsip AB

 

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From: "Gabriel Velo" <gvelo@astartec.com>
To: <sip@ietf.org>
Date: Tue, 2 Apr 2002 11:22:46 -0300
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Subject: [Sip] Residential UA security
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Dear All.

I need some help on this topic:

How can i protect a residential UA (my residential Sip Phone) from DOS
Attack like a flood of INVITE msg ??
In this scenario we don't have a validation since we need receive call from
anyone.
and .... anyone with a low bandwidth connection can make than my phone can't
stop of ring !!

Thanks in advance.

Gabriel.



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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Drage, Keith \(Keith\)'" <drage@lucent.com>, <sip@ietf.org>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
Date: Tue, 2 Apr 2002 08:39:51 -0600
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Keith said:

> I would prefer to keep Path, or at least something shorter 
> that the mouthfull that is currently proposed.

I suspect RRR would get used a lot . . .
 
> If you really must change it, then it should at least be 
> hyphenated (twice) for consistency with the other headers.

Register-Record-Route ?
 
> Note also that it no longer does the same as Record-Route, as 
> Record-Route is a bidirectional capability, and it was the 
> same strong suggestions in Minneapolis that precluded it 
> having that one and same functionality.

Actually, Record-Route entries are NOT changed in the response, just
copied back towards the source. That is, they accumulate on the INVITE
message, and are returned to the UA on the 200OK response. I think the
text in "path" now has exactly the same behaviour. If it doesn't, I did
something wrong, so please verify . . .

--
Dean


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From: "Dean Willis" <dean.willis@softarmor.com>
To: <aroychow@hns.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
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> -----Original Message-----
> From: aroychow@hns.com [mailto:aroychow@hns.com] 
> Sent: Monday, April 01, 2002 2:32 PM
> To: Dean Willis
> Cc: sip@ietf.org
> Subject: Re: [Sip] Header name in 
> draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
> 
> 
> 
> Some quick questions:
> 
> 
> *) Question 1
> ============
> Section 4.1 quotes:
> "It has been suggested that the UA MAY choose to store the 
> contents of    the Path header for future use as a preloaded
> Route for use when the  UA wishes to send a SIP message that, 
> for reasons of acquiring  services in the home network,
> need to transit the service proxies in    the home network.  
> Such usage is explicitly outside the scope of this
>  document."
> 
> 
> Section 4.4: Procedures at the Home Proxy
> 
> "With the addition of Path, the home service proxy also 
> copies  (inverted) the route set associated with the specific 
> contact in the  registrar database into the Route header of 
> the outgoing message as a preloaded route.
>  This causes the outoing message to transit the set   of 
> proxies that indicated that they were to be used in future
>    messages to that contact by including themseleves in the 
> Path header"
> 
> 
> ARC> Is it correct to assume that in such a n/w the homeproxy will 
> ARC> surely be in the path of all SIP requests originating
> from this UA ? If so, then Path orders are already being 
> converted to Route Sets by the outgoing proxy as per above 
> qoute. So if the UA were to 'pre load' this same route 
> (earlier quote) for whatever reason,  what would happen when 
> this pre-loaded route arrives at the Home Proxy who will 
> attempt the same operation ?

Doing something like that which is explicitly out of scope for this
document in 4.2 is an attempt to force the home proxy to be in the path
of all SIP requests from this UA. Without some kind of mechanism, the UA
may not have any way to route requests thru that home proxy. In this
scenario, the pre-loaded route is consumed getting to the home proxy, so
I don't thgink what you're worried about here actually happens.

See draft-willis-sip-svcrtdisco-00.txt for discussion of this technique.

> *) Question 2
> ============
> Also, the path header is being introduced because doing the 
> same in RR is not backward compatible. Just wondering: The RR 
> has progressed in such a way that we now have the old R-R (no 
> pre-loaded routes) and the new R-R (pre-loaded routes). The 
> current RFC states that record-routes should be ignored in 
> REGISTERs and not followed in its 200OK.  To allow this case, 
> cant this wording be changed in the RFC saying the REGISTRar 
> may choose to ignore Route sets in REG/OK ? Even as of today, 
> putting RR in REGISTER wont break impl - it would be ignored. 
> Anyway backwards compatibility hardly seems to be the concern 
> specifically with RR looking at how it has changed over time. 
> So why not relax its usage and avoid another header that does 
> exactly the same thing ?

There was list discussion of this several weeks ago, and I recall
Jonathan Lennox made a definitive report on why RR is not the way to go.
 


--
Dean


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From daemon@optimus.ietf.org  Tue Apr  2 10:10:18 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>
Cc: <sip@ietf.org>
Subject: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy draft)
Date: Tue, 2 Apr 2002 08:55:21 -0600
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Mark wrote:
-----
The root problem is this: A UA cannot place 'private' or 'potentially
private' information in the From or To fields.  

This is because there is no mechanism defined to ensure that such
information is not passed to other UAs. In the absence of such a
mechanism, no requirements can be placed on what a UA does with the
From/To fields. If we do not place any requirements on these fields,
then we cannot reliably use them for anything.
------

I think this is statement of the situation clearly brings out the
misunderstanding driving this whole thing.

This logic will eventually lead us to deleting all headers or bodies
provided by the UA in at attempt to enforce "privacy" against the wishes
of the actual user. It also precludes intentially authenticating the
user to a service on the other side of the home proxy, as such
authentication by definition is an exposure of private information.

I would restate as follows:

"The SIP protocol explicitly sends the From and To fields to the
destination UA. Therefore the sending UA should only insert information
in the From and To fields that is intended to be delivered to other UAs
and potentially displayed to the user of the other UA. If the user of
the UA wishes to remain anonymous, they must not insert any information
into the From or To fields which compromises this anonymity. This same
caveat applies to every other aspect of SIP, including headers and
bodies, except for those elements which are explicitly removed or
encrypted by intermediate elements known to the UA."

--
Dean


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From: "Dean Willis" <dean.willis@softarmor.com>
To: <gvelo@astartec.com>, <sip@ietf.org>
Subject: RE: [Sip] Residential UA security
Date: Tue, 2 Apr 2002 08:57:07 -0600
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I believe Henning has suggested doing a 401 challenge with null
authentication. This at least lets you know what IP address is flooding
you with INVITEs, which allows other steps to be taken.

--
Dean


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Gabriel Velo
> Sent: Tuesday, April 02, 2002 8:23 AM
> To: sip@ietf.org
> Subject: [Sip] Residential UA security
> 
> 
> Dear All.
> 
> I need some help on this topic:
> 
> How can i protect a residential UA (my residential Sip Phone) 
> from DOS Attack like a flood of INVITE msg ?? In this 
> scenario we don't have a validation since we need receive 
> call from anyone. and .... anyone with a low bandwidth 
> connection can make than my phone can't stop of ring !!
> 
> Thanks in advance.
> 
> Gabriel.
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 
> 


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        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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<snip>

> 3) Untrustworthy RPID: I think Flemming and Bill have demonstrated the
> utility of 'untrustworthy identity', and indeed this is what the From
> field already provides. But From just provides the users assertion. If
> a network has asserted additional identity information, why should the
> user should be denied this ?
>
> Having said that, the propogation of 'untrustworthy' RPID is strictly
> out of scope according to the (retro-fitted) Applicability Statement.

Actually, it's not. The Applicability Statement allows for more than a
single domain and does not specify whether such domains have to trust
each other or not, so either case is allowed and hence described in the
document.

-- Flemming


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Subject: RE: [Sip] Comment, SIP Privacy draft
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Batching up a few mails from Flemming, trying to catch up.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Monday, April 01, 2002 6:47 AM
> To: Peterson, Jon
> Cc: 'William Marshall'; sip@ietf.org
> Subject: Re: [Sip] Comment, SIP Privacy draft
> 

> 
> You are concerned about somebody doing something the draft does not
define, and
> in fact explicitly states it is not intended for. You have not provided
any
> technical problems with this, but are only concerned with the potential
for
> misuse. There is no point in either of us continuing to repeat the same
> arguments on this point. If somebody else feels strongly about this point
and
> has something technical to offer, they should speak up.
> 

I like Ben's way of putting it - that my argument is philosophical. The
philosophical part is "pluralitas non est ponenda sine necessitate". Or to
translate that into IETF speak, "The protocol isn't finished when you have
nothing more to add - it's finished when you have nothing more to take
away."

I believe these principles are very relevant to the current discussion.
There are mechanisms in sip-privacy-04 (whose motivation is at best obscure,
at worst non-existent) that allow sip-privacy-04 to be abused - and not just
a 'potential', this has already happened; we have concrete examples of
customers of sip-privacy-04 who designed architectures around the abuse of
the document. I have argued that sip-privacy-04 has unnecessarily large
number of moving parts, which makes its eventual usage in the field
difficult to predict (already it's been suggested for unpopular uses that
weren't anticipated), and little internal justification for these systems.
By removing these potentially non-essential mechanisms we could close the
door on some of these abuses without, I think, impairing the system's
ability to operate. The Iraqi thread is essentially about this last point.

> 
> Bill already explained how malicious call trace works - you will not be
able to
> call that party back.
> 

Maybe the draft should explain it rather than Bill.

------ Next note

[snip, condensing this down to the basics]

> ... this is completely bogus.

> ... where you then
> show up with a laundry list of items you want changed, and in most cases
not
> because there is anything technically wrong, but simply because you
personally   
> disagree with them... 

> I will furthermore also note,
> that you have not managed to produce a lot of support for your positions.
I
> think this goes a long way towards deciding what to do with some of your
> comments.

> Can it get any more hypocritical than this ?

> ... I'm
> becoming less and less convinced that you are interested in anything but
killing
> this draft.

Forgive me for not making a detailed answer to the points in this note,
though I'm sure the above citations will make it clear why I abstain.

I did want to speak to the consensus issue, though. I think most people are
monumentally indifferent to this draft, and probably the whole privacy
problem, and are looking for the fastest way out of this. Consensus is
impossible to gauge when nobody cares. That leaves us to debate the
technical merits (currently, the Iraqi thread). We should probably stick to
that rather than metadiscussion like the above.

> -- Flemming
> 
> --
> Flemming Andreasen
> Cisco Systems
> 
> 
> 

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From daemon@optimus.ietf.org  Tue Apr  2 10:53:51 2002
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More below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Monday, April 01, 2002 1:57 PM
> To: Peterson, Jon
> Cc: 'William Marshall'; 'sip@ietf.org'
> Subject: Re: FW: [Sip] Comment, SIP Privacy draft
> 
> 
> "Peterson, Jon" wrote:
> 
[snip]
> Certainly, but call trace may still work. Just becuase the information was
> untrusted does not mean it is false. If the Iraqi phone company keeps
sending us
> false information, we can always put appropriate procedures in place to
deal
> with that.

Okay, so it's not Caller-ID any more, now it's malicious call trace. I don't
immediately see how this is a workable motivation either, though.

In the past, you have said of the Iraqi network example:
> ... Now it just so happens that I
> can't (and won't) want to make a distiction between an untrusted user and
an
> untrusted something else, because it doesn't make any sense to me.

I agree with this, and I think it is at the crux of the matter at hand.
Trusted entities must consider a source from which they receive messages to
either be trusted or untrusted. The way the mechanism currently works, if
the originator of a request is untrusted, trust recipients add 'screen=no'
to any RPIDs in the request. Right? 

Okay, so now we take a user, Carol, and the Iraqi network. Both are
untrusted. Both send request to a trusted proxy, and both requests contain
an RPID. The trusted proxy adds 'screen=no' to each and forwards the
requests along. What fact about the requests, exactly, tells downstream
entities to 'kinda' trust the Iraqi RPID for malicious call trace, but not
to trust Carol's RPID? 

You go on to say:

[snip]
> 
> It is correct that Remote-Party-Id does not provide for non-repudiation,
but if
> you really want that, you have a hard problem on your hand. Present day
> telephony and data service provide trace solutions without any resemblance
of
> such guarantees and I suspect they will continue that way for a while.
> 

What this essentially means, I think, is that although you might not trust
the policies of the Iraqi network to ascertain identity, you still have to
treat it as if it is trusted, because well, that's the degree of certainty
with which the telephone network actually operates. What this means to me is
that you will configure your trusted proxies to treat requests coming from
the Iraqi proxy as if they are trusted - you should not set 'screen=no' on
these requests.

While I don't think Iraqi call trace provides any motivation for keeping
'screen=no' in the draft, this may suggest a requirement for somehow
describing the authentication policy that was used to ascertain someone's
identity (so that downstream entities can determine the degree to which they
should trust a particular RPID). Going down that road, however, would take
us far beyond the ISUP 'presentation restriction' concept, and certainly far
beyond anything in sip-privacy-04 today.

If we do, in the course of this discussion, eventually uncover a need for
'screen=no', I certainly hope it will be documented in sip-privacy-05.

> > So who will be using
> > the untrusted RPID? I see no reason why the answer isn't 'no one', and
that
> > seems to suggest to me that untrusted RPIDs can be removed at the trust
> > boundary.
> 
> As I have explained many times already, untrusted does not equal useless
in my
> philosophy book. Furthermore, if you remove it, you can't provide a more
secure
> solution (read: crypto signature) in the future for this mechanism without
> having everybody being able to verify such signatures. Any intermediary
not able
> to do that will will impose it's ugly policy on you and remove it even
though
> the signature could perhaps be verified by somebody else.
> 

I think you for some reason believe I am committed to the position that
signed RPIDs should be removed as 'untrusted'. In my last mail, I suggested
that signatures would render obsolete the concept of intrinsically trusted
or untrusted RPIDs. In this architecture there would be no role needed for
border elements that either set the screen parameter or remove RPIDs. That
whole concept goes away.

I think this is a place where we actually agree, but are using different
semantics. When you say:

[snip] 
> If you have a signature on a Remote-Party-Id, you don't want to
> remove an untrusted Remote-Party-Id since a downstream entity can verify
the
> signature to determine authenticity.
> 

... I say, "But it isn't 'untrusted' anymore if it has a signature; the
concept of being intrinsically 'untrusted' is no longer meaningful." The end
result is the same - we agree that if an RPID is signed, it shouldn't be
removed.

> 
> If you had paid any attention to previous (and even fairly recent)
discussions
> about this draft, you would know that it used to have a signature
mechanism, but
> it was noted that it was a special case of a general security problem.
General
> problems should have general solutions, and hence we are waiting for the
> security gurus to come up with that general security solution which could,
among
> other things, sign Remote-Party-Ids. This consensus has been formed long
time
> ago and was repeated recently.
> 

Agreed that it isn't sip-privacy-04's job to define how messages are signed,
it is a general security problem. But I think we also probably agree that
the requirements sip-privacy-04 attempts to fulfill will only be partially
realized without that signing mechanism. 

I'm not arguing (now, anyway) that sip-privacy-04 should be delayed until
the awaited signing mechanism has been invented. All I was saying is that I
think the signing mechanism is completely unrelated to the motivation for
'screen'.

> -- Flemming
> 

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Dean wrote:

> 
> "The SIP protocol explicitly sends the From and To fields to the
> destination UA. Therefore the sending UA should only insert 
> information
> in the From and To fields that is intended to be delivered to 
> other UAs
> and potentially displayed to the user of the other UA. If the user of
> the UA wishes to remain anonymous, they must not insert any 
> information
> into the From or To fields which compromises this anonymity. This same
> caveat applies to every other aspect of SIP, including headers and
> bodies, except for those elements which are explicitly removed or
> encrypted by intermediate elements known to the UA."
> 

OK, so let's work with this and see where we get...

1) The UA may insert information which later becomes private due to the
operation of services within the network. The device that detects this must
either edit the From/To fields itself, or explicitly route to an Application
Server which can do this & anything else needed to provide the necessary
privacy.

Is everyone happy with this ? Can we descibe it somewhere ?

2) The From/To fields cannot be used for communication from UE to network
entities of identity information which the user wishes to be kept private
(no way to indicate this)

3GPP has a requirement for information of type (2). They had planned to use
RPID, since this was an element 'which is explicitly removed or encrypted by
intermediate elements known to the UA'. They cannot use the Authentication
mechanism, because in 3GPP this is only done at registration (with first hop
integrity used for subsequent transactions) and the information to transport
can change on a call by call basis.

So what should 3GPP use for this information ?

Regards,

Mark

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<P><FONT SIZE=3D2>Dean wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;The SIP protocol explicitly sends the =
From and To fields to the</FONT>
<BR><FONT SIZE=3D2>&gt; destination UA. Therefore the sending UA should =
only insert </FONT>
<BR><FONT SIZE=3D2>&gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; in the From and To fields that is intended to =
be delivered to </FONT>
<BR><FONT SIZE=3D2>&gt; other UAs</FONT>
<BR><FONT SIZE=3D2>&gt; and potentially displayed to the user of the =
other UA. If the user of</FONT>
<BR><FONT SIZE=3D2>&gt; the UA wishes to remain anonymous, they must =
not insert any </FONT>
<BR><FONT SIZE=3D2>&gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; into the From or To fields which compromises =
this anonymity. This same</FONT>
<BR><FONT SIZE=3D2>&gt; caveat applies to every other aspect of SIP, =
including headers and</FONT>
<BR><FONT SIZE=3D2>&gt; bodies, except for those elements which are =
explicitly removed or</FONT>
<BR><FONT SIZE=3D2>&gt; encrypted by intermediate elements known to the =
UA.&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>OK, so let's work with this and see where we =
get...</FONT>
</P>

<P><FONT SIZE=3D2>1) The UA may insert information which later becomes =
private due to the operation of services within the network. The device =
that detects this must either edit the From/To fields itself, or =
explicitly route to an Application Server which can do this &amp; =
anything else needed to provide the necessary privacy.</FONT></P>

<P><FONT SIZE=3D2>Is everyone happy with this ? Can we descibe it =
somewhere ?</FONT>
</P>

<P><FONT SIZE=3D2>2) The From/To fields cannot be used for =
communication from UE to network entities of identity information which =
the user wishes to be kept private (no way to indicate this)</FONT></P>

<P><FONT SIZE=3D2>3GPP has a requirement for information of type (2). =
They had planned to use RPID, since this was an element 'which is =
explicitly removed or encrypted by intermediate elements known to the =
UA'. They cannot use the Authentication mechanism, because in 3GPP this =
is only done at registration (with first hop integrity used for =
subsequent transactions) and the information to transport can change on =
a call by call basis.</FONT></P>

<P><FONT SIZE=3D2>So what should 3GPP use for this information ?</FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Flemming Andreasen'"
	 <fandreas@cisco.com>
Cc: "'William Marshall'" <wtm@research.att.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: FW: [Sip] Comment, SIP Privacy draft
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Jon wrote:
> ... I say, "But it isn't 'untrusted' anymore if it has a 
> signature; the
> concept of being intrinsically 'untrusted' is no longer 
> meaningful." The end
> result is the same - we agree that if an RPID is signed, it 
> shouldn't be
> removed.
> 

I think if I sign an RPID with a signature that noone recognises, then this
is semantically equivalent to 'screen=no'.

Are you suggesting that a signed RPID should be discarded by the UAS if it
does not recognise the signature, or would it be displayed with a health
warning ?

...Mark

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<P><FONT SIZE=3D2>Jon wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; ... I say, &quot;But it isn't 'untrusted' =
anymore if it has a </FONT>
<BR><FONT SIZE=3D2>&gt; signature; the</FONT>
<BR><FONT SIZE=3D2>&gt; concept of being intrinsically 'untrusted' is =
no longer </FONT>
<BR><FONT SIZE=3D2>&gt; meaningful.&quot; The end</FONT>
<BR><FONT SIZE=3D2>&gt; result is the same - we agree that if an RPID =
is signed, it </FONT>
<BR><FONT SIZE=3D2>&gt; shouldn't be</FONT>
<BR><FONT SIZE=3D2>&gt; removed.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I think if I sign an RPID with a signature that noone =
recognises, then this is semantically equivalent to =
'screen=3Dno'.</FONT>
</P>

<P><FONT SIZE=3D2>Are you suggesting that a signed RPID should be =
discarded by the UAS if it does not recognise the signature, or would =
it be displayed with a health warning ?</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>, sip@ietf.org,
        Ben Campbell
	 <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "'William Marshall'" <wtm@research.att.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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This is an excellent summary, one with which I agree on almost all
particulars.
 
One or two small points below.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, April 02, 2002 6:05 AM
To: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon; 'William
Marshall'
Subject: Summary of RE: [Sip] Comment, SIP Privacy draft

<snip> 


3) 'Untrustworthy RPID' 

Flemming, Bill et al argued that RPID information is still useful when it is
'untrustworthy'. This can occur legitimately within the draft when an RPID
which is not 'private' (and so not encrypted) crosses a trust boundary. This
would represent an identity asserted by Network A and passed to Network B,
when B has no explicit trust relationship with A.

The contention was that since this information still has some value, it
should be passed to the UAS for display, and might also have some
application for call trace, although obviously not as useful as a
'trustworthy' identity.

Jon argued that the From field provides well enough for display, and the
value of untrustworthy information for call trace is debatable (and has been
debated at length).
[Peterson, Jon] Of greater concern to me is whether or not the existing
'screen' mechanism provides a way to use untrustworthy information for call
trace. If it does so (I still don't see how it does), then at the very least
I don't believe that usage is substantiated by the draft today.

There is agreement that some form of 'signed RPID' would be better.
Previously we have agreed on the need for a general mechanism to sign
information in SIP headers. Flemming has pointed out that if you agree to
have 'Signed RPID' then as a subset you have 'Untrustworthy RPID' (when you
don't recognise the signature). So removing 'Untrustworthy RPID' now would
be inconsistent, since there is already agreement to re-introduce it later.
That is unless you feel that the time interval during which Untrusted RPID
is not standardised is important.
[Peterson, Jon] I've argued (in a separate mail) that disallowing
'untrustworthy RPIDs' is not at all inconsistent with signing messages
later. Signing messages merely makes the concept of intrinsically trusted or
untrusted RPIDs go away - the concept of an intrinsically untrusted RPID is
not reintroduced by the ability to sign messages. Signing RPIDs just ushers
in an era of relativism, in which every entity is empowered to make its own
decision about trusting an RPID or not.


Many other points were made, but the above seemed to me to be the key
issues. 

And, now, my opinions: 

1) Call-Info: I think the point has been made here - we're into judgement
calls on how bad an abuse X is compared to Y - I would rather there was no
need for people to abuse the protocol at all.
[Peterson, Jon] Concisely put and undoubtedly accurate. 

2) From/To: RPID is not the problem here, is it a solution to an underlying
problem with From/To. Blocking RPID will not solve the underlying problem,
and people will just route around to find another, probably worse, solution.
Better to address the real problem which has led people to propose
obfuscating From/To in the first place. If this problem is solved, then
there is no reason for people to want untrusted UAs to insert RPID. See
separate mail on this.
[Peterson, Jon] Agreed. 

3) Untrustworthy RPID: I think Flemming and Bill have demonstrated the
utility of 'untrustworthy identity', and indeed this is what the From field
already provides. But From just provides the users assertion. If a network
has asserted additional identity information, why should the user should be
denied this ?
[Peterson, Jon] This is a line of reasoning we seem to come back to
repeatedly that really confuses me. RPID exists to provide call trace, not
Caller-ID, right? It allows the network to carry an identity when the user
wishes to remain anonymous, when they don't want to display their identity
to the other party in the dialog. If the network asserts identity
information, I don't think that information is intended to be displayed to
the user - so the user should actually be denied this information, right?

Having said that, the propogation of 'untrustworthy' RPID is strictly out of
scope according to the (retro-fitted) Applicability Statement. But then we
have agreement to add it 'in future' by virtue of the generic signing
mechanism - I guess this inconsistency is one for the ADs.

Regards, 

Mark 


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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=385274615-02042002>This 
is an excellent summary, one with which I agree on almost all 
particulars.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=385274615-02042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=385274615-02042002>One or 
two small points below.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=385274615-02042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=385274615-02042002>Jon 
Peterson</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=385274615-02042002>NeuStar, Inc.</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma><FONT 
  size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Tuesday, April 02, 2002 
  6:05 AM<BR><B>To:</B> sip@ietf.org; Ben Campbell; Flemming Andreasen; 
  Peterson, Jon; 'William Marshall'<BR><B>Subject:</B> Summary of RE: [Sip] 
  Comment, SIP Privacy draft<BR><BR><FONT color=#0000ff></FONT><FONT 
  size=2><FONT face=Arial><SPAN 
  class=385274615-02042002>&lt;snip&gt;&nbsp;</SPAN><BR></FONT></FONT></FONT></FONT></DIV>
  <P><FONT size=2>3) 'Untrustworthy RPID'</FONT> </P>
  <P><FONT size=2>Flemming, Bill et al argued that RPID information is still 
  useful when it is 'untrustworthy'. This can occur legitimately within the 
  draft when an RPID which is not 'private' (and so not encrypted) crosses a 
  trust boundary. This would represent an identity asserted by Network A and 
  passed to Network B, when B has no explicit trust relationship with 
  A.</FONT></P>
  <P><FONT size=2>The contention was that since this information still has some 
  value, it should be passed to the UAS for display, and might also have some 
  application for call trace, although obviously not as useful as a 
  'trustworthy' identity.</FONT></P>
  <P><FONT size=2>Jon argued that the From field provides well enough for 
  display, and the value of untrustworthy information for call trace is 
  debatable (and has been debated at length).<BR><FONT color=#0000ff 
  face=Arial><SPAN class=385274615-02042002>[Peterson, Jon]&nbsp;Of&nbsp;greater 
  concern to me is whether or not the existing 'screen' mechanism provides a way 
  to use untrustworthy information for call trace. If it does so (I still don't 
  see how it does), then at the very least I don't believe that usage is 
  substantiated by the draft today.</SPAN></FONT></FONT></P>
  <P><FONT size=2>There is agreement that some form of 'signed RPID' would be 
  better. Previously we have agreed on the need for a general mechanism to sign 
  information in SIP headers. Flemming has pointed out that if you agree to have 
  'Signed RPID' then as a subset you have 'Untrustworthy RPID' (when you don't 
  recognise the signature). So removing 'Untrustworthy RPID' now would be 
  inconsistent, since there is already agreement to re-introduce it later. That 
  is unless you feel that the time interval during which Untrusted RPID is not 
  standardised is important.<BR><FONT color=#0000ff face=Arial><SPAN 
  class=385274615-02042002>[Peterson, Jon]&nbsp;I've argued&nbsp;(in a separate 
  mail) that&nbsp;disallowing 'untrustworthy RPIDs' is not at all inconsistent 
  with signing messages later. Signing messages merely makes the&nbsp;concept 
  of&nbsp;intrinsically trusted or untrusted RPIDs go away - the concept of an 
  intrinsically untrusted RPID is not reintroduced by the ability to sign 
  messages. Signing RPIDs just ushers in an era of relativism, in which every 
  entity is empowered to make its own decision about trusting an RPID or 
  not.</SPAN></FONT></FONT></P><BR>
  <P><FONT size=2>Many other points were made, but the above seemed to me to be 
  the key issues.</FONT> </P>
  <P><FONT size=2>And, now, my opinions:</FONT> </P>
  <P><FONT size=2>1) Call-Info: I think the point has been made here - we're 
  into judgement calls on how bad an abuse X is compared to Y - I would rather 
  there was no need for people to abuse the protocol at all.<BR><FONT 
  color=#0000ff face=Arial><SPAN class=385274615-02042002>[Peterson, 
  Jon]&nbsp;Concisely put and undoubtedly 
  accurate.&nbsp;</SPAN></FONT></FONT></P>
  <P><FONT size=2>2) From/To: RPID is not the problem here, is it a solution to 
  an underlying problem with From/To. Blocking RPID will not solve the 
  underlying problem, and people will just route around to find another, 
  probably worse, solution. Better to address the real problem which has led 
  people to propose obfuscating From/To in the first place. If this problem is 
  solved, then there is no reason for people to want untrusted UAs to insert 
  RPID. See separate mail on this.<BR><FONT color=#0000ff face=Arial><SPAN 
  class=385274615-02042002>[Peterson, 
  Jon]&nbsp;Agreed.&nbsp;</SPAN></FONT></FONT></P>
  <P><FONT size=2>3) Untrustworthy RPID: I think Flemming and Bill have 
  demonstrated the utility of 'untrustworthy identity', and indeed this is what 
  the From field already provides. But From just provides the users assertion. 
  If a network has asserted additional identity information, why should the user 
  should be denied this ?<BR><FONT color=#0000ff face=Arial><SPAN 
  class=385274615-02042002>[Peterson, Jon]&nbsp;This is a line of reasoning we 
  seem to&nbsp;come back to repeatedly that really confuses me. RPID exists to 
  provide&nbsp;call trace, not&nbsp;Caller-ID, right? It allows the network to 
  carry an identity when the user wishes to remain anonymous, when they don't 
  want to display their identity to the other party in the dialog. If the 
  network asserts identity information, I don't think that information is 
  intended to be displayed to the user - so the&nbsp;user should actually be 
  denied this information, right?</SPAN></FONT></FONT></P>
  <P><FONT size=2>Having said that, the propogation of 'untrustworthy' RPID is 
  strictly out of scope according to the (retro-fitted) Applicability Statement. 
  But then we have agreement to add it 'in future' by virtue of the generic 
  signing mechanism - I guess this inconsistency is one for the ADs.</FONT></P>
  <P><FONT size=2>Regards,</FONT> </P>
  <P><FONT size=2>Mark</FONT> </P></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Tue Apr  2 11:50:13 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>, sip@ietf.org,
        Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen
	 <fandreas@cisco.com>,
        "'William Marshall'" <wtm@research.att.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 2 Apr 2002 17:31:54 +0100 
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> > 3) Untrustworthy RPID: I think Flemming and Bill have demonstrated the
utility of
> > 'untrustworthy identity', and indeed this is what the From field already
provides. But From
> > just provides the users assertion. If a network has asserted additional
identity information, 
> > why should the user should be denied this ?
> [Peterson, Jon] This is a line of reasoning we seem to come back to
repeatedly that really 
> confuses me. RPID exists to provide call trace, not Caller-ID, right? It
allows the network to 
> carry an identity when the user wishes to remain anonymous, when they
don't want to display 
> heir identity to the other party in the dialog. If the network asserts
identity information, I 
> don't think that information is intended to be displayed to the user - so
the user should 
> actually be denied this information, right?

I think there is an ambition to use RPID for Caller ID as well as for Call
Trace. In particular, when interworking with the PSTN, the CLI needs to be
Network asserted - user asserted identity in the From field will not do.
This CLI is then used for both call trace and caller display.

Just as in the PSTN, displaying the network asserted identity at a SIP UA
will give the called party greater confidence in the identity - at least
their confidence should be as great as the confidence they have in the
network entity from which they received the message.

Alternatively, you could provide a way to indicate that the value in the
From field really does represent the identity of the caller (i.e. is valid
for the user who authenticated). Either the proxy ascertaining this could
sign it, or within a trust domain, it could be passed forwards as a simple
indication attached to the From field. Somehow I'm not sure you'll like
this!

...Mark


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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<BODY>
<BR>

<P><FONT SIZE=3D2>&gt; &gt; 3) Untrustworthy RPID: I think Flemming and =
Bill have demonstrated the utility of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 'untrustworthy identity', and indeed this =
is what the From field already provides. But From</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; just provides the users assertion. If a =
network has asserted additional identity information, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; why should the user should be denied this =
?</FONT>
<BR><FONT SIZE=3D2>&gt; [Peterson, Jon] This is a line of reasoning we =
seem to come back to repeatedly that really </FONT>
<BR><FONT SIZE=3D2>&gt; confuses me. RPID exists to provide call trace, =
not Caller-ID, right? It allows the network to </FONT>
<BR><FONT SIZE=3D2>&gt; carry an identity when the user wishes to =
remain anonymous, when they don't want to display </FONT>
<BR><FONT SIZE=3D2>&gt; heir identity to the other party in the dialog. =
If the network asserts identity information, I </FONT>
<BR><FONT SIZE=3D2>&gt; don't think that information is intended to be =
displayed to the user - so the user should </FONT>
<BR><FONT SIZE=3D2>&gt; actually be denied this information, =
right?</FONT>
</P>

<P><FONT SIZE=3D2>I think there is an ambition to use RPID for Caller =
ID as well as for Call Trace. In particular, when interworking with the =
PSTN, the CLI needs to be Network asserted - user asserted identity in =
the From field will not do. This CLI is then used for both call trace =
and caller display.</FONT></P>

<P><FONT SIZE=3D2>Just as in the PSTN, displaying the network asserted =
identity at a SIP UA will give the called party greater confidence in =
the identity - at least their confidence should be as great as the =
confidence they have in the network entity from which they received the =
message.</FONT></P>

<P><FONT SIZE=3D2>Alternatively, you could provide a way to indicate =
that the value in the From field really does represent the identity of =
the caller (i.e. is valid for the user who authenticated). Either the =
proxy ascertaining this could sign it, or within a trust domain, it =
could be passed forwards as a simple indication attached to the From =
field. Somehow I'm not sure you'll like this!</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
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Dean Willis wrote:
> 
> Keith said:
> 
> 
>>I would prefer to keep Path, or at least something shorter 
>>that the mouthfull that is currently proposed.
> 
> 
> I suspect RRR would get used a lot . . .

It's a minor point of course, but I agree with Keith. Also, I suspect 
that for most implementations the decision of whether or not to use the 
short form is not made on a header by header basis, and so you're 
actually quite likely to see the long form most of the time.

Anders


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From daemon@optimus.ietf.org  Tue Apr  2 12:13:58 2002
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        "Rosen, Brian" <Brian.Rosen@marconi.com>,
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Subject: Re: [Sip] Comment, SIP Privacy draft
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"Peterson, Jon" wrote:

> I like Ben's way of putting it - that my argument is philosophical. The
> philosophical part is "pluralitas non est ponenda sine necessitate". Or to
> translate that into IETF speak, "The protocol isn't finished when you have
> nothing more to add - it's finished when you have nothing more to take
> away."
>

Humbug - how does one apply that to philosophy ? Besides, as recent discussions
has exemplified, this has hardly been a guiding principle so far, so why now all
of a sudden for this draft ?


> I believe these principles are very relevant to the current discussion.
> There are mechanisms in sip-privacy-04 (whose motivation is at best obscure,
> at worst non-existent)

Well, I have provided with you reasons several times. Mark has just summarized
them this morning. As for your judgement of these, you can only speak for
yourself, but I have yet to see a lot of support for your position.


>
> [snip, condensing this down to the basics]
>
> > ... this is completely bogus.
>
> > ... where you then
> > show up with a laundry list of items you want changed, and in most cases
> not
> > because there is anything technically wrong, but simply because you
> personally
> > disagree with them...
>
> > I will furthermore also note,
> > that you have not managed to produce a lot of support for your positions.
> I
> > think this goes a long way towards deciding what to do with some of your
> > comments.
>
> > Can it get any more hypocritical than this ?
>
> > ... I'm
> > becoming less and less convinced that you are interested in anything but
> killing
> > this draft.
>
> Forgive me for not making a detailed answer to the points in this note,
> though I'm sure the above citations will make it clear why I abstain.
>

No they don't - the comments still stand and you have done nothing to convince
me otherwise. The fact that I have to continue repeating the same points over
and over again and you have failed to produce much support for your positions
only reaffirms my impression.


> I did want to speak to the consensus issue, though. I think most people are
> monumentally indifferent to this draft, and probably the whole privacy
> problem, and are looking for the fastest way out of this.

Probably true. The people that care have contributed throughout the process and
hence have already shaped the draft. The people that chose not to participate,
well at least most of them recognize that there has been plenty of opportunity
to participate, and hence have the decency to not try and derail the whole thing
at the very last minute with a long list of more or less philosophical
objections.


> Consensus is
> impossible to gauge when nobody cares. That leaves us to debate the
> technical merits (currently, the Iraqi thread). We should probably stick to
> that rather than metadiscussion like the above.
>

Well, we are not getting anywhere on that obviously. WG Last Call has completed
(a while ago actually), and I think it is in order for our chairs to gauge
consensus at this point and ensure timely progress in accordance with the
Working Group procedures defined in RFC 2418 - hopefully we don't need to
proceed to the RFC 2026 process. So far, I have only seen persistent objections
from you which would hardly seem to counter rough consensus.

-- Flemming




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From daemon@optimus.ietf.org  Tue Apr  2 12:33:37 2002
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snip snip ...

"Peterson, Jon" wrote:

> In the past, you have said of the Iraqi network example:
> > ... Now it just so happens that I
> > can't (and won't) want to make a distiction between an untrusted user and
> an
> > untrusted something else, because it doesn't make any sense to me.
>
> I agree with this, and I think it is at the crux of the matter at hand.
> Trusted entities must consider a source from which they receive messages to
> either be trusted or untrusted. The way the mechanism currently works, if
> the originator of a request is untrusted, trust recipients add 'screen=no'
> to any RPIDs in the request. Right?

Correct.

>
> Okay, so now we take a user, Carol, and the Iraqi network. Both are
> untrusted. Both send request to a trusted proxy, and both requests contain
> an RPID. The trusted proxy adds 'screen=no' to each and forwards the
> requests along. What fact about the requests, exactly, tells downstream
> entities to 'kinda' trust the Iraqi RPID for malicious call trace, but not
> to trust Carol's RPID?
>

Nobody has argued that an untrusted Remote-Party-Id is a reliable way of
performing call trace. However, if the Iraqi network is actually cooperating
with us, the trace will be performed. Had we removed instead, there would be
absolutely no tracing going on. Reliable and trustworthy - no . Better than
nothing - yes.


>
> You go on to say:
>
> [snip]
> >
> > It is correct that Remote-Party-Id does not provide for non-repudiation,
> but if
> > you really want that, you have a hard problem on your hand. Present day
> > telephony and data service provide trace solutions without any resemblance
> of
> > such guarantees and I suspect they will continue that way for a while.
> >
>
> What this essentially means, I think, is that although you might not trust
> the policies of the Iraqi network to ascertain identity, you still have to
> treat it as if it is trusted, because well, that's the degree of certainty
> with which the telephone network actually operates. What this means to me is
> that you will configure your trusted proxies to treat requests coming from
> the Iraqi proxy as if they are trusted - you should not set 'screen=no' on
> these requests.
>

No, that's not what I said. I'm setting "screen=no" becuase I dont' trust it,
but if you request a call trace nevertheless, I will still try and do it since
that's the best we've got.

> I think you for some reason believe I am committed to the position that
> signed RPIDs should be removed as 'untrusted'. In my last mail, I suggested
> that signatures would render obsolete the concept of intrinsically trusted
> or untrusted RPIDs. In this architecture there would be no role needed for
> border elements that either set the screen parameter or remove RPIDs. That
> whole concept goes away.
>
> I think this is a place where we actually agree, but are using different
> semantics.

No - I think you haven't understood the point. A proxy that only uses the
mechanisms defined in the draft will remove a Remote-Party-Id it either doesn't
trust or can't verify. A (future) signature mechanism that can be associated
with a Remote-Party-Id is of no relevance to this proxy since it doesn't
understand it. Thus, despite the fact that it may have been a perfectly valid
Remote-Party-Id, as the signature indicates, the proxy will remove it anyway if
we follow your suggestion.

-- Flemming


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From: Flemming Andreasen <fandreas@cisco.com>
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        "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: FW: [Sip] Comment, SIP Privacy draft
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Mark Watson wrote:

>
>
> Jon wrote:
> > ... I say, "But it isn't 'untrusted' anymore if it has a
> > signature; the
> > concept of being intrinsically 'untrusted' is no longer
> > meaningful." The end
> > result is the same - we agree that if an RPID is signed, it
> > shouldn't be
> > removed.
> >
>
> I think if I sign an RPID with a signature that noone recognises, then
> this is semantically equivalent to 'screen=no'.
>

Good point (self-signed certificates come to mind as well). However,
let's not turn this into a crypto-discussion. We already have consensus
that this is not a requirement for the privacy draft.

-- Flemming



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CC: "'Peterson, Jon'" <jon.peterson@neustar.biz>, sip@ietf.org,
        Ben Campbell <bcampbell@dynamicsoft.com>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mark Watson wrote:

>
>
> > > 3) Untrustworthy RPID: I think Flemming and Bill have demonstrated
> the utility of
> > > 'untrustworthy identity', and indeed this is what the From field
> already provides. But From
> > > just provides the users assertion. If a network has asserted
> additional identity information,
> > > why should the user should be denied this ?
> > [Peterson, Jon] This is a line of reasoning we seem to come back to
> repeatedly that really
> > confuses me. RPID exists to provide call trace, not Caller-ID,
> right? It allows the network to
> > carry an identity when the user wishes to remain anonymous, when
> they don't want to display
> > heir identity to the other party in the dialog. If the network
> asserts identity information, I
> > don't think that information is intended to be displayed to the user
> - so the user should
> > actually be denied this information, right?
>
> I think there is an ambition to use RPID for Caller ID as well as for
> Call Trace.

Right - the second paragraph in Section 4 even explains this.

-- Flemming



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Date: Tue, 2 Apr 2002 09:45:33 -0800 (Pacific Standard Time)
From: Rohan Mahy <rohan@cisco.com>
To: Mark Watson <mwatson@nortelnetworks.com>
cc: "'Dean Willis'" <dean.willis@softarmor.com>, <sip@ietf.org>
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy
 draft)
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On Tue, 2 Apr 2002, Mark Watson wrote:

> Dean wrote:
>
> >
> > "The SIP protocol explicitly sends the From and To fields to the
> > destination UA. Therefore the sending UA should only insert
> > information
> > in the From and To fields that is intended to be delivered to
> > other UAs
> > and potentially displayed to the user of the other UA. If the user of
> > the UA wishes to remain anonymous, they must not insert any
> > information
> > into the From or To fields which compromises this anonymity. This same
> > caveat applies to every other aspect of SIP, including headers and
> > bodies, except for those elements which are explicitly removed or
> > encrypted by intermediate elements known to the UA."
> >
>
> OK, so let's work with this and see where we get...
>
> 1) The UA may insert information which later becomes private due to the
> operation of services within the network. The device that detects this must
> either edit the From/To fields itself, or explicitly route to an Application
> Server which can do this & anything else needed to provide the necessary
> privacy.
>
> Is everyone happy with this ? Can we descibe it somewhere ?
>
> 2) The From/To fields cannot be used for communication from UE to network
> entities of identity information which the user wishes to be kept private
> (no way to indicate this)
>
> 3GPP has a requirement for information of type (2). They had planned to use
> RPID, since this was an element 'which is explicitly removed or encrypted by
> intermediate elements known to the UA'. They cannot use the Authentication
> mechanism, because in 3GPP this is only done at registration

It seems that 3GPP could use an unsolicited Proxy-Authorization header
with the correct public identity identified as the username.

thanks,
-rohan

> (with first hop
> integrity used for subsequent transactions) and the information to transport
> can change on a call by call basis.
>
> So what should 3GPP use for this information ?
>
> Regards,
>
> Mark
>


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From daemon@optimus.ietf.org  Tue Apr  2 13:09:03 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Flemming Andreasen'" <fandreas@cisco.com>
Cc: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'William Marshall'"
	 <wtm@research.att.com>,
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Subject: RE: FW: [Sip] Comment, SIP Privacy draft
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Flemming wrote:


> > Jon wrote:
> > > ... I say, "But it isn't 'untrusted' anymore if it has a
> > > signature; the
> > > concept of being intrinsically 'untrusted' is no longer
> > > meaningful." The end
> > > result is the same - we agree that if an RPID is signed, it
> > > shouldn't be
> > > removed.
> > >
> >
> > I think if I sign an RPID with a signature that noone 
> recognises, then
> > this is semantically equivalent to 'screen=no'.
> >
> 
> Good point (self-signed certificates come to mind as well). However,
> let's not turn this into a crypto-discussion. We already have 
> consensus
> that this is not a requirement for the privacy draft.

Actually, I thought I was just restating a point you made in an earlier
mail: If we're fine with proxies passing on signed RPIDs where they don't
recognise the signature, we should also be fine with proxies passing on
untrustworthy RPIDs i.e. those with 'screen=no', because the two are
equivalent.

The important point, then, is what we expect *UAs* to do with such things.
Jon's contention was, I think, that (effectively) that UAs should discard
them. Your contention was that the (human) user should decide what meaning
if any to attach to the information. If we're being consistent we should
have the same answer for 'screen=no' RPIDs as we have for signed RPIDs where
we don't recognise the signature.

...Mark

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<TITLE>RE: FW: [Sip] Comment, SIP Privacy draft</TITLE>
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<P><FONT SIZE=3D2>Flemming wrote:</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; &gt; Jon wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; ... I say, &quot;But it isn't =
'untrusted' anymore if it has a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; signature; the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; concept of being intrinsically =
'untrusted' is no longer</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; meaningful.&quot; The end</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; result is the same - we agree that if =
an RPID is signed, it</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; shouldn't be</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; removed.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I think if I sign an RPID with a signature =
that noone </FONT>
<BR><FONT SIZE=3D2>&gt; recognises, then</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; this is semantically equivalent to =
'screen=3Dno'.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Good point (self-signed certificates come to =
mind as well). However,</FONT>
<BR><FONT SIZE=3D2>&gt; let's not turn this into a crypto-discussion. =
We already have </FONT>
<BR><FONT SIZE=3D2>&gt; consensus</FONT>
<BR><FONT SIZE=3D2>&gt; that this is not a requirement for the privacy =
draft.</FONT>
</P>

<P><FONT SIZE=3D2>Actually, I thought I was just restating a point you =
made in an earlier mail: If we're fine with proxies passing on signed =
RPIDs where they don't recognise the signature, we should also be fine =
with proxies passing on untrustworthy RPIDs i.e. those with =
'screen=3Dno', because the two are equivalent.</FONT></P>

<P><FONT SIZE=3D2>The important point, then, is what we expect *UAs* to =
do with such things. Jon's contention was, I think, that (effectively) =
that UAs should discard them. Your contention was that the (human) user =
should decide what meaning if any to attach to the information. If =
we're being consistent we should have the same answer for 'screen=3Dno' =
RPIDs as we have for signed RPIDs where we don't recognise the =
signature.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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From daemon@optimus.ietf.org  Tue Apr  2 13:16:26 2002
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From: "Cullen Jennings" <fluffy@cisco.com>
To: <gvelo@astartec.com>, <sip@ietf.org>
Subject: RE: [Sip] Residential UA security
Date: Tue, 2 Apr 2002 10:00:12 -0800
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It is an interesting question - it may not be solvable on the other hand it
may not need to solved as directly as you think.

With a modem set up as an auto dialer, I could make your current PSTN phone
unusable. I could flood your email box so you were likely over quota. I
could probably use up on the bandwidth on your WAN link. None of these DOS
issues seem to be a huge problem today. One reason is that there is some
hope you could trace back and figure out who was doing this to you then
perhaps filter or take legal action.

Cullen


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Gabriel
> Velo
> Sent: Tuesday, April 02, 2002 6:23 AM
> To: sip@ietf.org
> Subject: [Sip] Residential UA security
>
>
> Dear All.
>
> I need some help on this topic:
>
> How can i protect a residential UA (my residential Sip Phone) from DOS
> Attack like a flood of INVITE msg ??
> In this scenario we don't have a validation since we need receive
> call from
> anyone.
> and .... anyone with a low bandwidth connection can make than my
> phone can't
> stop of ring !!
>
> Thanks in advance.
>
> Gabriel.
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From daemon@optimus.ietf.org  Tue Apr  2 13:22:52 2002
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Anders Kristensen" <akristensen@dynamicsoft.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "'Drage, Keith \(Keith\)'" <drage@lucent.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
Date: Tue, 2 Apr 2002 10:05:03 -0800
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If this type of bandwidth matters to you, then you will compress and with
any half decent compression scheme the representation of the RRR token is
going to be the same no matter how long the uncompressed version is. Given
the length and design of SIP messages, it seems sort of futile to try and
optimize message size by any method other than compression at this point. I
don't care what we call it, but I don't think size matters.

Cullen

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Anders
> Kristensen
> Sent: Tuesday, April 02, 2002 8:45 AM
> To: Dean Willis
> Cc: 'Drage, Keith (Keith)'; sip@ietf.org
> Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D
> ACTION:draft-illis...
>
>
>
>
> Dean Willis wrote:
> >
> > Keith said:
> >
> >
> >>I would prefer to keep Path, or at least something shorter
> >>that the mouthfull that is currently proposed.
> >
> >
> > I suspect RRR would get used a lot . . .
>
> It's a minor point of course, but I agree with Keith. Also, I suspect
> that for most implementations the decision of whether or not to use the
> short form is not made on a header by header basis, and so you're
> actually quite likely to see the long form most of the time.
>
> Anders
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "David R. Oran" <oran@cisco.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, <gvelo@astartec.com>,
        <sip@ietf.org>
Subject: RE: [Sip] Residential UA security
Date: Tue, 2 Apr 2002 13:03:16 -0500
Organization: Cisco Systems
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Another possibility is to set your UA to only accept invites from a
known set of previous-hop proxies that you trust and are configured to
execute your chosen call screening policy.

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Cullen Jennings
> Sent: Tuesday, April 02, 2002 1:00 PM
> To: gvelo@astartec.com; sip@ietf.org
> Subject: RE: [Sip] Residential UA security
> 
> 
> 
> It is an interesting question - it may not be solvable on the 
> other hand it may not need to solved as directly as you think.
> 
> With a modem set up as an auto dialer, I could make your 
> current PSTN phone unusable. I could flood your email box so 
> you were likely over quota. I could probably use up on the 
> bandwidth on your WAN link. None of these DOS issues seem to 
> be a huge problem today. One reason is that there is some 
> hope you could trace back and figure out who was doing this 
> to you then perhaps filter or take legal action.
> 
> Cullen
> 
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of 
> > Gabriel Velo
> > Sent: Tuesday, April 02, 2002 6:23 AM
> > To: sip@ietf.org
> > Subject: [Sip] Residential UA security
> >
> >
> > Dear All.
> >
> > I need some help on this topic:
> >
> > How can i protect a residential UA (my residential Sip 
> Phone) from DOS 
> > Attack like a flood of INVITE msg ?? In this scenario we 
> don't have a 
> > validation since we need receive call from
> > anyone.
> > and .... anyone with a low bandwidth connection can make than my
> > phone can't
> > stop of ring !!
> >
> > Thanks in advance.
> >
> > Gabriel.
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr  2 13:41:07 2002
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Date: Tue, 02 Apr 2002 13:17:53 -0500
From: Anders Kristensen <akristensen@dynamicsoft.com>
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To: Cullen Jennings <fluffy@cisco.com>
CC: Dean Willis <dean.willis@softarmor.com>,
        "'Drage, Keith (Keith)'"
 <drage@lucent.com>, sip@ietf.org
Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D ACTION:draft-illis...
References: <DLEHICEBMNEIPCACNLPCKELCCCAA.fluffy@cisco.com>
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I would argue that on the contrary we know that size *does* matter. 
Compression is great but you can't assume it's available everywhere.

Anders

Cullen Jennings wrote:
> If this type of bandwidth matters to you, then you will compress and with
> any half decent compression scheme the representation of the RRR token is
> going to be the same no matter how long the uncompressed version is. Given
> the length and design of SIP messages, it seems sort of futile to try and
> optimize message size by any method other than compression at this point. I
> don't care what we call it, but I don't think size matters.
> 
> Cullen
> 
> 
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Anders
>>Kristensen
>>Sent: Tuesday, April 02, 2002 8:45 AM
>>To: Dean Willis
>>Cc: 'Drage, Keith (Keith)'; sip@ietf.org
>>Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D
>>ACTION:draft-illis...
>>
>>
>>
>>
>>Dean Willis wrote:
>>
>>>Keith said:
>>>
>>>
>>>
>>>>I would prefer to keep Path, or at least something shorter
>>>>that the mouthfull that is currently proposed.
>>>
>>>
>>>I suspect RRR would get used a lot . . .
>>
>>It's a minor point of course, but I agree with Keith. Also, I suspect
>>that for most implementations the decision of whether or not to use the
>>short form is not made on a header by header basis, and so you're
>>actually quite likely to see the long form most of the time.
>>
>>Anders
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Tue Apr  2 15:00:06 2002
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sip@ietf.org, "'Dean Willis'" <dean.willis@softarmor.com>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D A
	CTION:draft-illis...
Date: Tue, 2 Apr 2002 20:35:30 +0100 
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See below

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com

> ----------
> From: 	Dean Willis[SMTP:dean.willis@softarmor.com]
> Sent: 	02 April 2002 15:39
> To: 	'Drage, Keith (Keith)'; sip@ietf.org
> Subject: 	RE: [Sip] Header name in draft-willis-sip-path-02.txt, was:
> I-D ACTION:draft-illis...
> 
> 
> 
> Keith said:
> 
> > I would prefer to keep Path, or at least something shorter 
> > that the mouthfull that is currently proposed.
> 
> I suspect RRR would get used a lot . . .
> 
If it is allowed to be defined. Surely I remember a discussion on the list
some time back saying that no new short form codings would be defined? Also
none of them are more than one character, so do we now start inventing 3
character ones? Certainly none of the recent headers added seem to have
short forms.
>  
> > If you really must change it, then it should at least be 
> > hyphenated (twice) for consistency with the other headers.
> 
> Register-Record-Route ?
> 
Yes, this is consistent with other headers.
>  
> > Note also that it no longer does the same as Record-Route, as 
> > Record-Route is a bidirectional capability, and it was the 
> > same strong suggestions in Minneapolis that precluded it 
> > having that one and same functionality.
> 
> Actually, Record-Route entries are NOT changed in the response, just
> copied back towards the source. That is, they accumulate on the INVITE
> message, and are returned to the UA on the 200OK response. I think the
> text in "path" now has exactly the same behaviour. If it doesn't, I did
> something wrong, so please verify . . .
> 
The piece of funtionality I was referring to is the UA handling of the
response. At that point things do get different. Up to that point, I believe
the handling is identical.


> --
> Dean
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Tue, 02 Apr 2002 14:58:11 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Cullen Jennings <fluffy@cisco.com>
CC: gvelo@astartec.com, sip@ietf.org
Subject: Re: [Sip] Residential UA security
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Cullen Jennings wrote:
> 
> It is an interesting question - it may not be solvable on the other hand
> it
> may not need to solved as directly as you think.
> 
> With a modem set up as an auto dialer, I could make your current PSTN
> phone
> unusable. I could flood your email box so you were likely over quota. I
> could probably use up on the bandwidth on your WAN link. None of these
> DOS
> issues seem to be a huge problem today. One reason is that there is some
> hope you could trace back and figure out who was doing this to you then
> perhaps filter or take legal action.

For the PSTN, this kind of attack is not common because it costs money
to make a call. It doesn't cost money to send email, and thats why there
is spam.

That said, the anonymous authentication mechanism can prevent DDOS
attacks on my phone, but not a direct targeted attack from a single
source. 

-Jonathan R.

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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Christian Jansson wrote:
> 
> Supported header has lost its compact form 'k'.

No, it hasn't. THe BNF has the compact form listed. Its just not
mentioned in Section 20.37, and it probably should be listed there. I'll
note this.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From daemon@ns.ietf.org  Tue Apr  2 23:47:29 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>
Cc: <sip@ietf.org>
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy draft)
Date: Tue, 2 Apr 2002 22:18:34 -0600
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Mark wrote:

---------------
OK, so let's work with this and see where we get... 
1) The UA may insert information which later becomes private due to the
operation of services within the network. The device that detects this
must either edit the From/To fields itself, or explicitly route to an
Application Server which can do this & anything else needed to provide
the necessary privacy.
Is everyone happy with this ? Can we descibe it somewhere ? 
2) The From/To fields cannot be used for communication from UE to
network entities of identity information which the user wishes to be
kept private (no way to indicate this)
3GPP has a requirement for information of type (2). They had planned to
use RPID, since this was an element 'which is explicitly removed or
encrypted by intermediate elements known to the UA'. They cannot use the
Authentication mechanism, because in 3GPP this is only done at
registration (with first hop integrity used for subsequent transactions)
and the information to transport can change on a call by call basis.
So what should 3GPP use for this information ? 
Regards, 
Mark 
-------

1) Absolutely not. The information is what was inserted by the UA. It
cannot become private "somewhere else in the network" for that dialog.
The only way to change it is to make a new dialog. Or maybe that's what
you mean and I'm just being dense.

2) part 1: Yes! Part 2: 3GPP terminals had better be able to use
Authentication anytime it's required by a downstream system or they
aren't conformant to the SIP model as I understand it. Just because they
think they are only going to authenticate REGISTER doesn't mean they can
get away with not supporting it in the other usual transactions. So I'd
suggest that if IETF can come up with something like a third-party
authentication model that adresses the transitive identity expression
requirements, that  3GPP could figure out how to use it.

--
Dean


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From daemon@ns.ietf.org  Wed Apr  3 00:01:59 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Cc: <Rohan@cicso.com>, <jo@ipdialog.com>, <brian.rosen@marconi.com>
Date: Tue, 2 Apr 2002 22:38:43 -0600
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It looks like UPDATE and Manyfolks are likely to be dependent on the
Reason header mechanism (imho). Given this (and the other requirements
discussions that we've heard), how do you people feel about proposing
this as a SIP WG effort? And yes, the SIP WG can identify work that
needs to be done to support other SIP WG efforts without having a
third-party generate a requirements RFC.

--
Dean


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From daemon@ns.ietf.org  Wed Apr  3 02:40:34 2002
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Eric Tremblay wrote:
> 
> Ouch! This one fell through the cracks, and I'm surprised it did not
> generate more comments. I know it's kind of late to comment on that and
> I
> don't expect any changes, but I want to make sure I understand a few
> things.
> 
> 1- Bis06 and later won't interoperate with Bis05 and earlier if Bis05
> devices add ports to their From/To headers of their requests. (the port
> can
> get stripped in the response, causing a failure to match the response to
> the
> request). Our devices are placing the port number in From/To if the home
> server is listening to something else than 5060 :( Of course, if the
> Bis06+
> device echo the port in responses, everything should work fine.

Not clear what will happen, I suspect most will echo it. However, don't
put it in.

> 
> 2- I thought that when I was sending an INVITE to establish a dialog,
> the
> URL found in the From: could be used to call me back later on. This URL
> usually pointed to my home server. If this server has no SRV entries and
> if
> it is using a port different than 5060, there is no way for the user I
> have
> called to call me back. Am I right?  It should work fine as long as SRV
> records are used.

If you don't have SRV, AND if the server for the domain is not on 5060
(both of which are bad, by the way), then use Reply-To.

The perils of putting ports in the From field are well documented, and
that is what motivated this decision. It needs to be a logical
identifier, period.

Thanks,
Jonathan R.


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

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From daemon@ns.ietf.org  Wed Apr  3 02:43:54 2002
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Date: Wed, 03 Apr 2002 02:26:44 -0500
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To: Tom-PT Taylor <taylor@nortelnetworks.com>
CC: "'Vijay K. Gurbani'" <vkg@lucent.com>, Sean Olson <seancolson@yahoo.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>,
        sip <sip@ietf.org>
Subject: Re: [Sip] Thoughts on the reason header field scope
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Tom-PT Taylor wrote:
> 
> 183 responses are a key element of early media too.  It might not hurt
> to
> have a way to indicate why one of these has been generated.

There is nothing special about 183. If you want to say "early media",
then let us just define a 1xx for it, rather than overloading 183. I
really don't get why everyone loves 183. The only provisional resposne I
think Reason makes sense for is 155.

-Jonathan R.

> 
> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> Sent: Wednesday, March 27, 2002 2:56 PM
> To: Sean Olson
> Cc: Gonzalo Camarillo; sip
> Subject: Re: [Sip] Thoughts on the reason header field scope
> 
> Sean Olson wrote:
> 
> >>There
> >>are already elaborate response codes in SIP; why add
> >>one more way of
> >>propagating responses.  155 is a special provisional
> >>response which
> >>elicits a further request (UPDATE), so the Reason
> >>header does add some
> >>value to a generic 155 response.  I think that The
> >>I-D should state that
> >>155 is the only response to which the Reason header
> >>can be applied to.
> >>
> >
> > Would a Reason: header be allowed in a 183?
> 
> I don't speak for the authors of course, but my feeling is that 183 as a
> 
> response code is associated with QoS preconditions.  It has a well
> defined profile, so to speak.  So carrying a Reason phrase in it
> probably does not add too much.
> 
> 155 on the other hand is a response code for solving the early media,
> HERFP, as well as repairable error responses.  So there is some good to
> further qualifying the 155 response for the UAC.
> 
> Regards,
> 
> - vijay
> --
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Internet Software and eServices Group
> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713
> 0184
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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

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From daemon@optimus.ietf.org  Wed Apr  3 02:49:11 2002
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        Sean Olson <seancolson@yahoo.com>, sip <sip@ietf.org>
Subject: Re: [Sip] Thoughts on the reason header field scope
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Hi,

Yes, the only response that can carry Reason will be 155.

About the "early media" response, I do not think it is needed. The SDP
indicates everything needed for the user plane already.

Regards,

Gonzalo

Jonathan Rosenberg wrote:
> 
> Tom-PT Taylor wrote:
> >
> > 183 responses are a key element of early media too.  It might not hurt
> > to
> > have a way to indicate why one of these has been generated.
> 
> There is nothing special about 183. If you want to say "early media",
> then let us just define a 1xx for it, rather than overloading 183. I
> really don't get why everyone loves 183. The only provisional resposne I
> think Reason makes sense for is 155.
> 
> -Jonathan R.
> 
> >
> > -----Original Message-----
> > From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> > Sent: Wednesday, March 27, 2002 2:56 PM
> > To: Sean Olson
> > Cc: Gonzalo Camarillo; sip
> > Subject: Re: [Sip] Thoughts on the reason header field scope
> >
> > Sean Olson wrote:
> >
> > >>There
> > >>are already elaborate response codes in SIP; why add
> > >>one more way of
> > >>propagating responses.  155 is a special provisional
> > >>response which
> > >>elicits a further request (UPDATE), so the Reason
> > >>header does add some
> > >>value to a generic 155 response.  I think that The
> > >>I-D should state that
> > >>155 is the only response to which the Reason header
> > >>can be applied to.
> > >>
> > >
> > > Would a Reason: header be allowed in a 183?
> >
> > I don't speak for the authors of course, but my feeling is that 183 as a
> >
> > response code is associated with QoS preconditions.  It has a well
> > defined profile, so to speak.  So carrying a Reason phrase in it
> > probably does not add too much.
> >
> > 155 on the other hand is a response code for solving the early media,
> > HERFP, as well as repairable error responses.  So there is some good to
> > further qualifying the 155 response for the UAC.
> >
> > Regards,
> >
> > - vijay
> > --
> > Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> > Internet Software and eServices Group
> > Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> > Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713
> > 0184
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com

-- 
Gonzalo Camarillo                    Phone :   +1 212 939 71 71
Columbia University                  Mobile:  +358 40 702 35 35
472 Computer Science Building        Fax   :  +358  9 299 30 52
1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
New York, NY 10027                   
USA                              Gonzalo.Camarillo@ericsson.com

_______________________________________________
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy draft)
Date: Wed, 3 Apr 2002 11:05:23 +0100 
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Rohan,

I'm not sure Proxy-Authorization is appropriate because the information that
we need to supply is nothing to do with Authentication or Authorisation. The
user is already authenticated and the first-hop proxy knows which user the
INVITE is from by virtue of the first-hop integrity mechanism.

However, for a given user, there are multiple identities which that user is
allowed to use, and the user needs to select on a call by call basis which
of those is being used for this call.

If the user does not require any privacy, then this could go in the From
field. But if the user requires privacy, where should it put it ?

...Mark


> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 02 April 2002 18:46
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: 'Dean Willis'; sip@ietf.org
> Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> Privacy draft)
> 
> 
> 
> 
> On Tue, 2 Apr 2002, Mark Watson wrote:
> 
> > Dean wrote:
> >
> > >
> > > "The SIP protocol explicitly sends the From and To fields to the
> > > destination UA. Therefore the sending UA should only insert
> > > information
> > > in the From and To fields that is intended to be delivered to
> > > other UAs
> > > and potentially displayed to the user of the other UA. If 
> the user of
> > > the UA wishes to remain anonymous, they must not insert any
> > > information
> > > into the From or To fields which compromises this 
> anonymity. This same
> > > caveat applies to every other aspect of SIP, including headers and
> > > bodies, except for those elements which are explicitly removed or
> > > encrypted by intermediate elements known to the UA."
> > >
> >
> > OK, so let's work with this and see where we get...
> >
> > 1) The UA may insert information which later becomes 
> private due to the
> > operation of services within the network. The device that 
> detects this must
> > either edit the From/To fields itself, or explicitly route 
> to an Application
> > Server which can do this & anything else needed to provide 
> the necessary
> > privacy.
> >
> > Is everyone happy with this ? Can we descibe it somewhere ?
> >
> > 2) The From/To fields cannot be used for communication from 
> UE to network
> > entities of identity information which the user wishes to 
> be kept private
> > (no way to indicate this)
> >
> > 3GPP has a requirement for information of type (2). They 
> had planned to use
> > RPID, since this was an element 'which is explicitly 
> removed or encrypted by
> > intermediate elements known to the UA'. They cannot use the 
> Authentication
> > mechanism, because in 3GPP this is only done at registration
> 
> It seems that 3GPP could use an unsolicited Proxy-Authorization header
> with the correct public identity identified as the username.
> 
> thanks,
> -rohan
> 
> > (with first hop
> > integrity used for subsequent transactions) and the 
> information to transport
> > can change on a call by call basis.
> >
> > So what should 3GPP use for this information ?
> >
> > Regards,
> >
> > Mark
> >
> 
> 

------_=_NextPart_001_01C1DAF7.11F8F952
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<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP =
Privacy draft)</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Rohan,</FONT>
</P>

<P><FONT SIZE=3D2>I'm not sure Proxy-Authorization is appropriate =
because the information that we need to supply is nothing to do with =
Authentication or Authorisation. The user is already authenticated and =
the first-hop proxy knows which user the INVITE is from by virtue of =
the first-hop integrity mechanism.</FONT></P>

<P><FONT SIZE=3D2>However, for a given user, there are multiple =
identities which that user is allowed to use, and the user needs to =
select on a call by call basis which of those is being used for this =
call.</FONT></P>

<P><FONT SIZE=3D2>If the user does not require any privacy, then this =
could go in the From field. But if the user requires privacy, where =
should it put it ?</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Rohan Mahy [<A =
HREF=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 02 April 2002 18:46</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Dean Willis'; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Private Info in To/From (was RE: =
FW: [Sip] Comment, SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Privacy draft)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; On Tue, 2 Apr 2002, Mark Watson wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Dean wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; &quot;The SIP protocol explicitly =
sends the From and To fields to the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; destination UA. Therefore the sending =
UA should only insert</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; in the From and To fields that is =
intended to be delivered to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; other UAs</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; and potentially displayed to the user =
of the other UA. If </FONT>
<BR><FONT SIZE=3D2>&gt; the user of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the UA wishes to remain anonymous, =
they must not insert any</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; into the From or To fields which =
compromises this </FONT>
<BR><FONT SIZE=3D2>&gt; anonymity. This same</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; caveat applies to every other aspect =
of SIP, including headers and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; bodies, except for those elements =
which are explicitly removed or</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; encrypted by intermediate elements =
known to the UA.&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; OK, so let's work with this and see where =
we get...</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 1) The UA may insert information which =
later becomes </FONT>
<BR><FONT SIZE=3D2>&gt; private due to the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; operation of services within the network. =
The device that </FONT>
<BR><FONT SIZE=3D2>&gt; detects this must</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; either edit the From/To fields itself, or =
explicitly route </FONT>
<BR><FONT SIZE=3D2>&gt; to an Application</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Server which can do this &amp; anything =
else needed to provide </FONT>
<BR><FONT SIZE=3D2>&gt; the necessary</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; privacy.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Is everyone happy with this ? Can we =
descibe it somewhere ?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 2) The From/To fields cannot be used for =
communication from </FONT>
<BR><FONT SIZE=3D2>&gt; UE to network</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; entities of identity information which the =
user wishes to </FONT>
<BR><FONT SIZE=3D2>&gt; be kept private</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (no way to indicate this)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 3GPP has a requirement for information of =
type (2). They </FONT>
<BR><FONT SIZE=3D2>&gt; had planned to use</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; RPID, since this was an element 'which is =
explicitly </FONT>
<BR><FONT SIZE=3D2>&gt; removed or encrypted by</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; intermediate elements known to the UA'. =
They cannot use the </FONT>
<BR><FONT SIZE=3D2>&gt; Authentication</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; mechanism, because in 3GPP this is only =
done at registration</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; It seems that 3GPP could use an unsolicited =
Proxy-Authorization header</FONT>
<BR><FONT SIZE=3D2>&gt; with the correct public identity identified as =
the username.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; -rohan</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (with first hop</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; integrity used for subsequent =
transactions) and the </FONT>
<BR><FONT SIZE=3D2>&gt; information to transport</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; can change on a call by call basis.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; So what should 3GPP use for this =
information ?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Mark</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

</BODY>
</HTML>
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From daemon@ns.ietf.org  Wed Apr  3 05:34:13 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy draft)
Date: Wed, 3 Apr 2002 11:10:05 +0100 
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Dean wrote:

> 1) Absolutely not. The information is what was inserted by the UA. It
> cannot become private "somewhere else in the network" for that dialog.
> The only way to change it is to make a new dialog. Or maybe 
> that's what
> you mean and I'm just being dense.
>

Information inserted by the UA absoutely can become private somewhere else
in the network.

Now if this means that a new dialog is required, then so be it, and then a
genuine full-blown B2BUA is required to mediate between incoming and
outgoing dialog.

Examples: (1) The user inserts their identity, but the subscriber of the
outgoing service being used requires outgoing sessions to be anonymous. Only
the outgoing proxy knows this - well, it's a B2BUA now.

(2) A calls B, who forwards to C. B requires privacy, but unfortunately A
has inserted B's identity in the To field. So the proxy performing
forwarding for B must modify the To field - OK, so it's a B2BUA too.

Anyone offering a public service will need to deal with (1).

Anyone offering a public service who wants to provide call forwarding will
need to deal with (2).

Also, as per my reply to Christian yesterday, it's unclear to me why these
requirements should not also apply to 'private' services in the case that
these are extended outside individual organsiations so as to effectively
replace 'public' services.

If we place no requirements whatsoever on the contents of the To/From
fields, then I guess we could argue that, like Subject: etc., it's not our
lookout what the UA puts in those fields. But we are a very long away from
this - the clear intention of the specification is that the To/From fields
should contain URIs for the called/calling parties resp. A genuine
'anonymous forwarding' service would probably have to remove things like
Subject as well.

It's hard to see how we could argue that service which did not operate as I
described in (1) & (2) had the necessary regulatory privacy capabilities
when we know very well that most UAs will put this information in the
From/To fields.

So, you can't offer a public SIP service without B2BUAs. I'm not casting
judgement on this situation, but I think we should be clear about where we
are. Is everyone happy with this ?




> 2) part 1: Yes! Part 2: 3GPP terminals had better be able to use
> Authentication anytime it's required by a downstream system or they
> aren't conformant to the SIP model as I understand it. Just 
> because they
> think they are only going to authenticate REGISTER doesn't 
> mean they can
> get away with not supporting it in the other usual 
> transactions. So I'd
> suggest that if IETF can come up with something like a third-party
> authentication model that adresses the transitive identity expression
> requirements, that  3GPP could figure out how to use it.

I'm sure that SIP clients on 3GPP mobiles will be able to cope with
Authentication requests from downstream proxies just as well as any other
SIP client. However the requirement in this case is for information to be
passed to the first-hop proxy on all calls.

There is no authentication exchange between UA and first-hop proxy, and
there is no need for an authentication function here because we are using a
first-hop integrity mechanism to link all messages between UA and first hop
proxy back to the authentication performed at registration.

This is not about authentication. It's about the need to pass some
supplementary information to the first-hop proxy which is needed for the
processing of the call, but which must not be shown to the UAS. This
information just so happens to be in the form of an identity.

...Mark




> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 03 April 2002 05:19
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: sip@ietf.org
> Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> Privacy draft)
> 
> 
> Mark wrote:
> 
> ---------------
> OK, so let's work with this and see where we get... 
> 1) The UA may insert information which later becomes private 
> due to the
> operation of services within the network. The device that detects this
> must either edit the From/To fields itself, or explicitly route to an
> Application Server which can do this & anything else needed to provide
> the necessary privacy.
> Is everyone happy with this ? Can we descibe it somewhere ? 
> 2) The From/To fields cannot be used for communication from UE to
> network entities of identity information which the user wishes to be
> kept private (no way to indicate this)
> 3GPP has a requirement for information of type (2). They had 
> planned to
> use RPID, since this was an element 'which is explicitly removed or
> encrypted by intermediate elements known to the UA'. They 
> cannot use the
> Authentication mechanism, because in 3GPP this is only done at
> registration (with first hop integrity used for subsequent 
> transactions)
> and the information to transport can change on a call by call basis.
> So what should 3GPP use for this information ? 
> Regards, 
> Mark 
> -------
> 
> 1) Absolutely not. The information is what was inserted by the UA. It
> cannot become private "somewhere else in the network" for that dialog.
> The only way to change it is to make a new dialog. Or maybe 
> that's what
> you mean and I'm just being dense.
> 
> 2) part 1: Yes! Part 2: 3GPP terminals had better be able to use
> Authentication anytime it's required by a downstream system or they
> aren't conformant to the SIP model as I understand it. Just 
> because they
> think they are only going to authenticate REGISTER doesn't 
> mean they can
> get away with not supporting it in the other usual 
> transactions. So I'd
> suggest that if IETF can come up with something like a third-party
> authentication model that adresses the transitive identity expression
> requirements, that  3GPP could figure out how to use it.
> 
> --
> Dean
> 
> 

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<P><FONT SIZE=3D2>Dean wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; 1) Absolutely not. The information is what was =
inserted by the UA. It</FONT>
<BR><FONT SIZE=3D2>&gt; cannot become private &quot;somewhere else in =
the network&quot; for that dialog.</FONT>
<BR><FONT SIZE=3D2>&gt; The only way to change it is to make a new =
dialog. Or maybe </FONT>
<BR><FONT SIZE=3D2>&gt; that's what</FONT>
<BR><FONT SIZE=3D2>&gt; you mean and I'm just being dense.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>Information inserted by the UA absoutely can become =
private somewhere else in the network.</FONT>
</P>

<P><FONT SIZE=3D2>Now if this means that a new dialog is required, then =
so be it, and then a genuine full-blown B2BUA is required to mediate =
between incoming and outgoing dialog.</FONT></P>

<P><FONT SIZE=3D2>Examples: (1) The user inserts their identity, but =
the subscriber of the outgoing service being used requires outgoing =
sessions to be anonymous. Only the outgoing proxy knows this - well, =
it's a B2BUA now.</FONT></P>

<P><FONT SIZE=3D2>(2) A calls B, who forwards to C. B requires privacy, =
but unfortunately A has inserted B's identity in the To field. So the =
proxy performing forwarding for B must modify the To field - OK, so =
it's a B2BUA too.</FONT></P>

<P><FONT SIZE=3D2>Anyone offering a public service will need to deal =
with (1).</FONT>
</P>

<P><FONT SIZE=3D2>Anyone offering a public service who wants to provide =
call forwarding will need to deal with (2).</FONT>
</P>

<P><FONT SIZE=3D2>Also, as per my reply to Christian yesterday, it's =
unclear to me why these requirements should not also apply to 'private' =
services in the case that these are extended outside individual =
organsiations so as to effectively replace 'public' =
services.</FONT></P>

<P><FONT SIZE=3D2>If we place no requirements whatsoever on the =
contents of the To/From fields, then I guess we could argue that, like =
Subject: etc., it's not our lookout what the UA puts in those fields. =
But we are a very long away from this - the clear intention of the =
specification is that the To/From fields should contain URIs for the =
called/calling parties resp. A genuine 'anonymous forwarding' service =
would probably have to remove things like Subject as well.</FONT></P>

<P><FONT SIZE=3D2>It's hard to see how we could argue that service =
which did not operate as I described in (1) &amp; (2) had the necessary =
regulatory privacy capabilities when we know very well that most UAs =
will put this information in the From/To fields.</FONT></P>

<P><FONT SIZE=3D2>So, you can't offer a public SIP service without =
B2BUAs. I'm not casting judgement on this situation, but I think we =
should be clear about where we are. Is everyone happy with this =
?</FONT></P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; 2) part 1: Yes! Part 2: 3GPP terminals had =
better be able to use</FONT>
<BR><FONT SIZE=3D2>&gt; Authentication anytime it's required by a =
downstream system or they</FONT>
<BR><FONT SIZE=3D2>&gt; aren't conformant to the SIP model as I =
understand it. Just </FONT>
<BR><FONT SIZE=3D2>&gt; because they</FONT>
<BR><FONT SIZE=3D2>&gt; think they are only going to authenticate =
REGISTER doesn't </FONT>
<BR><FONT SIZE=3D2>&gt; mean they can</FONT>
<BR><FONT SIZE=3D2>&gt; get away with not supporting it in the other =
usual </FONT>
<BR><FONT SIZE=3D2>&gt; transactions. So I'd</FONT>
<BR><FONT SIZE=3D2>&gt; suggest that if IETF can come up with something =
like a third-party</FONT>
<BR><FONT SIZE=3D2>&gt; authentication model that adresses the =
transitive identity expression</FONT>
<BR><FONT SIZE=3D2>&gt; requirements, that&nbsp; 3GPP could figure out =
how to use it.</FONT>
</P>

<P><FONT SIZE=3D2>I'm sure that SIP clients on 3GPP mobiles will be =
able to cope with Authentication requests from downstream proxies just =
as well as any other SIP client. However the requirement in this case =
is for information to be passed to the first-hop proxy on all =
calls.</FONT></P>

<P><FONT SIZE=3D2>There is no authentication exchange between UA and =
first-hop proxy, and there is no need for an authentication function =
here because we are using a first-hop integrity mechanism to link all =
messages between UA and first hop proxy back to the authentication =
performed at registration.</FONT></P>

<P><FONT SIZE=3D2>This is not about authentication. It's about the need =
to pass some supplementary information to the first-hop proxy which is =
needed for the processing of the call, but which must not be shown to =
the UAS. This information just so happens to be in the form of an =
identity.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Dean Willis [<A =
HREF=3D"mailto:dean.willis@softarmor.com">mailto:dean.willis@softarmor.c=
om</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 03 April 2002 05:19</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Private Info in To/From (was RE: =
FW: [Sip] Comment, SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Privacy draft)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; ---------------</FONT>
<BR><FONT SIZE=3D2>&gt; OK, so let's work with this and see where we =
get... </FONT>
<BR><FONT SIZE=3D2>&gt; 1) The UA may insert information which later =
becomes private </FONT>
<BR><FONT SIZE=3D2>&gt; due to the</FONT>
<BR><FONT SIZE=3D2>&gt; operation of services within the network. The =
device that detects this</FONT>
<BR><FONT SIZE=3D2>&gt; must either edit the From/To fields itself, or =
explicitly route to an</FONT>
<BR><FONT SIZE=3D2>&gt; Application Server which can do this &amp; =
anything else needed to provide</FONT>
<BR><FONT SIZE=3D2>&gt; the necessary privacy.</FONT>
<BR><FONT SIZE=3D2>&gt; Is everyone happy with this ? Can we descibe it =
somewhere ? </FONT>
<BR><FONT SIZE=3D2>&gt; 2) The From/To fields cannot be used for =
communication from UE to</FONT>
<BR><FONT SIZE=3D2>&gt; network entities of identity information which =
the user wishes to be</FONT>
<BR><FONT SIZE=3D2>&gt; kept private (no way to indicate this)</FONT>
<BR><FONT SIZE=3D2>&gt; 3GPP has a requirement for information of type =
(2). They had </FONT>
<BR><FONT SIZE=3D2>&gt; planned to</FONT>
<BR><FONT SIZE=3D2>&gt; use RPID, since this was an element 'which is =
explicitly removed or</FONT>
<BR><FONT SIZE=3D2>&gt; encrypted by intermediate elements known to the =
UA'. They </FONT>
<BR><FONT SIZE=3D2>&gt; cannot use the</FONT>
<BR><FONT SIZE=3D2>&gt; Authentication mechanism, because in 3GPP this =
is only done at</FONT>
<BR><FONT SIZE=3D2>&gt; registration (with first hop integrity used for =
subsequent </FONT>
<BR><FONT SIZE=3D2>&gt; transactions)</FONT>
<BR><FONT SIZE=3D2>&gt; and the information to transport can change on =
a call by call basis.</FONT>
<BR><FONT SIZE=3D2>&gt; So what should 3GPP use for this information ? =
</FONT>
<BR><FONT SIZE=3D2>&gt; Regards, </FONT>
<BR><FONT SIZE=3D2>&gt; Mark </FONT>
<BR><FONT SIZE=3D2>&gt; -------</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1) Absolutely not. The information is what was =
inserted by the UA. It</FONT>
<BR><FONT SIZE=3D2>&gt; cannot become private &quot;somewhere else in =
the network&quot; for that dialog.</FONT>
<BR><FONT SIZE=3D2>&gt; The only way to change it is to make a new =
dialog. Or maybe </FONT>
<BR><FONT SIZE=3D2>&gt; that's what</FONT>
<BR><FONT SIZE=3D2>&gt; you mean and I'm just being dense.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 2) part 1: Yes! Part 2: 3GPP terminals had =
better be able to use</FONT>
<BR><FONT SIZE=3D2>&gt; Authentication anytime it's required by a =
downstream system or they</FONT>
<BR><FONT SIZE=3D2>&gt; aren't conformant to the SIP model as I =
understand it. Just </FONT>
<BR><FONT SIZE=3D2>&gt; because they</FONT>
<BR><FONT SIZE=3D2>&gt; think they are only going to authenticate =
REGISTER doesn't </FONT>
<BR><FONT SIZE=3D2>&gt; mean they can</FONT>
<BR><FONT SIZE=3D2>&gt; get away with not supporting it in the other =
usual </FONT>
<BR><FONT SIZE=3D2>&gt; transactions. So I'd</FONT>
<BR><FONT SIZE=3D2>&gt; suggest that if IETF can come up with something =
like a third-party</FONT>
<BR><FONT SIZE=3D2>&gt; authentication model that adresses the =
transitive identity expression</FONT>
<BR><FONT SIZE=3D2>&gt; requirements, that&nbsp; 3GPP could figure out =
how to use it.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; Dean</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@ns.ietf.org  Wed Apr  3 05:34:20 2002
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
Date: Wed, 3 Apr 2002 11:17:46 +0100 
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I am less than interested in the screen parameter. It has never, even in the
PSTN, managed to convey information in the absence of the special
arrangement.

However I do want to see RPID (or identity in some form) from untrusted UAs.
Rather than the trusted entity performing verification in this case, I see
the functionality of this as allowing the real trusted entity to select from
a number of valid identities that it might otherwise insert. The fact that
it the identity inserted may appear identical to the received value does not
mean that it has been passed on transparently. All we need to do is clearly
indicate that this guidance was provided by an untrusted entity.

The PSTN case of the special arrangement has to be considered as a trusted
entity sending the information in the first place, and therefore is outside
the scope of this discussion, and therefore we do not need a screen
parameter in the way that this discussion is interpreting it.

Keith 

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com  

-----Original Message-----
From: Flemming Andreasen [mailto:fandreas@cisco.com]
Sent: 01 April 2002 15:47
To: Peterson, Jon
Cc: 'William Marshall'; sip@ietf.org
Subject: Re: [Sip] Comment, SIP Privacy draft




"Peterson, Jon" wrote:

>
> Until sip-privacy removes the 'screen=' parameter, sip-privacy still has
> sunny-day handling for the insertion of RPID headers by untrusted UAs. I'm
> not suggesting that we add text saying that untrusted UAs MUST NOT add the
> RPID. I'm suggesting that the concept of an 'untrusted' RPID be expunged
(by
> removing the 'screen' concept and the associated protocol apparatus), and
> that proxies MUST remove RPIDs they receive from untrusted sources. I
think
> would be much more effective.

You are concerned about somebody doing something the draft does not define,
and
in fact explicitly states it is not intended for. You have not provided any
technical problems with this, but are only concerned with the potential for
misuse. There is no point in either of us continuing to repeat the same
arguments on this point. If somebody else feels strongly about this point
and
has something technical to offer, they should speak up.


> But now I can see how this admits of either reading. However, I'm a little
> confused about how this is then supplying a valuable privacy service. When
I
> request privacy for a call, I don't exactly expect that the person I'm
> calling will be able to call me back.

Bill already explained how malicious call trace works - you will not be able
to
call that party back.

-- Flemming

--
Flemming Andreasen
Cisco Systems




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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Dean Willis'"
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
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A quick couple of notes below.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Wednesday, April 03, 2002 2:10 AM
To: 'Dean Willis'
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
cy draft)


Examples: (1) The user inserts their identity, but the subscriber of the
outgoing service being used requires outgoing sessions to be anonymous. Only
the outgoing proxy knows this - well, it's a B2BUA now.
[Peterson, Jon] One could just as easily argue that this is a requirement
not to force such services into the local outbound proxy. The UA could
always dynamically learn about privacy preferences from the service provider
in some fashion, and the service provider could reject requests (at the
local outbound) that don't conform to the privacy rules. And we should when
possible push SIP features to the endpoints anyway.

(2) A calls B, who forwards to C. B requires privacy, but unfortunately A
has inserted B's identity in the To field. So the proxy performing
forwarding for B must modify the To field - OK, so it's a B2BUA too.
[Peterson, Jon] If B redirects A to C instead, we might get some better
leverage on this problem - that could be one way that B could 'require
privacy'. Today the SIP spec RECOMMENDs that the To field in the new request
following the redirection be the same as the To field in the original
request (although it says UAs MAY update the To and whatnot if they are so
inclined) - however, the spec also allows you to insert replacement headers
into the URL, like:

Contact: sip:C@cleveland.com?To=anonymous@invalid.net
<mailto:sip:C@cleveland.com?To=anonymous@invalid.net> 

I bring these points up not because I disagree that intermediaries providing
privacy and identity will often need to be More Than Just A  Proxy - but I
think there are reasonable attacks on the examples you happen to raise here
that don't require resorting to a B2BUA. Let's not be too quick to close
doors.


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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=640444710-03042002>A 
quick couple of notes below.</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=640444710-03042002>Jon 
Peterson</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=640444710-03042002>NeuStar, Inc.</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Wednesday, April 03, 2002 
  2:10 AM<BR><B>To:</B> 'Dean Willis'<BR><B>Cc:</B> 
  sip@ietf.org<BR><B>Subject:</B> RE: Private Info in To/From (was RE: FW: [Sip] 
  Comment, SIP Priva cy draft)<BR></FONT></DIV>
  <P><FONT size=2>Examples: (1) The user inserts their identity, but the 
  subscriber of the outgoing service being used requires outgoing sessions to be 
  anonymous. Only the outgoing proxy knows this - well, it's a B2BUA 
  now.<BR><FONT color=#0000ff face=Arial><SPAN 
  class=640444710-03042002>[Peterson, Jon]&nbsp;One could just as easily argue 
  that this is a requirement not to force such services into the&nbsp;local 
  outbound proxy.&nbsp;The UA could always dynamically learn about privacy 
  preferences from the service provider in some fashion, and the service 
  provider could reject requests (at the local outbound) that don't conform to 
  the privacy rules. And we should when possible push SIP features to the 
  endpoints anyway.</SPAN></FONT></FONT></P>
  <P><FONT size=2>(2) A calls B, who forwards to C. B requires privacy, but 
  unfortunately A has inserted B's identity in the To field. So the proxy 
  performing forwarding for B must modify the To field - OK, so it's a B2BUA 
  too.<BR><FONT color=#0000ff face=Arial><SPAN 
  class=640444710-03042002>[Peterson, Jon]&nbsp;If B redirects A to C instead, 
  we might get some better leverage on this problem - that could be one way that 
  B could 'require privacy'. Today the SIP spec RECOMMENDs that the To field in 
  the new request following the redirection&nbsp;be the same as the To field in 
  the original request (although it says UAs MAY update the To and whatnot if 
  they are so inclined) - however, the spec also allows you to insert 
  replacement headers into the URL, like:</SPAN></FONT></FONT></P>
  <P><FONT size=2><FONT color=#0000ff face=Arial><SPAN 
  class=640444710-03042002>Contact: <A 
  href="mailto:sip:C@cleveland.com?To=anonymous@invalid.net">sip:C@cleveland.com?To=anonymous@invalid.net</A></SPAN></FONT></FONT></P>
  <P><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
  class=640444710-03042002>I bring these points up not because I disagree that 
  intermediaries providing privacy and identity will often need to be More Than 
  Just A&nbsp; Proxy&nbsp;- but I think there are reasonable attacks on the 
  examples you happen to raise here that don't require resorting to a B2BUA. 
  Let's not be too quick to close 
doors.</SPAN></FONT></FONT></FONT></P></BLOCKQUOTE></BODY></HTML>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Dean Willis'"
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	 cy draft)
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: (1) The user inserts their identity, but the subscriber of the outgoing
service being used requires outgoing sessions to be anonymous. Only the
outgoing proxy knows this - well, it's a B2BUA now.
[Peterson, Jon] One could just as easily argue that this is a requirement
not to force such services into the local outbound proxy. The UA could
always dynamically learn about privacy preferences from the service provider
in some fashion, and the service provider could reject requests (at the
local outbound) that don't conform to the privacy rules. And we should when
possible push SIP features to the endpoints anyway. 

MW> I'm happy with any solution which works & the above sounds promising.
How would we get the privacy preferences to the UAC & is it backwards
compatible ? 

(2) A calls B, who forwards to C. B requires privacy, but unfortunately A
has inserted B's identity in the To field. So the proxy performing
forwarding for B must modify the To field - OK, so it's a B2BUA too.
[Peterson, Jon] If B redirects A to C instead, we might get some better
leverage on this problem - that could be one way that B could 'require
privacy'. Today the SIP spec RECOMMENDs that the To field in the new request
following the redirection be the same as the To field in the original
request (although it says UAs MAY update the To and whatnot if they are so
inclined) - however, the spec also allows you to insert replacement headers
into the URL, like:

Contact: sip:C@cleveland.com?To=anonymous@invalid.net
<mailto:sip:C@cleveland.com?To=anonymous@invalid.net> 

I bring these points up not because I disagree that intermediaries providing
privacy and identity will often need to be More Than Just A  Proxy - but I
think there are reasonable attacks on the examples you happen to raise here
that don't require resorting to a B2BUA. Let's not be too quick to close
doors. 

MW> The point I'm trying to make is exactly that intermediaries providing
privacy *will* often need to be More Than Just A Proxy, and I'm trying to
see if others on the list agree with this. Some of the 'Bad Things' that
people are concerned about with respect to From/To are happening because
people are trying to solve these privacy problems *without* realising that
they need More Than Just A Proxy.

With respect to your example, I can't see that you could trust A's UA to do
this correctly, and there is no device that the new A->C call will pass
through which could reject it if A does not follow the rules in the new URI.
Passing the privacy requirements back to the UA is a good idea, but only
works if there are proxies in a position to reject requests where the UA has
not obeyed the privacy requirements.

...Mark


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<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy draft)</TITLE>

<META content="MSHTML 5.00.3315.2870" name=GENERATOR></HEAD>
<BODY>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <P><FONT size=2>: (1) The user inserts their identity, but the subscriber of 
    the outgoing service being used requires outgoing sessions to be anonymous. 
    Only the outgoing proxy knows this - well, it's a B2BUA now.<BR><FONT 
    color=#0000ff><FONT face=Arial><SPAN class=640444710-03042002>[Peterson, 
    Jon]&nbsp;One could just as easily argue that this is a requirement not to 
    force such services into the&nbsp;local outbound proxy.&nbsp;The UA could 
    always dynamically learn about privacy preferences from the service provider 
    in some fashion, and the service provider could reject requests (at the 
    local outbound) that don't conform to the privacy rules. And we should when 
    possible push SIP features to the endpoints anyway.<FONT face=Verdana 
    size=1><SPAN 
    class=147255411-03042002>&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></P></BLOCKQUOTE></BLOCKQUOTE>
<P><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=640444710-03042002><FONT face=Verdana size=1><SPAN 
class=147255411-03042002>MW&gt; I'm happy with any solution which works &amp; 
the above sounds promising. How would we get the privacy preferences to the UAC 
&amp; is it backwards compatible 
?&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></P>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <P><FONT size=2>(2) A calls B, who forwards to C. B requires privacy, but 
    unfortunately A has inserted B's identity in the To field. So the proxy 
    performing forwarding for B must modify the To field - OK, so it's a B2BUA 
    too.<BR><FONT color=#0000ff face=Arial><SPAN 
    class=640444710-03042002>[Peterson, Jon]&nbsp;If B redirects A to C instead, 
    we might get some better leverage on this problem - that could be one way 
    that B could 'require privacy'. Today the SIP spec RECOMMENDs that the To 
    field in the new request following the redirection&nbsp;be the same as the 
    To field in the original request (although it says UAs MAY update the To and 
    whatnot if they are so inclined) - however, the spec also allows you to 
    insert replacement headers into the URL, like:</SPAN></FONT></FONT></P>
    <P><FONT size=2><FONT color=#0000ff face=Arial><SPAN 
    class=640444710-03042002>Contact: <A 
    href="mailto:sip:C@cleveland.com?To=anonymous@invalid.net">sip:C@cleveland.com?To=anonymous@invalid.net</A></SPAN></FONT></FONT></P>
    <P><FONT color=#0000ff><FONT size=2><FONT face=Arial><SPAN 
    class=640444710-03042002>I bring these points up not because I disagree that 
    intermediaries providing privacy and identity will often need to be More 
    Than Just A&nbsp; Proxy&nbsp;- but I think there are reasonable attacks on 
    the examples you happen to raise here that don't require resorting to a 
    B2BUA. Let's not be too quick to close doors.<FONT face=Verdana size=1><SPAN 
    class=147255411-03042002>&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></P></BLOCKQUOTE></BLOCKQUOTE>
<P><FONT color=#0000ff><FONT size=2><FONT face=Arial><SPAN 
class=640444710-03042002><FONT face=Verdana size=1><SPAN 
class=147255411-03042002>MW&gt; The point I'm trying to make is exactly that 
intermediaries providing privacy *will* often need to be More Than Just A Proxy, 
and I'm trying to see if others on the list agree with this. Some of the 'Bad 
Things' that people are concerned about with respect to From/To are happening 
because people are trying to solve these privacy problems *without* realising 
that they need More Than Just A 
Proxy.</SPAN></FONT></SPAN></FONT></FONT></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=147255411-03042002>With 
respect to your example, I can't see that you could trust A's UA to do this 
correctly, and there is no device that the new A-&gt;C call will pass through 
which could reject it if A does not follow the rules in the new URI. Passing the 
privacy requirements back to the UA is a good idea, but only works if there are 
proxies in a position to reject requests where the UA has not obeyed the privacy 
requirements.</SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN 
class=147255411-03042002>...Mark</SPAN></FONT></P></BODY></HTML>

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Subject: RE: [Sip] Poll: interest in Reason header work
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Works for me (of course I'm not exactly un-involved in this little
effort :-) )

Dave.

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Tuesday, April 02, 2002 11:39 PM
> To: sip@ietf.org
> Cc: Rohan@cicso.com; jo@ipdialog.com; brian.rosen@marconi.com
> Subject: [Sip] Poll: interest in Reason header work
> 
> 
> It looks like UPDATE and Manyfolks are likely to be dependent 
> on the Reason header mechanism (imho). Given this (and the 
> other requirements discussions that we've heard), how do you 
> people feel about proposing this as a SIP WG effort? And yes, 
> the SIP WG can identify work that needs to be done to support 
> other SIP WG efforts without having a third-party generate a 
> requirements RFC.
> 
> --
> Dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


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Subject: RE: [Sip] Poll: interest in Reason header work
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I support bringing this into SIP

--Andy

> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Tuesday, April 02, 2002 9:39 PM
> To: sip@ietf.org
> Cc: Rohan@cicso.com; jo@ipdialog.com; brian.rosen@marconi.com
> Subject: [Sip] Poll: interest in Reason header work
> 
> 
> It looks like UPDATE and Manyfolks are likely to be dependent on the
> Reason header mechanism (imho). Given this (and the other requirements
> discussions that we've heard), how do you people feel about proposing
> this as a SIP WG effort? And yes, the SIP WG can identify work that
> needs to be done to support other SIP WG efforts without having a
> third-party generate a requirements RFC.
> 
> --
> Dean

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From daemon@ns.ietf.org  Wed Apr  3 08:46:19 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: sip <sip@ietf.org>
Subject: RE: [Sip] Thoughts on the reason header field scope
Date: Wed, 3 Apr 2002 08:17:21 -0500 
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You can get early media as a normal part of call setup (e.g. ringing
provided at the terminating end), in preparation for moving you to a new
destination ("Your call is being forwarded to ..."), or as part of a call
failure sequence (e.g. "That number has changed.  Youc can now reach your
party at ...").  It might be useful for the caller's application to be able
to display the difference.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, April 03, 2002 2:27 AM
To: Taylor, Tom-PT [CAR:B800:EXCH]
Cc: 'Vijay K. Gurbani'; Sean Olson; Gonzalo Camarillo; sip
Subject: Re: [Sip] Thoughts on the reason header field scope




Tom-PT Taylor wrote:
> 
> 183 responses are a key element of early media too.  It might not hurt
> to
> have a way to indicate why one of these has been generated.

There is nothing special about 183. If you want to say "early media",
then let us just define a 1xx for it, rather than overloading 183. I
really don't get why everyone loves 183. The only provisional resposne I
think Reason makes sense for is 155.

-Jonathan R.

> 
> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> Sent: Wednesday, March 27, 2002 2:56 PM
> To: Sean Olson
> Cc: Gonzalo Camarillo; sip
> Subject: Re: [Sip] Thoughts on the reason header field scope
> 
> Sean Olson wrote:
> 
> >>There
> >>are already elaborate response codes in SIP; why add
> >>one more way of
> >>propagating responses.  155 is a special provisional
> >>response which
> >>elicits a further request (UPDATE), so the Reason
> >>header does add some
> >>value to a generic 155 response.  I think that The
> >>I-D should state that
> >>155 is the only response to which the Reason header
> >>can be applied to.
> >>
> >
> > Would a Reason: header be allowed in a 183?
> 
> I don't speak for the authors of course, but my feeling is that 183 as a
> 
> response code is associated with QoS preconditions.  It has a well
> defined profile, so to speak.  So carrying a Reason phrase in it
> probably does not add too much.
> 
> 155 on the other hand is a response code for solving the early media,
> HERFP, as well as repairable error responses.  So there is some good to
> further qualifying the 155 response for the UAC.
> 
> Regards,
> 
> - vijay
> --
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Internet Software and eServices Group
> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713
> 0184
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From: "David R. Oran" <oran@cisco.com>
To: "'Tom-PT Taylor'" <taylor@nortelnetworks.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'sip'" <sip@ietf.org>
Subject: RE: [Sip] Thoughts on the reason header field scope
Date: Wed, 3 Apr 2002 08:17:18 -0500
Organization: Cisco Systems
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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Tom-PT Taylor
> Sent: Wednesday, April 03, 2002 8:17 AM
> To: 'Jonathan Rosenberg'
> Cc: sip
> Subject: RE: [Sip] Thoughts on the reason header field scope
> 
> 
> You can get early media as a normal part of call setup (e.g. 
> ringing provided at the terminating end), in preparation for 
> moving you to a new destination ("Your call is being 
> forwarded to ..."), or as part of a call failure sequence 
> (e.g. "That number has changed.  Youc can now reach your 
> party at ...").  It might be useful for the caller's 
> application to be able to display the difference.
>
Uh, if the failure/redirection is known in the signaling, then you
should be sending back a 3xx or whatever. The *only* purpose I know of
for early media is to deal with cases where you are talking to something
that CAN'T tell you via the signaling and instead sends you media.

Color me super confused.

Dave.



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva	 cy draft)
To: "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: sip@ietf.org
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Police departments or emergency shelters may be interested in a forwarding
service that requires anonymity.





"Peterson, Jon" <jon.peterson@neustar.biz>@ietf.org on 04/03/2002 09:32:27
AM

Sent by:  sip-admin@ietf.org


To:   "'Mark Watson'" <mwatson@nortelnetworks.com>, "'Dean Willis'"
      <dean.willis@softarmor.com>
cc:   sip@ietf.org

Subject:  RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
       cy draft)



A few  more notes below.

Jon  Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson  [mailto:mwatson@nortelnetworks.com]
Sent: Wednesday, April 03, 2002  3:59 AM
To: 'Peterson, Jon'; 'Dean Willis'
Cc:  sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip]  Comment, SIP Priva
cy draft)



MW> I'm happy with any solution which works &  the above sounds promising.
How would we get the privacy preferences to the  UAC & is it backwards
compatible ?
[Peterson, Jon]  It sounds to me like an  instance of the general SIP UA
configuration problem - we already do have an  action item to tackle that.
Sure, there's work to be done, but the same is  true of the B2BUA approach.
Not sure what you're concerned about this being  backwards compatible with,
exactly.

MW> The point I'm trying to make is exactly that intermediaries  providing
privacy *will* often need to be More Than Just A Proxy, and I'm  trying to
see if others on the list agree with this. Some of the 'Bad Things'  that
people are concerned about with respect to From/To are happening because
people are trying to solve these privacy problems *without* realising that
they need More Than Just A Proxy.
[Peterson, Jon] Agreed (but we knew  that).

With respect to your example, I can't see that you could trust  A's UA to
do this correctly, and there is no device that the new A->C call  will pass
through which could reject it if A does not follow the rules in the  new
URI. Passing the privacy requirements back to the UA is a good idea, but
only works if there are proxies in a position to reject requests where the
UA  has not obeyed the privacy requirements.
[Peterson,  Jon] I can't imagine that, in any network likely to interest
you, A  wouldn't have a hardwired pre-loaded Route to its local outbound
that would  ensure the A->C path has some adult supervision, and that the
local  outbound would be in the redirection response path anyway (from the
Vias on  the original A->B request). If it is so exceptionally paranoid
about  misbehaving UAs, it could enact any number of policies to make sure
that  nothing goes awry.


But I wonder, do the regulatory  constraints that make this a requirement
also levy a hefty fine on A if he  reveals to C in the ensuing voice
conversation that he had originally dialed B? When the  redirection with
the Contact address I described in my last mail arrives at A, A would  have
to intentionally countermand it in order to reveal B's identityin the
signaling. But A is of course empowered to reveal  B's identity through
other means anyway. I have a hard time seeing what the  big deal is. B only
remains anonymous at  A's sufferance  anyway.

...Mark





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From daemon@ns.ietf.org  Wed Apr  3 09:26:43 2002
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To: "Dean Willis" <dwillis@dynamicsoft.com>, <sip@ietf.org>
Cc: <Rohan@cicso.com>, <jo@ipdialog.com>, <brian.rosen@marconi.com>
References: <005e01c1dac9$71f57d40$73630a40@TXDWILLIS2>
Subject: Re: [Sip] Poll: interest in Reason header work
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----- Original Message ----- 
From: "Dean Willis" <dwillis@dynamicsoft.com>


> It looks like UPDATE and Manyfolks are likely to be dependent on the
> Reason header mechanism (imho). Given this (and the other requirements
> discussions that we've heard), how do you people feel about proposing
> this as a SIP WG effort? And yes, the SIP WG can identify work that
> needs to be done to support other SIP WG efforts without having a
> third-party generate a requirements RFC.
>
I think the SIP WG should do the Reason header.

Are you suggesting that it would be added it to Bundle 2?

cheers,
(-:bob


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From daemon@ns.ietf.org  Wed Apr  3 09:27:51 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'David R. Oran'" <oran@cisco.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
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Subject: RE: [Sip] Thoughts on the reason header field scope
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OK, I'll 'fess up.  I'm thinking of PSTN interworking situations, where, for
example, the forwarding is happening in the PSTN.

-----Original Message-----
From: David R. Oran [mailto:oran@cisco.com]
Sent: Wednesday, April 03, 2002 8:17 AM
To: Taylor, Tom-PT [CAR:B800:EXCH]; 'Jonathan Rosenberg'
Cc: 'sip'
Subject: RE: [Sip] Thoughts on the reason header field scope


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Tom-PT Taylor
> Sent: Wednesday, April 03, 2002 8:17 AM
> To: 'Jonathan Rosenberg'
> Cc: sip
> Subject: RE: [Sip] Thoughts on the reason header field scope
> 
> 
> You can get early media as a normal part of call setup (e.g. 
> ringing provided at the terminating end), in preparation for 
> moving you to a new destination ("Your call is being 
> forwarded to ..."), or as part of a call failure sequence 
> (e.g. "That number has changed.  Youc can now reach your 
> party at ...").  It might be useful for the caller's 
> application to be able to display the difference.
>
Uh, if the failure/redirection is known in the signaling, then you
should be sending back a 3xx or whatever. The *only* purpose I know of
for early media is to deal with cases where you are talking to something
that CAN'T tell you via the signaling and instead sends you media.

Color me super confused.

Dave.



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: "Bob Penfield" <bpenfield@acmepacket.com>
To: "Rohan Mahy" <rohan@cisco.com>, <sip@ietf.org>
References: <Pine.WNT.4.44.0203301007540.-451835@chorizo.rapidconvergence.com>
Subject: Re: [Sip] Summarization of Reason header discussion
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see below
----- Original Message -----
From: "Rohan Mahy" <rohan@cisco.com>
>
> I haven't heard anything more about the Reason header discussion, so I'm
> going to summarize what I think we have consensus on.
>
> 1) Gonzalo sent a bunch of requirements.  Nobody fundamentally challenges
> that the requirements are wrong or unimportant.  We've also seen a few
> proposed additions.  Lots of interesting stuff could reference this work.
>
> 2) Dave Oran proposed that we keep a split between the reason a call
> arrived, and the request history (which contains names).  The later has
> lots of privacy concerns, while the privacy concerns for the former are
> minimal.  I didn't see much in the way of objections here.
>
> 3) From a syntax perspective, I think we got agreement that the Reason
> should always begin with a SIP response code and a textual phrase, but may
> also include other responses/code families as extra parameters (for
> example, CPIM error messages, Q.850 error codes, etc.).  The motivation is
> that a SIP device only needs to know about one set of response codes (SIP)
> to do something appropriate with the Reason, but additional information is
> available if you care.

I think all we need is 3 elements: a type (SIP, CPIM, Q.850, etc), a status
code, and a reason phrase. If an element does not know the type, it just
ignores it. It seems wasteful to have to make up and include a SIP status
code when all I want to do is pass the ISUP cause code. I also think we
should allow multiple Reason headers in case we have multiple types of
status codes.

cheers,
(-:bob


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Dean Willis'"
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	 cy draft)
Date: Wed, 3 Apr 2002 07:32:27 -0600 
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A few more notes below.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Wednesday, April 03, 2002 3:59 AM
To: 'Peterson, Jon'; 'Dean Willis'
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
cy draft)



MW> I'm happy with any solution which works & the above sounds promising.
How would we get the privacy preferences to the UAC & is it backwards
compatible ? 
[Peterson, Jon]  It sounds to me like an instance of the general SIP UA
configuration problem - we already do have an action item to tackle that.
Sure, there's work to be done, but the same is true of the B2BUA approach.
Not sure what you're concerned about this being backwards compatible with,
exactly.

MW> The point I'm trying to make is exactly that intermediaries providing
privacy *will* often need to be More Than Just A Proxy, and I'm trying to
see if others on the list agree with this. Some of the 'Bad Things' that
people are concerned about with respect to From/To are happening because
people are trying to solve these privacy problems *without* realising that
they need More Than Just A Proxy.
[Peterson, Jon] Agreed (but we knew that). 

With respect to your example, I can't see that you could trust A's UA to do
this correctly, and there is no device that the new A->C call will pass
through which could reject it if A does not follow the rules in the new URI.
Passing the privacy requirements back to the UA is a good idea, but only
works if there are proxies in a position to reject requests where the UA has
not obeyed the privacy requirements.
[Peterson, Jon] I can't imagine that, in any network likely to interest you,
A wouldn't have a hardwired pre-loaded Route to its local outbound that
would ensure the A->C path has some adult supervision, and that the local
outbound would be in the redirection response path anyway (from the Vias on
the original A->B request). If it is so exceptionally paranoid about
misbehaving UAs, it could enact any number of policies to make sure that
nothing goes awry.

But I wonder, do the regulatory constraints that make this a requirement
also levy a hefty fine on A if he reveals to C in the ensuing voice
conversation that he had originally dialed B? When the redirection with the
Contact address I described in my last mail arrives at A, A would have to
intentionally countermand it in order to reveal B's identity in the
signaling. But A is of course empowered to reveal B's identity through other
means anyway. I have a hard time seeing what the big deal is. B only remains
anonymous at A's sufferance anyway.

...Mark


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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=309002312-03042002>A few 
more notes below.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=309002312-03042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=309002312-03042002>Jon 
Peterson</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=309002312-03042002>NeuStar, Inc.</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Wednesday, April 03, 2002 
  3:59 AM<BR><B>To:</B> 'Peterson, Jon'; 'Dean Willis'<BR><B>Cc:</B> 
  sip@ietf.org<BR><B>Subject:</B> RE: Private Info in To/From (was RE: FW: [Sip] 
  Comment, SIP Priva cy draft)<BR><BR></FONT></DIV>
  <P><FONT color=#0000ff><FONT size=2><FONT face=Arial><SPAN 
  class=640444710-03042002><FONT face=Verdana size=1><SPAN 
  class=147255411-03042002>MW&gt; I'm happy with any solution which works &amp; 
  the above sounds promising. How would we get the privacy preferences to the 
  UAC &amp; is it backwards compatible ?&nbsp;<BR><FONT face=Arial size=2><SPAN 
  class=309002312-03042002>[Peterson, Jon]&nbsp;&nbsp;It sounds to me like an 
  instance of the general SIP UA configuration problem - we already do have an 
  action item to tackle that. Sure, there's work to be done, but the same is 
  true of the B2BUA approach. Not sure what you're concerned about this being 
  backwards compatible with, 
  exactly.</SPAN></FONT></SPAN></FONT></SPAN></FONT></FONT></FONT></P>
  <P><FONT color=#0000ff><FONT face=Arial><FONT size=2><SPAN 
  class=640444710-03042002><SPAN class=147255411-03042002><FONT face=Verdana 
  size=1>MW&gt; The point I'm trying to make is exactly that intermediaries 
  providing privacy *will* often need to be More Than Just A Proxy, and I'm 
  trying to see if others on the list agree with this. Some of the 'Bad Things' 
  that people are concerned about with respect to From/To are happening because 
  people are trying to solve these privacy problems *without* realising that 
  they need More Than Just A Proxy.</FONT><BR><SPAN 
  class=309002312-03042002>[Peterson, Jon]&nbsp;</SPAN></SPAN></SPAN><SPAN 
  class=309002312-03042002>Agreed (but we knew 
  that).&nbsp;</SPAN></FONT></FONT></FONT></P>
  <P><FONT face=Verdana size=1><SPAN class=147255411-03042002><FONT 
  color=#0000ff>With respect to your example, I can't see that you could trust 
  A's UA to do this correctly, and there is no device that the new A-&gt;C call 
  will pass through which could reject it if A does not follow the rules in the 
  new URI. Passing the privacy requirements back to the UA is a good idea, but 
  only works if there are proxies in a position to reject requests where the UA 
  has not obeyed the privacy requirements.<BR></FONT><SPAN 
  class=309002312-03042002><FONT color=#0000ff face=Arial size=2>[Peterson, 
  Jon]&nbsp;I can't imagine that, in any network likely to interest you, A 
  wouldn't have a hardwired pre-loaded Route to its local outbound that would 
  ensure the A-&gt;C path has some adult supervision, and that the local 
  outbound would be in the redirection response path anyway (from the Vias on 
  the original A-&gt;B request). If it is so exceptionally paranoid about 
  misbehaving UAs, it could enact any number of policies to make sure that 
  nothing goes awry.</FONT></P><FONT size=2><FONT color=#0000ff><FONT 
face=Arial>
  <P><SPAN class=464170412-03042002><SPAN 
  class=309002312-03042002>But&nbsp;</SPAN>I wonder, do the regulatory 
  constraints that make this a requirement also levy a hefty fine on A if he 
  reveals to C in <SPAN class=309002312-03042002>the&nbsp;</SPAN>ensuing <SPAN 
  class=309002312-03042002>voice </SPAN>conversation that he <SPAN 
  class=309002312-03042002>had originally dialed&nbsp;</SPAN>B? When the 
  redirection with the Contact address I described <SPAN 
  class=309002312-03042002>in my last mail&nbsp;</SPAN>arrives at A, A would 
  have to intentionally countermand it in order to reveal B's identity<SPAN 
  class=309002312-03042002> in the signaling</SPAN>.&nbsp;<SPAN 
  class=309002312-03042002>But&nbsp;</SPAN>A is of course empowered to reveal 
  B's identity through other means anyway. I have a hard time seeing what the 
  big deal is.<SPAN class=309002312-03042002>&nbsp;B only remains anonymous at 
  A's sufferance 
  anyway.</SPAN></SPAN></SPAN></FONT></FONT></FONT></SPAN></FONT></P>
  <P><FONT color=#0000ff face=Verdana size=1><SPAN 
  class=147255411-03042002>...Mark</SPAN></FONT></P></BLOCKQUOTE></BODY></HTML>

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From: "Mahey, Sonit" <SMahey@unispherenetworks.com>
To: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>
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	 <fluffy@cisco.com>,
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Subject:  Re: [Sip] Residential UA security
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Hi Jonathan,

I did not follow the last para in your response. Could you please explain
further the "anonymous" authentication mechanism that you are referring to
and how it can be effective in DDOS and your thoughts on direct DOS attack
mitigation, if "anonymous" method is ineffective.

Thanks,

- sonit



Date: Tue, 02 Apr 2002 14:58:11 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
To: Cullen Jennings <fluffy@cisco.com>
CC: gvelo@astartec.com, sip@ietf.org


Cullen Jennings wrote:
> 
> It is an interesting question - it may not be solvable on the other hand
> it
> may not need to solved as directly as you think.
> 
> With a modem set up as an auto dialer, I could make your current PSTN
> phone
> unusable. I could flood your email box so you were likely over quota. I
> could probably use up on the bandwidth on your WAN link. None of these
> DOS
> issues seem to be a huge problem today. One reason is that there is some
> hope you could trace back and figure out who was doing this to you then
> perhaps filter or take legal action.

For the PSTN, this kind of attack is not common because it costs money
to make a call. It doesn't cost money to send email, and thats why there
is spam.

That said, the anonymous authentication mechanism can prevent DDOS
attacks on my phone, but not a direct targeted attack from a single
source. 

-Jonathan R.



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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Dean Willis'"
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
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MW> I'm happy with any solution which works & the above sounds promising.
How would we get the privacy preferences to the UAC & is it backwards
compatible ? 
[Peterson, Jon]  It sounds to me like an instance of the general SIP UA
configuration problem - we already do have an action item to tackle that.
Sure, there's work to be done, but the same is true of the B2BUA approach.
Not sure what you're concerned about this being backwards compatible with,
exactly. 

MW2> RFC2543 clients - how can I ensure the subscribers privacy requirements
are met if the user has an RFC2543 client ? If I'm not worried about this
case, then my proxy can easily just modify the From field, since dialog
matching is based only on the tags. 

MW> The point I'm trying to make is exactly that intermediaries providing
privacy *will* often need to be More Than Just A Proxy, and I'm trying to
see if others on the list agree with this. Some of the 'Bad Things' that
people are concerned about with respect to From/To are happening because
people are trying to solve these privacy problems *without* realising that
they need More Than Just A Proxy.
[Peterson, Jon] Agreed (but we knew that). 

With respect to your example, I can't see that you could trust A's UA to do
this correctly, and there is no device that the new A->C call will pass
through which could reject it if A does not follow the rules in the new URI.
Passing the privacy requirements back to the UA is a good idea, but only
works if there are proxies in a position to reject requests where the UA has
not obeyed the privacy requirements.
[Peterson, Jon] I can't imagine that, in any network likely to interest you,
A wouldn't have a hardwired pre-loaded Route to its local outbound that
would ensure the A->C path has some adult supervision, and that the local
outbound would be in the redirection response path anyway (from the Vias on
the original A->B request). If it is so exceptionally paranoid about
misbehaving UAs, it could enact any number of policies to make sure that
nothing goes awry. 

MW2> So the outbound proxy keeps a note of the Redirection redirection
request going back to A and then policies new INVITES coming in to spot the
redirected call and enforce B's privacy request ? So I need to keep this
state information about outstanding redirect requests that have been passed
to A. I guess it could be done, but it does not sound too great. 

 But I wonder, do the regulatory constraints that make this a requirement
also levy a hefty fine on A if he reveals to C in the ensuing voice
conversation that he had originally dialed B? 

MW2> No, the constraint in this case is really on B's Service Provider. If
this Service Provider offers a forwarding service, then they also ought to
offer an 'Anonymous Forwarding Service' where B's identity is not revealed
to C *by the network*. There is nothing to stop A revealing B's identity to
C.

The problem with just using SIP redirection, is that B's Service Provider
would then have chosen an implementation of the Call Forwarding service
which intrinsically causes B's identity to be revealed to C. Again, I think
the B's Service Provider would be on dodgy ground if they tried to represent
this as an 'Anonymous Forwarding Service' on the basis that 'it wasn't me
wot revealed B's identity, it was that villian A'.

I'll admit it is an open question as to how the identities in SIP will be
interpreted by regulators. The regulatory constraints apply to third parties
(e.g. Service Providers) releasing identity information about the end
users/subscribers. Is the information in the From/To fields being supplied
by the Service Provider, or the end user themselves ?? From a technical
perspective, obviously, it's being supplied by the end user, but it's not
the technical perspective which is used to evaluate the effect of these
regulations, it's the customer's perspective, which is not related to the
technology.

If I as a Service Provider *require* that the end user insert certain
information in these fields, then I am clearly responsible for the insertion
of the information, so I think I inherit some responsibility for preventing
privacy breachs as a result of this.

If I require that end users use SIP-compliant clients, then we inherit all
the requirements of the SIP spec. As I argued before, there is certainly an
implicit 'soft' requirement that the To/From fields should contain the
called/calling party address.

So, I would find it hard as a Service Provider to credibly argue that I have
no responsibility whatsoever for the information placed in these fields by
my users' equipment, when I have required them to be SIP compliant and when
the SIP spec implies so strongly that this information will be there. In the
case of 3GPP it is clearer, because the 3GPP spec will explicitly state what
will or won't be in the To/From fields. That is why it currently explicitly
states that there should be garbage in there, so I can be sure I'm not
responsible for it!

 When the redirection with the Contact address I described in my last mail
arrives at A, A would have to intentionally countermand it in order to
reveal B's identity in the signaling. But A is of course empowered to reveal
B's identity through other means anyway. I have a hard time seeing what the
big deal is. B only remains anonymous at A's sufferance anyway. 

MW2> Again, it's the distinction between the devices supporting the service
(UAs, Proxies etc.) releasing the information because this is intrinsic in
the Service I am offering, and information being released entirely due to
the whim of the human users themselves. The former is regulated, the latter
is freedom of speech.

Anyway, after all that, I was only arguing (a) that in a Service Provider
model, protection of the subscribers privacy would require More Than Just a
Proxy and (b) that Anonymous Call Forwarding would require 'More Than Just a
Proxy'. (assuming RFC2543 compatibility in both cases).

In both cases, you have provided alternative implementations. For (a) I
think your proposal is not compatible with 2543 clients, and for (b) it
requires MTJAP anyway. I actually think that if you drop the 2543
requirement, then you arrive at better solutions for all this. The
overloading of To/From with two completely distinct functions (identity and
dialog matching) in 2543 is a real pain.

...Mark


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<BODY>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
    size=2><BR></FONT></DIV>
    <P><FONT color=#0000ff><FONT size=2><FONT face=Arial><SPAN 
    class=640444710-03042002><FONT face=Verdana size=1><SPAN 
    class=147255411-03042002>MW&gt; I'm happy with any solution which works 
    &amp; the above sounds promising. How would we get the privacy preferences 
    to the UAC &amp; is it backwards compatible ?&nbsp;<BR><FONT face=Arial 
    size=2><SPAN class=309002312-03042002>[Peterson, Jon]&nbsp;&nbsp;It sounds 
    to me like an instance of the general SIP UA configuration problem - we 
    already do have an action item to tackle that. Sure, there's work to be 
    done, but the same is true of the B2BUA approach. Not sure what you're 
    concerned about this being backwards compatible with, exactly.<FONT 
    face=Verdana size=1><SPAN 
    class=212093314-03042002>&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></FONT></FONT></P></BLOCKQUOTE></BLOCKQUOTE>
<P><FONT color=#0000ff><FONT size=2><FONT face=Arial><SPAN 
class=640444710-03042002><FONT face=Verdana size=1><SPAN 
class=147255411-03042002><FONT face=Arial size=2><SPAN 
class=309002312-03042002><FONT face=Verdana size=1><SPAN 
class=212093314-03042002>MW2&gt; RFC2543 clients - how can I ensure the 
subscribers privacy requirements are&nbsp;met if the user&nbsp;has 
an&nbsp;RFC2543 client ? If I'm not worried about this case, then&nbsp;my proxy 
can easily just modify the From field, since dialog matching is based only on 
the 
tags.&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></FONT></FONT></P>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <P><FONT color=#0000ff><FONT face=Arial><FONT size=2><SPAN 
    class=640444710-03042002><SPAN class=147255411-03042002><FONT face=Verdana 
    size=1>MW&gt; The point I'm trying to make is exactly that intermediaries 
    providing privacy *will* often need to be More Than Just A Proxy, and I'm 
    trying to see if others on the list agree with this. Some of the 'Bad 
    Things' that people are concerned about with respect to From/To are 
    happening because people are trying to solve these privacy problems 
    *without* realising that they need More Than Just A Proxy.</FONT><BR><SPAN 
    class=309002312-03042002>[Peterson, Jon]&nbsp;</SPAN></SPAN></SPAN><SPAN 
    class=309002312-03042002>Agreed (but we knew 
    that).&nbsp;</SPAN></FONT></FONT></FONT></P>
    <P><FONT face=Verdana size=1><SPAN class=147255411-03042002><FONT 
    color=#0000ff>With respect to your example, I can't see that you could trust 
    A's UA to do this correctly, and there is no device that the new A-&gt;C 
    call will pass through which could reject it if A does not follow the rules 
    in the new URI. Passing the privacy requirements back to the UA is a good 
    idea, but only works if there are proxies in a position to reject requests 
    where the UA has not obeyed the privacy requirements.<BR></FONT><SPAN 
    class=309002312-03042002><FONT color=#0000ff><FONT face=Arial 
    size=2>[Peterson, Jon]&nbsp;I can't imagine that, in any network likely to 
    interest you, A wouldn't have a hardwired pre-loaded Route to its local 
    outbound that would ensure the A-&gt;C path has some adult supervision, and 
    that the local outbound would be in the redirection response path anyway 
    (from the Vias on the original A-&gt;B request). If it is so exceptionally 
    paranoid about misbehaving UAs, it could enact any number of policies to 
    make sure that nothing goes awry.<FONT face=Verdana size=1><SPAN 
    class=212093314-03042002>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></SPAN></FONT></P></BLOCKQUOTE></BLOCKQUOTE>
<P><FONT face=Verdana size=1><FONT color=#0000ff><SPAN 
class=147255411-03042002><SPAN class=309002312-03042002><FONT size=2><FONT 
size=1><SPAN class=212093314-03042002>MW2&gt; So the outbound proxy&nbsp;keeps a 
note of the Redirection redirection request going back to A and then policies 
new INVITES coming in to spot the redirected call and enforce B's privacy 
request ? So I need to keep this state&nbsp;</SPAN></FONT></FONT><FONT 
size=2><FONT face=Arial><SPAN class=464170412-03042002><SPAN 
class=309002312-03042002><SPAN class=212093314-03042002><FONT face=Verdana 
size=1>information about outstanding redirect requests that have 
been&nbsp;passed to A. I guess it could be done, but it does not sound too 
great.&nbsp;</FONT></SPAN></SPAN></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></P>
<P><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=147255411-03042002><SPAN class=309002312-03042002><FONT size=2><FONT 
face=Arial><SPAN class=464170412-03042002><SPAN class=309002312-03042002><SPAN 
class=212093314-03042002>&nbsp;</SPAN>But&nbsp;</SPAN>I wonder, do the 
regulatory constraints that make this a requirement also levy a hefty fine on A 
if he reveals to C in <SPAN class=309002312-03042002>the&nbsp;</SPAN>ensuing 
<SPAN class=309002312-03042002>voice </SPAN>conversation that he <SPAN 
class=309002312-03042002>had originally dialed&nbsp;</SPAN>B?<SPAN 
class=212093314-03042002><FONT face=Verdana 
size=1>&nbsp;</FONT></SPAN></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></P>
<P><FONT color=#0000ff><SPAN class=147255411-03042002><SPAN 
class=309002312-03042002><FONT face=Verdana size=1><SPAN 
class=464170412-03042002><SPAN class=212093314-03042002>MW2&gt; No, the 
constraint in this case is really on B's Service Provider. If this Service 
Provider offers a forwarding service, then they also ought to offer an 
'Anonymous Forwarding Service' where B's identity is not revealed to C *by the 
network*. There is nothing to stop A revealing B's identity to 
C.</SPAN></SPAN></FONT></SPAN></SPAN></FONT></P>
<P><FONT color=#0000ff><SPAN class=147255411-03042002><SPAN 
class=309002312-03042002><FONT face=Verdana size=1><SPAN 
class=464170412-03042002><SPAN class=212093314-03042002>The problem with just 
using SIP redirection, is that B's Service Provider would then have chosen an 
implementation of the Call Forwarding service which intrinsically causes B's 
identity to be revealed to C. Again, I think the B's Service Provider would be 
on dodgy ground if they tried to represent this as an 'Anonymous Forwarding 
Service' on the basis that 'it wasn't me wot revealed B's identity, it was that 
villian A'.</SPAN></SPAN></FONT></SPAN></SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=212093314-03042002>I'll 
admit it is an open question as to how the identities in SIP will be interpreted 
by regulators. The regulatory constraints apply to third parties (e.g. Service 
Providers) releasing identity information about the end users/subscribers. Is 
the information in the From/To fields being supplied by the Service Provider, or 
the end user themselves ?? From a technical perspective, obviously, it's being 
supplied by the end user, but it's not the technical perspective which is used 
to evaluate the effect of these regulations, it's the customer's perspective, 
which is not related to the technology.</SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=212093314-03042002>If I 
as a Service Provider *require* that the end user insert certain information in 
these fields, then I am&nbsp;clearly responsible for the insertion of the 
information, so I think I inherit some responsibility for preventing privacy 
breachs as a result of this.</SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=212093314-03042002>If I 
require that end users use SIP-compliant clients, then we inherit all the 
requirements of the SIP spec. As I argued before, there is certainly an implicit 
'soft' requirement that the To/From fields should contain the called/calling 
party address.</SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=212093314-03042002>So, I 
would find it hard as a Service Provider to credibly argue that I have no 
responsibility whatsoever for the information placed in these fields by my 
users' equipment, when I have required them to be SIP compliant and when the SIP 
spec implies so strongly that this information will be there. In the case of 
3GPP it is clearer, because the 3GPP spec will explicitly state what will or 
won't be in the To/From fields. That is why it currently explicitly states that 
there should be garbage in there, so I can be sure I'm not responsible for 
it!</SPAN></FONT></P>
<P><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=147255411-03042002><SPAN class=309002312-03042002><FONT size=2><FONT 
face=Arial><SPAN class=464170412-03042002><SPAN 
class=212093314-03042002>&nbsp;</SPAN>When the redirection with the Contact 
address I described <SPAN class=309002312-03042002>in my last 
mail&nbsp;</SPAN>arrives at A, A would have to intentionally countermand it in 
order to reveal B's identity<SPAN class=309002312-03042002> in the 
signaling</SPAN>.&nbsp;<SPAN class=309002312-03042002>But&nbsp;</SPAN>A is of 
course empowered to reveal B's identity through other means anyway. I have a 
hard time seeing what the big deal is.<SPAN class=309002312-03042002>&nbsp;B 
only remains anonymous at A's sufferance anyway.<FONT face=Verdana size=1><SPAN 
class=212093314-03042002>&nbsp;</SPAN></FONT></SPAN></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></P>
<P><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=147255411-03042002><SPAN class=309002312-03042002><FONT size=2><FONT 
face=Arial><SPAN class=464170412-03042002><SPAN class=309002312-03042002><FONT 
face=Verdana size=1><SPAN class=212093314-03042002>MW2&gt; Again, it's the 
distinction between the devices supporting the service (UAs, Proxies etc.) 
releasing the information because this is intrinsic in the Service I am 
offering, and information being released entirely due to the whim of the human 
users themselves. The former is regulated, the latter is freedom of 
speech.</SPAN></FONT></SPAN></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN 
class=212093314-03042002>Anyway, after all that, I was only arguing (a) that in 
a Service Provider model, protection of the subscribers privacy would require 
More Than Just a Proxy and (b) that Anonymous Call Forwarding would require 
'More Than Just a Proxy'. (assuming RFC2543 compatibility in both 
cases).</SPAN></FONT></P>
<P><FONT color=#0000ff face=Verdana size=1><SPAN class=212093314-03042002>In 
both cases, you have provided alternative implementations. For (a) I think your 
proposal is not compatible with 2543 clients, and for (b) it requires MTJAP 
anyway. I actually think that if you drop the 2543 requirement, then you arrive 
at better solutions for all this. The overloading of To/From with two completely 
distinct functions (identity and dialog matching) in 2543 is a real 
pain.</SPAN></FONT></P>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <P><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=147255411-03042002>...Mark</SPAN></FONT></P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C1DB21.AD2A3434--

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From daemon@ns.ietf.org  Wed Apr  3 11:12:03 2002
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Date: Wed, 03 Apr 2002 17:49:07 +0300
From: Jari Arkko <jari.arkko@piuha.net>
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To: "Mahey, Sonit" <SMahey@unispherenetworks.com>
Cc: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
        "'sip@ietf.org'" <sip@ietf.org>,
        "'fluffy@cisco.com'" <fluffy@cisco.com>,
        "'gvelo@astartec.com'" <gvelo@astartec.com>
Subject: Re: [Sip] Residential UA security
References: <9DCB6C9DC7C3D311B835009027DE069F049A08AC@email2.it.west.unispherenetworks.com>
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Mahey, Sonit wrote:


> I did not follow the last para in your response. Could you please explain
> further the "anonymous" authentication mechanism that you are referring to


Bis section 22. It is effective in the sense that you don't need to alert
the user or devote any resources or memory until the other side responds
to your null authentication request in a way that he proves he has seen
you challenge. Essentially, this verifies that the IP address that was
used in the first request is not completely bogus.


> and how it can be effective in DDOS and your thoughts on direct DOS attack
> mitigation, if "anonymous" method is ineffective.

and originally Jonathan wrote:

 > That said, the anonymous authentication mechanism can prevent DDOS
 > attacks on my phone, but not a direct targeted attack from a single
 > source.

To be more exact, the anon method can prevent DDoS and DoS attacks that
use a forged IP address. This is pretty good in the sense that ingress
filtering isn't deployed all over in the Internet, and most attacks can
be prevented by verifying the addresses in this manner.

However, this does not prevent any of the attacks where the attacker is
willing to compromise its own address and position. There is little we
can do about such attacks for SIP or other protocols, except to send
in the police. (Though a virus-infected PC might not care about
its address being revealed, or the police finding it.)

Jari


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From daemon@ns.ietf.org  Wed Apr  3 11:35:54 2002
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From: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [Sip] Draft capturing the Refer security discussion at IETF53
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I've submitted a draft that attempts to capture the presentation
and discussion of the refer security issues at IETF53. It adds a little
more detail to the message contents than shows in the slides. Until it
appears in the archive, you can retrieve it from:

http://www.nostrum.com/~rjsparks/draft-sparks-sip-refer-sec-options-00.txt
http://www.nostrum.com/~rjsparks/draft-sparks-sip-refer-sec-options-00.html

RjS



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Subject: RE: [Sip] Residential UA security
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ok . but how my UA validate an invite message from a proxy ?? anyone can
impersonate my proxy.
in other words, a proxy can validate a UA but can a UA validate a proxy ?

thanks.


-----Mensaje original-----
De: sip-admin@ietf.org [mailto:sip-admin@ietf.org]En nombre de David R.
Oran
Enviado el: martes, 02 de abril de 2002 15:03
Para: 'Cullen Jennings'; gvelo@astartec.com; sip@ietf.org
Asunto: RE: [Sip] Residential UA security


Another possibility is to set your UA to only accept invites from a
known set of previous-hop proxies that you trust and are configured to
execute your chosen call screening policy.

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Tuesday, April 02, 2002 1:00 PM
> To: gvelo@astartec.com; sip@ietf.org
> Subject: RE: [Sip] Residential UA security
>
>
>
> It is an interesting question - it may not be solvable on the
> other hand it may not need to solved as directly as you think.
>
> With a modem set up as an auto dialer, I could make your
> current PSTN phone unusable. I could flood your email box so
> you were likely over quota. I could probably use up on the
> bandwidth on your WAN link. None of these DOS issues seem to
> be a huge problem today. One reason is that there is some
> hope you could trace back and figure out who was doing this
> to you then perhaps filter or take legal action.
>
> Cullen
>
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Gabriel Velo
> > Sent: Tuesday, April 02, 2002 6:23 AM
> > To: sip@ietf.org
> > Subject: [Sip] Residential UA security
> >
> >
> > Dear All.
> >
> > I need some help on this topic:
> >
> > How can i protect a residential UA (my residential Sip
> Phone) from DOS
> > Attack like a flood of INVITE msg ?? In this scenario we
> don't have a
> > validation since we need receive call from
> > anyone.
> > and .... anyone with a low bandwidth connection can make than my
> > phone can't
> > stop of ring !!
> >
> > Thanks in advance.
> >
> > Gabriel.
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on
> current sip Use
> > sipping@ietf.org for new developments on the application of sip
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> sip Use sipping@ietf.org for new developments on the
> application of sip
>


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_______________________________________________
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Date: Wed, 03 Apr 2002 12:10:55 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: [Sip] Residential UA security
References: <000e01c1db2a$c345de90$0901a8c0@roberto>
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Gabriel Velo wrote:
> 
> ok . but how my UA validate an invite message from a proxy ?? anyone can
> impersonate my proxy.
> in other words, a proxy can validate a UA but can a UA validate a proxy
> ?
> 

THis is supported in bis using TLS. The UAS would connect to its proxy
via TLS, and register to receive incoming invites on that connection. It
would validate that the proxy is its own proxy using the TLS procedures,
and then reject any incoming calls that do not arrive on that
connection.

Of course, this requires a long lived tls connection from the client to
the server. Nothing is for free....

-Jonathan R.


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

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From daemon@ns.ietf.org  Wed Apr  3 14:51:05 2002
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From: "Mahey, Sonit" <SMahey@unispherenetworks.com>
To: "'gvelo@astartec.com'" <gvelo@astartec.com>
Cc: "'oran@cisco.com'" <oran@cisco.com>,
        "'fluffy@cisco.com'"
	 <fluffy@cisco.com>,
        "'sip@ietf.org'" <sip@ietf.org>,
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Subject: RE: [Sip] Residential UA security
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Gvelo writes:

ok . but how my UA validate an invite message from a proxy ?? anyone can
impersonate my proxy.
in other words, a proxy can validate a UA but can a UA validate a proxy ?

thanks.
------------------------
[SonitMahey] Sure can. Take a look at Proxy-Authentication in the draft(s)
from James Undery - draft-undery-.....

Sonit Mahey,
Architect - Voice Business Unit,
Unisphere Networks,
10 Technology Park Drive,
Westford, MA 01886,
USA.


======================================= 
This email message is for the sole use of the intended recipient (s) and may
contain confidential and privileged information, including without
limitation, Confidential and/or Proprietary Information belonging to
Unisphere Networks, Inc. Any unauthorized review, use, disclosure or
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From daemon@optimus.ietf.org  Wed Apr  3 15:59:37 2002
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Date: Wed, 03 Apr 2002 14:24:41 -0600
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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To: Dean Willis <dwillis@dynamicsoft.com>
CC: sip@ietf.org, Rohan@cicso.com, jo@ipdialog.com, brian.rosen@marconi.com
Subject: Re: [Sip] Poll: interest in Reason header work
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Dean Willis wrote:

> It looks like UPDATE and Manyfolks are likely to be dependent on the
> Reason header mechanism (imho). Given this (and the other requirements
> discussions that we've heard), how do you people feel about proposing
> this as a SIP WG effort? And yes, the SIP WG can identify work that
> needs to be done to support other SIP WG efforts without having a
> third-party generate a requirements RFC.


Fine with me.  Update (155) and Manyfolks (183) are the two responses
identified in my previous emails as benefitting mostly from the
Reason method.

I have not heard from the authors yet, but it would be good to limit
the Reason header in responses to the above two, as opposed to
making it a general header field for all responses.  Otherwise,
there'll be too many ways to propogate reasons in responses -- the
normal SIP response codes and the Warning header in responses
already exist.

 

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@optimus.ietf.org  Thu Apr  4 05:42:31 2002
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From: "Vesa Torvinen (LMF)" <Vesa.Torvinen@lmf.ericsson.se>
To: "'Mahey, Sonit'" <SMahey@unispherenetworks.com>,
        "'gvelo@astartec.com'" <gvelo@astartec.com>
Cc: "'oran@cisco.com'" <oran@cisco.com>,
        "'fluffy@cisco.com'"
	 <fluffy@cisco.com>,
        "'sip@ietf.org'" <sip@ietf.org>,
        "'jundery@ubiquity.net'" <jundery@ubiquity.net>
Subject: RE: [Sip] Residential UA security
Date: Thu, 4 Apr 2002 12:06:01 +0200 
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We (James Undery et al) got very clear negative feedback for our 
Proxy-to-UAS authentication extensions in Minneapolis. It currenly 
seems that UAS will not be able to validate a Proxy with HTTP 
authentication headers - unless you don't want to put B2BUA's in 
every last-hop proxy. 

Another potential extension to SIP/HTTP authentication would have 
been to allow proxy to forse UAS for authentication with 
terminating calls (because it can already authenticate UAC with 
originating calls). HTTP authentication framework really works 
only for one direction between UA and proxy - three other 
'directions' are uncovered. The case where the clients forse the 
servers to authenticate was never covered by undery-draft(s). 

I think that folks don't care about the last-hop-proxy authentication 
because UAS can always challenge the UAC at the other end e.g. with 
this "anonymous" authentication. This provides some security even if 
you didn't have the TLS connection open with the proxy. However, I 
think that the security related to the last-hop-proxy is still 
somehow problematic issue in SIP. 

Vesa 

> -----Original Message-----
> From: Mahey, Sonit [mailto:SMahey@unispherenetworks.com]
> Sent: 3. huhtikuuta 2002 22:25
> To: 'gvelo@astartec.com'
> Cc: 'oran@cisco.com'; 'fluffy@cisco.com'; 'sip@ietf.org';
> 'jundery@ubiquity.net'
> Subject: RE: [Sip] Residential UA security
> 
> 
> Gvelo writes:
> 
> ok . but how my UA validate an invite message from a proxy ?? 
> anyone can
> impersonate my proxy.
> in other words, a proxy can validate a UA but can a UA 
> validate a proxy ?
> 
> thanks.
> ------------------------
> [SonitMahey] Sure can. Take a look at Proxy-Authentication in 
> the draft(s)
> from James Undery - draft-undery-.....
> 
> Sonit Mahey,
> Architect - Voice Business Unit,
> Unisphere Networks,
> 10 Technology Park Drive,
> Westford, MA 01886,
> USA.
> 
> 
> ======================================= 
> This email message is for the sole use of the intended 
> recipient (s) and may
> contain confidential and privileged information, including without
> limitation, Confidential and/or Proprietary Information belonging to
> Unisphere Networks, Inc. Any unauthorized review, use, disclosure or
> distribution is prohibited. If you are not the intended 
> recipient, please
> contact the sender by reply email and destroy all copies of 
> the original
> message.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Thu Apr  4 09:18:45 2002
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Subject: [Sip] comments on draft-ietf-sip-refer-02.txt
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The draft talks about the use of REFER outside a dialog:

"A REFER request MAY be placed outside the scope of a dialog created =
with an INVITE."

"A UA accepting a well-formed REFER request SHOULD request approval from =
the user to proceed"

I am not sure what kind of approval are we talking about here ?
If a REFER comes from a proxy to a SIP phone or a gateway (with multiple =
lines on it), what are we supposed to do ? Surely not ring the phone ?

------=_NextPart_000_009F_01C1DBB9.7B21CD10
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DCourier>The draft talks about the use of REFER outside =
a=20
dialog:</FONT></DIV>
<DIV><FONT face=3DCourier></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier>"A REFER request MAY be placed outside the =
scope of a=20
dialog created with an INVITE."</FONT></DIV>
<DIV><FONT face=3DCourier></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier>"A UA accepting a well-formed REFER request =
SHOULD=20
request approval from the user to proceed"</FONT></DIV>
<DIV><FONT face=3DCourier></FONT>&nbsp;</DIV>
<DIV><FONT face=3DCourier>I am not sure what kind of approval are we =
talking about=20
here ?</FONT></DIV>
<DIV><FONT face=3DCourier>If a REFER comes from a proxy to a SIP phone =
or a=20
gateway (with multiple lines on it), what are we supposed to do ? Surely =
not=20
ring the phone ?</FONT></DIV></BODY></HTML>

------=_NextPart_000_009F_01C1DBB9.7B21CD10--


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Date: Thu, 4 Apr 2002 10:12:51 EST
Subject: Re: [Sip] Comment, SIP Privacy draft
To: mwatson@nortelnetworks.com, huitema@windows.microsoft.com,
        fandreas@cisco.com
CC: dean.willis@softarmor.com, sip@ietf.org
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In a message dated 4/2/02 9:06:58 AM Eastern Standard Time, 
mwatson@nortelnetworks.com writes:


> What makes you think this ? At present a user receiving malicious calls at 
> work (from the PSTN) can ask for these to be traced. Why should they lose 
> this ability if their employer switches to a SIP-based telecommunications 
> service using the Internet ? (as well as getting worse QoS :-)
> I think if it became widespread that corporations used Internet-based 
> telecommunications which did not support malicious call trace, then 
> regulators, not to say consumer groups and groups representing employees 
> might become concerned.
> I would think we could make the SIP-based service at least as good as the 
> PSTN in this regard. 
> The 'trusted third party' in the form of the public service provider 
> performs a valuable service in mediating between the legitimate desire or 
> the caller to remain anonymous, and the equally legitimate desire for 
> called parties to be protected from malicious calls. A third party of some 
> kind is required to guard the privacy of the caller, whilst ensuring the 
> tracebility of the call.
> Now there is no reason why this 'third party' needs to be a public service 
> provider. The same 'third party' function could be provided by proxies 
> owned by the corporations between which calls are being made.
> We should be clear that if users choose to receive calls directly, without 
> such a 'trusted third party', then they will not receive the protection 
> from malicious calls that they are used to on the PSTN. It may be fine for 
> individual users to make this choice, but it is not something that anyone 
> providing service to many users (be that a commercial service, or an 
> employer providing telecommunications to their employees) could choose to 
> do.
> So, we need mechanisms to enable this in as many scenarios as possible. 
> 

Mark,

I like your synopsis above of the situation. Likewise, you could add, if 
users choose to originate their calls without such a "trusted third party", 
then they will not receive the protection from having their identity passed 
to some unknown user that they are used to on the PSTN.

There needs to be an understanding that we are addressing two different 
architectures in these discussions:

1. With a trusted third party (Read: service provider) who guarantees 
privacy/protection/etc. (usually because the law says they must)
2. Without a trusted third party (Read: Wild-wild west) in which no one using 
it is guaranteed of anything.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/2/02 9:06:58 AM Eastern Standard Time, mwatson@nortelnetworks.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">What makes you think this ? At present a user receiving malicious calls at work (from the PSTN) can ask for these to be traced. Why should they lose this ability if their employer switches to a SIP-based telecommunications service using the Internet ? (as well as getting worse QoS :-)
<BR>I think if it became widespread that corporations used Internet-based telecommunications which did not support malicious call trace, then regulators, not to say consumer groups and groups representing employees might become concerned.
<BR>I would think we could make the SIP-based service at least as good as the PSTN in this regard. 
<BR>The 'trusted third party' in the form of the public service provider performs a valuable service in mediating between the legitimate desire or the caller to remain anonymous, and the equally legitimate desire for called parties to be protected from malicious calls. A third party of some kind is required to guard the privacy of the caller, whilst ensuring the tracebility of the call.
<BR>Now there is no reason why this 'third party' needs to be a public service provider. The same 'third party' function could be provided by proxies owned by the corporations between which calls are being made.
<BR>We should be clear that if users choose to receive calls directly, without such a 'trusted third party', then they will not receive the protection from malicious calls that they are used to on the PSTN. It may be fine for individual users to make this choice, but it is not something that anyone providing service to many users (be that a commercial service, or an employer providing telecommunications to their employees) could choose to do.
<BR>So, we need mechanisms to enable this in as many scenarios as possible. 
<BR></BLOCKQUOTE>
<BR>
<BR>Mark,
<BR>
<BR>I like your synopsis above of the situation. Likewise, you could add, if users choose to originate their calls without such a "trusted third party", then they will not receive the protection from having their identity passed to some unknown user that they are used to on the PSTN.
<BR>
<BR>There needs to be an understanding that we are addressing two different architectures in these discussions:
<BR>
<BR>1. With a trusted third party (Read: service provider) who guarantees privacy/protection/etc. (usually because the law says they must)
<BR>2. Without a trusted third party (Read: Wild-wild west) in which no one using it is guaranteed of anything.
<BR>
<BR>Mike
<BR></FONT></HTML>

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Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Medhavi Bhatia <mbhatia@nextone.com>
Cc: sip@ietf.org
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On Thu, 2002-04-04 at 08:17, Medhavi Bhatia wrote:
> The draft talks about the use of REFER outside a dialog:
>  
> "A REFER request MAY be placed outside the scope of a dialog created
> with an INVITE."
>  
> "A UA accepting a well-formed REFER request SHOULD request approval from
> the user to proceed"
>  
> I am not sure what kind of approval are we talking about here ?
Whatever is appropriate for the device and application - it may
be a direct prompt of the user, it may be a matter of using policy
that the user has placed in the device. 
> If a REFER comes from a proxy to a SIP phone or a gateway (with multiple
> lines on it), what are we supposed to do ? Surely not ring the phone ?
Why not? I've heard of phone implementations that provide a special
alert and put a "X is requesting that you call Y, proceed?" type prompt
on the display. On a gateway, your interface is obviously more limited.
If you were ambitious, you could invoke media/ivr servers, alert the
black phone, play the prompt above to it and ask for what to do.Or you
could decide its not worth extending this kind of functionality to the
old phones. There's a lot of room for innovation in between.

RjS


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From daemon@optimus.ietf.org  Thu Apr  4 11:16:32 2002
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References: <00a201c1dbe3$64720e20$0100000a@deewana> <1017933016.1187.11.camel@dhcp222.dfw.dynamicsoft.com>
Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
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I think by the IVR approach, you mean REFER in a
dialog context.

That kind of an approach seems to be much more
preferable, as it works for all endpoints unlike the one w/o
a dialog which tries to do similar things. The caller obviously
wont know that the destination is a smart phone or not.

Ironically, w/o dialog seems to be syntactically more general.
Maybe its use should be restricted somehow for interop
reasons or so ?


----- Original Message -----
From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "Medhavi Bhatia" <mbhatia@nextone.com>
Cc: <sip@ietf.org>
Sent: Thursday, April 04, 2002 10:10 AM
Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt


> On Thu, 2002-04-04 at 08:17, Medhavi Bhatia wrote:
> > The draft talks about the use of REFER outside a dialog:
> >
> > "A REFER request MAY be placed outside the scope of a dialog created
> > with an INVITE."
> >
> > "A UA accepting a well-formed REFER request SHOULD request approval from
> > the user to proceed"
> >
> > I am not sure what kind of approval are we talking about here ?
> Whatever is appropriate for the device and application - it may
> be a direct prompt of the user, it may be a matter of using policy
> that the user has placed in the device.
> > If a REFER comes from a proxy to a SIP phone or a gateway (with multiple
> > lines on it), what are we supposed to do ? Surely not ring the phone ?
> Why not? I've heard of phone implementations that provide a special
> alert and put a "X is requesting that you call Y, proceed?" type prompt
> on the display. On a gateway, your interface is obviously more limited.
> If you were ambitious, you could invoke media/ivr servers, alert the
> black phone, play the prompt above to it and ask for what to do.Or you
> could decide its not worth extending this kind of functionality to the
> old phones. There's a lot of room for innovation in between.
>
> RjS
>
>


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From daemon@optimus.ietf.org  Thu Apr  4 11:21:57 2002
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Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Medhavi Bhatia <mbhatia@nextone.com>
Cc: sip@ietf.org
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On Thu, 2002-04-04 at 09:54, Medhavi Bhatia wrote:
> I think by the IVR approach, you mean REFER in a
> dialog context.
No - I didn't.

> 
> That kind of an approach seems to be much more
> preferable, as it works for all endpoints unlike the one w/o
> a dialog which tries to do similar things. The caller obviously
> wont know that the destination is a smart phone or not.

A caller doesn't know if the destination will accept an INVITE
either. That's what the response part of the SIP transaction is for.
If you send an out-of-band refer to something that by policy or
reason of physical limitations can't deal with it, it will decline.

REFER is a primitive, not an application. We've been able to build
one application on top of REFER (transfer) without invoking Require:.
Some future application may need to know that the far end is going
to do something special with a REFER before it sends it - that 
application will use a Require: token.

> 
> Ironically, w/o dialog seems to be syntactically more general.
> Maybe its use should be restricted somehow for interop
> reasons or so ?
I've seen no reason to introduce any such restriction.

> 
> 
> ----- Original Message -----
> From: "Robert Sparks" <rsparks@dynamicsoft.com>
> To: "Medhavi Bhatia" <mbhatia@nextone.com>
> Cc: <sip@ietf.org>
> Sent: Thursday, April 04, 2002 10:10 AM
> Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
> 
> 
> > On Thu, 2002-04-04 at 08:17, Medhavi Bhatia wrote:
> > > The draft talks about the use of REFER outside a dialog:
> > >
> > > "A REFER request MAY be placed outside the scope of a dialog created
> > > with an INVITE."
> > >
> > > "A UA accepting a well-formed REFER request SHOULD request approval from
> > > the user to proceed"
> > >
> > > I am not sure what kind of approval are we talking about here ?
> > Whatever is appropriate for the device and application - it may
> > be a direct prompt of the user, it may be a matter of using policy
> > that the user has placed in the device.
> > > If a REFER comes from a proxy to a SIP phone or a gateway (with multiple
> > > lines on it), what are we supposed to do ? Surely not ring the phone ?
> > Why not? I've heard of phone implementations that provide a special
> > alert and put a "X is requesting that you call Y, proceed?" type prompt
> > on the display. On a gateway, your interface is obviously more limited.
> > If you were ambitious, you could invoke media/ivr servers, alert the
> > black phone, play the prompt above to it and ask for what to do.Or you
> > could decide its not worth extending this kind of functionality to the
> > old phones. There's a lot of room for innovation in between.
> >
> > RjS
> >
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Thu Apr  4 11:53:23 2002
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        fandreas@cisco.com
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Subject: RE: [Sip] Comment, SIP Privacy draft
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...bit quiet today ... 

Mike Pierce wrote:
 
> Likewise, you could add, if users choose to originate their calls without
such a "trusted third 
> party", then they will not receive the   protection from having their
identity passed to some 
> unknown user that they are used to on the PSTN. 

If there's no "third party" then the user does not need to include any
identity information in the message at all. The end-to-end model tips the
balance completely to the side of the caller & their desire to remain
anonymous (or in the absence of a PKI, to mis-represent themselves).

> There needs to be an understanding that we are addressing two different
architectures in these
> discussions: 
>
> 1. With a trusted third party (Read: service provider) who guarantees
privacy/protection/etc. 
> (usually because the law says they must) 
> 2. Without a trusted third party (Read: Wild-wild west) in which no one
using it is guaranteed 
> of anything. 

Absolutely, and I think our generic privacy requirements draft will need to
start with a description of the models we are addressing and an analysis of
exactly *whose* privacy we are protecting in each model.

For example, the concept of 'subscriber' only exists in a 'service provider'
model.

I wonder whether the case of enterprise networks sending calls over the
Internet is a third model - its characteristics are slightly different.
There is no 'Service Provider' as such, but the enterprise(s) are 'providing
a service' to the users.

...Mark

> Mike 

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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] Comment, SIP Privacy draft</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>...bit quiet today ... </FONT>
</P>

<P><FONT SIZE=3D2>Mike Pierce wrote:</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; Likewise, you could add, if users choose to =
originate their calls without such a &quot;trusted third </FONT>
<BR><FONT SIZE=3D2>&gt; party&quot;, then they will not receive =
the&nbsp;&nbsp; protection from having their identity passed to some =
</FONT>
<BR><FONT SIZE=3D2>&gt; unknown user that they are used to on the PSTN. =
</FONT>
</P>

<P><FONT SIZE=3D2>If there's no &quot;third party&quot; then the user =
does not need to include any identity information in the message at =
all. The end-to-end model tips the balance completely to the side of =
the caller &amp; their desire to remain anonymous (or in the absence of =
a PKI, to mis-represent themselves).</FONT></P>

<P><FONT SIZE=3D2>&gt; There needs to be an understanding that we are =
addressing two different architectures in these</FONT>
<BR><FONT SIZE=3D2>&gt; discussions: </FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; 1. With a trusted third party (Read: service =
provider) who guarantees privacy/protection/etc. </FONT>
<BR><FONT SIZE=3D2>&gt; (usually because the law says they must) =
</FONT>
<BR><FONT SIZE=3D2>&gt; 2. Without a trusted third party (Read: =
Wild-wild west) in which no one using it is guaranteed </FONT>
<BR><FONT SIZE=3D2>&gt; of anything. </FONT>
</P>

<P><FONT SIZE=3D2>Absolutely, and I think our generic privacy =
requirements draft will need to start with a description of the models =
we are addressing and an analysis of exactly *whose* privacy we are =
protecting in each model.</FONT></P>

<P><FONT SIZE=3D2>For example, the concept of 'subscriber' only exists =
in a 'service provider' model.</FONT>
</P>

<P><FONT SIZE=3D2>I wonder whether the case of enterprise networks =
sending calls over the Internet is a third model - its characteristics =
are slightly different. There is no 'Service Provider' as such, but the =
enterprise(s) are 'providing a service' to the users.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Mike </FONT>
</P>

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From: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [Sip] REFER security options - removing Referred-By
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I want to explore the option of removing Referred-By a little.

As the refer-sec-options draft points out, traditional phone
systems don't provide the kind of information Referred-By is
trying to provide during a transfer. We've set our goals higher
for SIP transfer, which is a good thing to do, but is it necessary
for a base version of the application? Would a transfer application
that didn't tell the transfer target who initiated the transfer be
sufficient?

Perhaps the Referred-By functionality should be an extension (Requires:)
to REFER instead of part of its base definition?

This would remove the security issue with base REFER (B's triggered
request would be indistinguishable from a request that B initiated on
its own - there is no claim in the request that the request exits
because A asked for it to happen). 

The current work on securing Referred-By would continue in an extension
draft, but advancement of REFER would not be blocked on its completion.


RjS


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From daemon@optimus.ietf.org  Thu Apr  4 12:18:28 2002
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Date: Thu, 4 Apr 2002 09:01:41 -0800 (Pacific Standard Time)
From: Rohan Mahy <rohan@cisco.com>
To: sipping@ietf.org, <sip@ietf.org>
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Subject: [Sip] Poll for interest in an interim SIP/SIPPING meeting
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Hi,

I've heard from some folks that they would like to have another interim
SIP/SIPPING meeting like we did in Dallas between IETF 49 and 50.  I'd
like to here from folks *privately* please if they are interested in such
an interim meeting, probably during the month of May.  Also if you are
strongly opposed to such a meeting, I would like to hear from you
*privately* as well.

As for the possible locations, I am currently entertaining both Flagstaff,
Arizona, and Boston.  I can get a very good room rate for one, and the
other is easier for most folks to get to.  If you send me *private* mail
in support of an interim meeting, your location suggestion is also appreciated.

As always, any interim meeting has to be announced on ietf-announce at
least 30 days before the meeting, so please get me your comments early.
For your convenience, you can just click on the apropriate link.

Yes Reply:  mailto:rohan@cisco.com?Subject=[sip-interim-yes]

No  Reply:  mailto:rohan@cisco.com?Subject=[sip-interim-no]

thanks,
-rohan


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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Thu Apr  4 12:46:13 2002
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Jon Peterson wrote:
> Until sip-privacy removes the 'screen=' parameter, sip-privacy still has
> sunny-day handling for the insertion of RPID headers by untrusted UAs. I'm
> not suggesting that we add text saying that untrusted UAs MUST NOT add the
> RPID. I'm suggesting that the concept of an 'untrusted' RPID be expunged (by
> removing the 'screen' concept and the associated protocol apparatus), and
> that proxies MUST remove RPIDs they receive from untrusted sources. I think
> would be much more effective.

The compelling case I still see for the screen parameter is due to the
extensibility of identity types.  If an RPID header arrives at a proxy
with an unknown identity type (only because that proxy has not yet
received its upgrade), it does not seem reasonable to strip it.
Otherwise, a new identity type can't be used until _every_ proxy
in the path has been updated.

Bill Marshall
wtm@research.att.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Apr  4 13:39:05 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
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Bill makes a good point. Plus it's worryingly quiet today, so let's see if I
can summarise again...

There are actually two applications for 'screen=no' RPID which we have been
discussing:
1) RPID inserted by a UA, to give a 'hint' to the outbound proxy as to which
identity it would like to be known by
2) RPID inserted by a Proxy with 'screen=yes' which is later downgraded to
'screen=no' because it passes into another trust domain

(1) is required by 3GPP - well *a* solution is required, not necessarily
RPID. Some people have suggested using unsolicited Proxy-Authorization
instead, but as this function has nothing to do with authentication, I'm not
sure this is appropriate - does anyone disagree, or have another suggestion
?

(2) is a question of whether such 'screen=no' information has any value. We
had a long thread (the Iraqi thread) about whether this would be useful for
Call Trace, which I think foundered on the inability to distinguish between
a type (2) 'screen=no' RPID and a type (1) 'screen=no' RPID which has made
its way unchanged from a UAC which is trying to mis-represent itself.

So the remaining question is whether type (2) 'screen=no' RPID is worth
displaying to the user (with appropriate health warnings) - I can't see how
it's any less valuable than the From: field, and everyone wants to display
that.


I do think that the discussion about (1) above, and the 3GPP requirement, is
really quite a distinct issue from the 'Bad Things' that are presently in
the 3GPP specifications, namely the obfuscation of the To/From fields. This
is much more related to our other thread on how to provide privacy services
and the requirement to modify To/From to provide certain privacy services.

I think that if we were agreed that modification of To/From *is* required in
order to provide (certain) privacy services (and the device that does this
thus needs to be More Than Just A Proxy), then there is no excuse for always
obfuscating To/From i.e. I don't think the 'Bad Things' and RPID are in fact
related as has been claimed.

Regards...Mark



> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: 04 April 2002 18:25
> To: sip@ietf.org
> Subject: RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Jon Peterson wrote:
> > Until sip-privacy removes the 'screen=' parameter, 
> sip-privacy still has
> > sunny-day handling for the insertion of RPID headers by 
> untrusted UAs. I'm
> > not suggesting that we add text saying that untrusted UAs 
> MUST NOT add the
> > RPID. I'm suggesting that the concept of an 'untrusted' 
> RPID be expunged (by
> > removing the 'screen' concept and the associated protocol 
> apparatus), and
> > that proxies MUST remove RPIDs they receive from untrusted 
> sources. I think
> > would be much more effective.
> 
> The compelling case I still see for the screen parameter is due to the
> extensibility of identity types.  If an RPID header arrives at a proxy
> with an unknown identity type (only because that proxy has not yet
> received its upgrade), it does not seem reasonable to strip it.
> Otherwise, a new identity type can't be used until _every_ proxy
> in the path has been updated.
> 
> Bill Marshall
> wtm@research.att.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] Comment, SIP Privacy draft</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Bill makes a good point. Plus it's worryingly quiet =
today, so let's see if I can summarise again...</FONT>
</P>

<P><FONT SIZE=3D2>There are actually two applications for 'screen=3Dno' =
RPID which we have been discussing:</FONT>
<BR><FONT SIZE=3D2>1) RPID inserted by a UA, to give a 'hint' to the =
outbound proxy as to which identity it would like to be known by</FONT>
<BR><FONT SIZE=3D2>2) RPID inserted by a Proxy with 'screen=3Dyes' =
which is later downgraded to 'screen=3Dno' because it passes into =
another trust domain</FONT></P>

<P><FONT SIZE=3D2>(1) is required by 3GPP - well *a* solution is =
required, not necessarily RPID. Some people have suggested using =
unsolicited Proxy-Authorization instead, but as this function has =
nothing to do with authentication, I'm not sure this is appropriate - =
does anyone disagree, or have another suggestion ?</FONT></P>

<P><FONT SIZE=3D2>(2) is a question of whether such 'screen=3Dno' =
information has any value. We had a long thread (the Iraqi thread) =
about whether this would be useful for Call Trace, which I think =
foundered on the inability to distinguish between a type (2) =
'screen=3Dno' RPID and a type (1) 'screen=3Dno' RPID which has made its =
way unchanged from a UAC which is trying to mis-represent =
itself.</FONT></P>

<P><FONT SIZE=3D2>So the remaining question is whether type (2) =
'screen=3Dno' RPID is worth displaying to the user (with appropriate =
health warnings) - I can't see how it's any less valuable than the =
From: field, and everyone wants to display that.</FONT></P>
<BR>

<P><FONT SIZE=3D2>I do think that the discussion about (1) above, and =
the 3GPP requirement, is really quite a distinct issue from the 'Bad =
Things' that are presently in the 3GPP specifications, namely the =
obfuscation of the To/From fields. This is much more related to our =
other thread on how to provide privacy services and the requirement to =
modify To/From to provide certain privacy services.</FONT></P>

<P><FONT SIZE=3D2>I think that if we were agreed that modification of =
To/From *is* required in order to provide (certain) privacy services =
(and the device that does this thus needs to be More Than Just A =
Proxy), then there is no excuse for always obfuscating To/From i.e. I =
don't think the 'Bad Things' and RPID are in fact related as has been =
claimed.</FONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: William Marshall [<A =
HREF=3D"mailto:wtm@research.att.com">mailto:wtm@research.att.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>&gt; Sent: 04 April 2002 18:25</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] Comment, SIP Privacy =
draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jon Peterson wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Until sip-privacy removes the 'screen=3D' =
parameter, </FONT>
<BR><FONT SIZE=3D2>&gt; sip-privacy still has</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; sunny-day handling for the insertion of =
RPID headers by </FONT>
<BR><FONT SIZE=3D2>&gt; untrusted UAs. I'm</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; not suggesting that we add text saying =
that untrusted UAs </FONT>
<BR><FONT SIZE=3D2>&gt; MUST NOT add the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; RPID. I'm suggesting that the concept of =
an 'untrusted' </FONT>
<BR><FONT SIZE=3D2>&gt; RPID be expunged (by</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; removing the 'screen' concept and the =
associated protocol </FONT>
<BR><FONT SIZE=3D2>&gt; apparatus), and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that proxies MUST remove RPIDs they =
receive from untrusted </FONT>
<BR><FONT SIZE=3D2>&gt; sources. I think</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; would be much more effective.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The compelling case I still see for the screen =
parameter is due to the</FONT>
<BR><FONT SIZE=3D2>&gt; extensibility of identity types.&nbsp; If an =
RPID header arrives at a proxy</FONT>
<BR><FONT SIZE=3D2>&gt; with an unknown identity type (only because =
that proxy has not yet</FONT>
<BR><FONT SIZE=3D2>&gt; received its upgrade), it does not seem =
reasonable to strip it.</FONT>
<BR><FONT SIZE=3D2>&gt; Otherwise, a new identity type can't be used =
until _every_ proxy</FONT>
<BR><FONT SIZE=3D2>&gt; in the path has been updated.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Bill Marshall</FONT>
<BR><FONT SIZE=3D2>&gt; wtm@research.att.com</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: William Marshall <wtm@research.att.com>
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Subject: RE: [Sip] Comment, SIP Privacy draft
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Mark Watson wrote:
> 1) RPID inserted by a UA, to give a 'hint' to the outbound proxy as to which
> identity it would like to be known by

Regarding the issue of UA inserting RPID, versus the proxy inserting 
RPID, ---

There was a lengthy discussion around bis-WGLC about proxies
modifying the SDP (not allowed according to bis) and the fact that
firewall control proxies do.  The conclusion, IIRC, was that the
definition acording the spec would not always be consistent with
the marketing term for a product -- mainly due to the fact that
firewall control proxies did not want to be lumped in the same
category of "bad things" as B2BUAs.

I see the same thing likely here.  Whether the box that interfaces
to the user is a UA, or a "UA-with-integrated-proxy", it will certainly
be marketed as a UA.  If the market for the device requires that it 
do functions that are only allowed to be done by a proxy, it will
be done.  Perhaps the term will then be "B2BUA with Proxy".  The
point is, it shouldn't matter how the architectural elements are combined
and optimized to do the job.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] Comment, SIP Privacy draft
Date: Thu, 4 Apr 2002 19:02:50 +0100 

Bill makes a good point. Plus it's worryingly quiet today, so let's see if I
can summarise again...

There are actually two applications for 'screen=no' RPID which we have been
discussing:
1) RPID inserted by a UA, to give a 'hint' to the outbound proxy as to which
identity it would like to be known by
2) RPID inserted by a Proxy with 'screen=yes' which is later downgraded to
'screen=no' because it passes into another trust domain

(1) is required by 3GPP - well *a* solution is required, not necessarily
RPID. Some people have suggested using unsolicited Proxy-Authorization
instead, but as this function has nothing to do with authentication, I'm not
sure this is appropriate - does anyone disagree, or have another suggestion
?

(2) is a question of whether such 'screen=no' information has any value. We
had a long thread (the Iraqi thread) about whether this would be useful for
Call Trace, which I think foundered on the inability to distinguish between
a type (2) 'screen=no' RPID and a type (1) 'screen=no' RPID which has made
its way unchanged from a UAC which is trying to mis-represent itself.

So the remaining question is whether type (2) 'screen=no' RPID is worth
displaying to the user (with appropriate health warnings) - I can't see how
it's any less valuable than the From: field, and everyone wants to display
that.


I do think that the discussion about (1) above, and the 3GPP requirement, is
really quite a distinct issue from the 'Bad Things' that are presently in
the 3GPP specifications, namely the obfuscation of the To/From fields. This
is much more related to our other thread on how to provide privacy services
and the requirement to modify To/From to provide certain privacy services.

I think that if we were agreed that modification of To/From *is* required in
order to provide (certain) privacy services (and the device that does this
thus needs to be More Than Just A Proxy), then there is no excuse for always
obfuscating To/From i.e. I don't think the 'Bad Things' and RPID are in fact
related as has been claimed.

Regards...Mark

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Subject: Re: [Sip] Comment, SIP Privacy draft
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In a message dated 4/4/02 11:33:48 AM Eastern Standard Time, 
mwatson@nortelnetworks.com writes:

> 
> Mike Pierce wrote:
> 
> > Likewise, you could add, if users choose to originate their calls without
> such a "trusted third 
> > party", then they will not receive the   protection from having their
> identity passed to some 
> > unknown user that they are used to on the PSTN. 
> 
> (Mark answered): If there's no "third party" then the user does not need to 
> include any
> identity information in the message at all. The end-to-end model tips the
> balance completely to the side of the caller & their desire to remain
> anonymous (or in the absence of a PKI, to mis-represent themselves).

But as has been noted before, the IP address itself may be enough to violate 
someone's privacy. And the end user can't leave that out. If that is to be 
hidden in certain cases, there needs to be another device in between. 
Likewise, if the calling party identity is not provided, in many networks the 
call will never be carried, due to the requirements at the terminating end.

> 
> (Mark said): For example, the concept of 'subscriber' only exists in a 
> 'service provider'
> model.

And I would argue that, without a "subscriber" or a "service provider", there 
is no concept of a service being provided. And most of what we're talking 
around are "services". So that is a real quandry. I would suggest that we 
further expand on the two situations I listed as follows:

1. With a trusted third party (Read: service provider) which provides 
services, one of those services being to guarantee privacy/protection/etc.
2. Without a trusted third party (Read: Wild-wild west) meaning that there 
are no services being provided. End users do anything they want and no one is 
guaranteed of anything (e.g., privacy, protection, or that anything they try 
to do will neccesarily function as they expect).

Maybe all discussions need to be split between these two cases, since they 
are so much different.

> 
> (Mark asked): I wonder whether the case of enterprise networks sending 
> calls over the
> Internet is a third model - its characteristics are slightly different.
> There is no 'Service Provider' as such, but the enterprise(s) are 'providing
> a service' to the users.

Yes, I would consider the enterprise network as the "service provider" even 
if distinct parts of it are interconnected by some other IP network. This 
includes two cooperating and trusting enterprise networks. 
Privacy/authentication/encryption/etc. between various parts of the 
enterprise network(s) are the responsibility of that network(s). The user 
doesn't need to deal with these issues. It only needs to worry about between 
itself and its "trusted third party" that is providing service.

Mike (again)


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/4/02 11:33:48 AM Eastern Standard Time, mwatson@nortelnetworks.com writes:
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>Mike Pierce wrote:
<BR>
<BR>&gt; Likewise, you could add, if users choose to originate their calls without
<BR>such a "trusted third 
<BR>&gt; party", then they will not receive the &nbsp;&nbsp;protection from having their
<BR>identity passed to some 
<BR>&gt; unknown user that they are used to on the PSTN. 
<BR>
<BR>(Mark answered): If there's no "third party" then the user does not need to include any
<BR>identity information in the message at all. The end-to-end model tips the
<BR>balance completely to the side of the caller &amp; their desire to remain
<BR>anonymous (or in the absence of a PKI, to mis-represent themselves).</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">But as has been noted before, the IP address itself may be enough to violate someone's privacy. And the end user can't leave that out. If that is to be hidden in certain cases, there needs to be another device in between. Likewise, if the calling party identity is not provided, in many networks the call will never be carried, due to the requirements at the terminating end.
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>(Mark said): For example, the concept of 'subscriber' only exists in a 'service provider'
<BR>model.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">And I would argue that, without a "subscriber" or a "service provider", there is no concept of a service being provided. And most of what we're talking around are "services". So that is a real quandry. I would suggest that we further expand on the two situations I listed as follows:
<BR>
<BR>1. With a trusted third party (Read: service provider) which provides services, one of those services being to guarantee privacy/protection/etc.
<BR>2. Without a trusted third party (Read: Wild-wild west) meaning that there are no services being provided. End users do anything they want and no one is guaranteed of anything (e.g., privacy, protection, or that anything they try to do will neccesarily function as they expect).
<BR>
<BR>Maybe all discussions need to be split between these two cases, since they are so much different.
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>(Mark asked): I wonder whether the case of enterprise networks sending calls over the
<BR>Internet is a third model - its characteristics are slightly different.
<BR>There is no 'Service Provider' as such, but the enterprise(s) are 'providing
<BR>a service' to the users.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">Yes, I would consider the enterprise network as the "service provider" even if distinct parts of it are interconnected by some other IP network. This includes two cooperating and trusting enterprise networks. Privacy/authentication/encryption/etc. between various parts of the enterprise network(s) are the responsibility of that network(s). The user doesn't need to deal with these issues. It only needs to worry about between itself and its "trusted third party" that is providing service.
<BR>
<BR>Mike (again)
<BR></FONT></HTML>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>
Cc: <sip@ietf.org>
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva cy draft)
Date: Thu, 4 Apr 2002 17:00:55 -0600
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> RFC2543 clients - how can I ensure the subscribers privacy
requirements are met if the user has an RFC2543 client ? If I'm not
worried about this case, then my proxy can easily just modify the From
field, since dialog matching is based only on the tags.  
 
You can't. You can NEVER prevent a client from giving away his identity.
For exampe, they might add  an SMIME body conatining their name and a
URL to their home-nudie-pix site. At the very best, the privacy draft
documents a means by which a client requests the server not to add
additional private information over and above that which the client has
already seen fit to disclose.
 
The UA is reponsible for the actions of the UA. The proxy is only
responsible for the actions of the proxy . . .
 
 
--
Dean

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT color=3D#0000ff>&gt; RFC2543 clients - =
how can I=20
ensure the subscribers privacy requirements are&nbsp;met if the =
user&nbsp;has=20
an&nbsp;RFC2543 client ? If I'm not worried about this case, =
then&nbsp;my proxy=20
can easily just modify the From field, since dialog matching is based =
only on=20
the tags.&nbsp;<SPAN class=3D801245021-04042002><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></=
FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT color=3D#0000ff><SPAN=20
class=3D801245021-04042002></SPAN></FONT></SPAN></FONT></SPAN></FONT></SP=
AN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT color=3D#0000ff><SPAN=20
class=3D801245021-04042002><FONT face=3DArial size=3D2>You can't. You =
can NEVER=20
prevent a client from giving away his identity.&nbsp;For exampe, they =
might add=20
&nbsp;an SMIME body conatining their name and a URL to their =
home-nudie-pix=20
site. At the very best, the privacy draft documents a means by which a=20
client</FONT>&nbsp;<FONT face=3DArial size=3D2>requests the server not =
to add=20
additional private information over and above that which the client has =
already=20
seen fit to=20
disclose.</FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT><=
/SPAN></FONT></FONT></DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D801245021-04042002></SPAN></FONT></SPAN></FONT></SPAN></FONT></SP=
AN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D801245021-04042002>The UA is reponsible for the actions of the =
UA. The=20
proxy is only responsible for the actions of the proxy . .=20
.</SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></SPAN></FONT></=
FONT></DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D640444710-03042002><FONT=20
face=3DVerdana size=3D1><SPAN class=3D147255411-03042002><FONT =
face=3DArial size=3D2><SPAN=20
class=3D309002312-03042002><FONT face=3DVerdana size=3D1><SPAN=20
class=3D212093314-03042002><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D801245021-04042002></SPAN></FONT></SPAN></FONT></SPAN></FONT></SP=
AN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
class=3D147255411-03042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
class=3D147255411-03042002><SPAN=20
class=3D801245021-04042002>--</SPAN></SPAN></FONT></DIV>
<DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
class=3D147255411-03042002><SPAN=20
class=3D801245021-04042002>Dean</SPAN></SPAN></FONT></DIV></BODY></HTML>

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Date: Thu, 4 Apr 2002 15:33:47 -0800 (Pacific Standard Time)
From: Rohan Mahy <rohan@cisco.com>
To: sip@ietf.org, <sipping@ietf.org>
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Subject: [Sip] possible agenda/motivation for interim meeting
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Hi,

A couple folks have asked for my motivation for suggesting an interim.
Basically we've got a lot of stuff on our plate in both WGs, and much of
it is in that midway done phase were face time is most valuable.
An interim would give us time to resolve several sticky technical issues
and then get drafts out before the cutoff date for IETF54.  I think
we could have a much more productive meeting in Japan as a result.

Topics in serious need of face time:

- the longer term privacy solution
- a whole raft of security issues
- REFER, replaces, and cc-transfer
- call info and conf info packages
- multiparty and conferencing requirements/frameworks
- possibly more ISUP interworking issues
- possibly a working session to revise the AAA requirements draft

The last two would depend on critical mass of folks willing to work on
these.

thanks,
-rohan



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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Apr  4 19:01:49 2002
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Message-ID: <018901c1dc34$c31a37d0$0100000a@deewana>
From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: "Robert Sparks" <rsparks@dynamicsoft.com>
Cc: <sip@ietf.org>
References: <00a201c1dbe3$64720e20$0100000a@deewana><1017933016.1187.11.camel@dhcp222.dfw.dynamicsoft.com> <002401c1dbf1$05900390$1fe4b4cc@laptopeng3> <1017936505.1186.39.camel@dhcp222.dfw.dynamicsoft.com>
Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
Date: Thu, 4 Apr 2002 18:59:30 -0500
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I am not sure we are talking about the same thing, so let me describe
the scenario:

A and B are two endpoints not involved in any call. So A sends
a REFER to B. (Both A and B support the REFER).

Now going back to your reply, I dont understand how you can
alert B and play an ivr etc, without first setting up a dialog with it.

Also, I am confused how the "Require:" header fits into this
discussion.

----- Original Message -----
From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "Medhavi Bhatia" <mbhatia@nextone.com>
Cc: <sip@ietf.org>
Sent: Thursday, April 04, 2002 11:08 AM
Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt


> On Thu, 2002-04-04 at 09:54, Medhavi Bhatia wrote:
> > I think by the IVR approach, you mean REFER in a
> > dialog context.
> No - I didn't.
>
> >
> > That kind of an approach seems to be much more
> > preferable, as it works for all endpoints unlike the one w/o
> > a dialog which tries to do similar things. The caller obviously
> > wont know that the destination is a smart phone or not.
>
> A caller doesn't know if the destination will accept an INVITE
> either. That's what the response part of the SIP transaction is for.
> If you send an out-of-band refer to something that by policy or
> reason of physical limitations can't deal with it, it will decline.
>
> REFER is a primitive, not an application. We've been able to build
> one application on top of REFER (transfer) without invoking Require:.
> Some future application may need to know that the far end is going
> to do something special with a REFER before it sends it - that
> application will use a Require: token.
>
> >
> > Ironically, w/o dialog seems to be syntactically more general.
> > Maybe its use should be restricted somehow for interop
> > reasons or so ?
> I've seen no reason to introduce any such restriction.
>
> >
> >
> > ----- Original Message -----
> > From: "Robert Sparks" <rsparks@dynamicsoft.com>
> > To: "Medhavi Bhatia" <mbhatia@nextone.com>
> > Cc: <sip@ietf.org>
> > Sent: Thursday, April 04, 2002 10:10 AM
> > Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
> >
> >
> > > On Thu, 2002-04-04 at 08:17, Medhavi Bhatia wrote:
> > > > The draft talks about the use of REFER outside a dialog:
> > > >
> > > > "A REFER request MAY be placed outside the scope of a dialog created
> > > > with an INVITE."
> > > >
> > > > "A UA accepting a well-formed REFER request SHOULD request approval
from
> > > > the user to proceed"
> > > >
> > > > I am not sure what kind of approval are we talking about here ?
> > > Whatever is appropriate for the device and application - it may
> > > be a direct prompt of the user, it may be a matter of using policy
> > > that the user has placed in the device.
> > > > If a REFER comes from a proxy to a SIP phone or a gateway (with
multiple
> > > > lines on it), what are we supposed to do ? Surely not ring the phone
?
> > > Why not? I've heard of phone implementations that provide a special
> > > alert and put a "X is requesting that you call Y, proceed?" type
prompt
> > > on the display. On a gateway, your interface is obviously more
limited.
> > > If you were ambitious, you could invoke media/ivr servers, alert the
> > > black phone, play the prompt above to it and ask for what to do.Or you
> > > could decide its not worth extending this kind of functionality to the
> > > old phones. There's a lot of room for innovation in between.
> > >
> > > RjS
> > >
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva cy 
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> not to add additional private information over and above that which
> the client has already seen fit to disclose.

And this restriction can only be relevant to a proxy that has a trust or
contractual relationship with the request origin. There is nothing,
morally, legally and technically, that a caller can and should do about
my inbound proxy adding whatever information my inbound proxy has legal
access to, even if that includes the google search result that reveals
that 'playboy' is the top site related to the caller's name.

I'd only wish that the privacy concerns expressed here could be
implanted into the louds using cell phones in public places.

> 
> The UA is reponsible for the actions of the UA. The proxy is only
> responsible for the actions of the proxy . . .
> 
> 
> --
> Dean

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Apr  4 20:57:30 2002
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Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Medhavi Bhatia <mbhatia@nextone.com>
Cc: sip@ietf.org
In-Reply-To: <018901c1dc34$c31a37d0$0100000a@deewana>
References: 
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On Thu, 2002-04-04 at 17:59, Medhavi Bhatia wrote:
> I am not sure we are talking about the same thing, so let me describe
> the scenario:
> 
> A and B are two endpoints not involved in any call. So A sends
> a REFER to B. (Both A and B support the REFER).
> 
> Now going back to your reply, I dont understand how you can
> alert B and play an ivr etc, without first setting up a dialog with it.
If you have a sip phone, you don't need to invoke an ivr - you don't
need a dialog. The implementations I describe provide a special alert
and a prompt.

To get to the IVR driving black phone example, you have to have
an application that fronts the pstn gateway - the pstn
gateway itself will likely reject out-of-band refers for lack
of a way to get user permission to do something reasonable. 
(likely only in a time-local sense - there are _vast_ opportunities
for innovation here). The server running that application will
terminate the REFER and enter into a series of INVITEs, establishing
the sessions you talk about, to build authorization, then possibly
entering into a 3pcc-like control mode with the gateway to deal with
the referred action.

> 
> Also, I am confused how the "Require:" header fits into this
> discussion.
You suggested (abstractly) to limit out-of-band REFERs because
of potential interop problems. I counter that the first application
of REFER does not have those interop problems and that if interop 
problems are encountered in future applications, they can be addressed
through judicious application of Require:.

RjS
> 
> ----- Original Message -----
> From: "Robert Sparks" <rsparks@dynamicsoft.com>
> To: "Medhavi Bhatia" <mbhatia@nextone.com>
> Cc: <sip@ietf.org>
> Sent: Thursday, April 04, 2002 11:08 AM
> Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
> 
> 
> > On Thu, 2002-04-04 at 09:54, Medhavi Bhatia wrote:
> > > I think by the IVR approach, you mean REFER in a
> > > dialog context.
> > No - I didn't.
> >
> > >
> > > That kind of an approach seems to be much more
> > > preferable, as it works for all endpoints unlike the one w/o
> > > a dialog which tries to do similar things. The caller obviously
> > > wont know that the destination is a smart phone or not.
> >
> > A caller doesn't know if the destination will accept an INVITE
> > either. That's what the response part of the SIP transaction is for.
> > If you send an out-of-band refer to something that by policy or
> > reason of physical limitations can't deal with it, it will decline.
> >
> > REFER is a primitive, not an application. We've been able to build
> > one application on top of REFER (transfer) without invoking Require:.
> > Some future application may need to know that the far end is going
> > to do something special with a REFER before it sends it - that
> > application will use a Require: token.
> >
> > >
> > > Ironically, w/o dialog seems to be syntactically more general.
> > > Maybe its use should be restricted somehow for interop
> > > reasons or so ?
> > I've seen no reason to introduce any such restriction.
> >
> > >
> > >
> > > ----- Original Message -----
> > > From: "Robert Sparks" <rsparks@dynamicsoft.com>
> > > To: "Medhavi Bhatia" <mbhatia@nextone.com>
> > > Cc: <sip@ietf.org>
> > > Sent: Thursday, April 04, 2002 10:10 AM
> > > Subject: Re: [Sip] comments on draft-ietf-sip-refer-02.txt
> > >
> > >
> > > > On Thu, 2002-04-04 at 08:17, Medhavi Bhatia wrote:
> > > > > The draft talks about the use of REFER outside a dialog:
> > > > >
> > > > > "A REFER request MAY be placed outside the scope of a dialog created
> > > > > with an INVITE."
> > > > >
> > > > > "A UA accepting a well-formed REFER request SHOULD request approval
> from
> > > > > the user to proceed"
> > > > >
> > > > > I am not sure what kind of approval are we talking about here ?
> > > > Whatever is appropriate for the device and application - it may
> > > > be a direct prompt of the user, it may be a matter of using policy
> > > > that the user has placed in the device.
> > > > > If a REFER comes from a proxy to a SIP phone or a gateway (with
> multiple
> > > > > lines on it), what are we supposed to do ? Surely not ring the phone
> ?
> > > > Why not? I've heard of phone implementations that provide a special
> > > > alert and put a "X is requesting that you call Y, proceed?" type
> prompt
> > > > on the display. On a gateway, your interface is obviously more
> limited.
> > > > If you were ambitious, you could invoke media/ivr servers, alert the
> > > > black phone, play the prompt above to it and ask for what to do.Or you
> > > > could decide its not worth extending this kind of functionality to the
> > > > old phones. There's a lot of room for innovation in between.
> > > >
> > > > RjS
> > > >
> > > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Robert Sparks" <rsparks@dynamicsoft.com>, <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Thu, 4 Apr 2002 19:55:02 -0800
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Sounds reasonable to me. I think it would be possible to phrase the draft in
a way that when we add in a Referred-By that it does not need a Requires and
that REFERs without a Referred-By are just treaded as the Referred-By as
unknown or anonymous.

Cullen

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Robert
> Sparks
> Sent: Thursday, April 04, 2002 8:42 AM
> To: sip@ietf.org
> Subject: [Sip] REFER security options - removing Referred-By
>
>
> I want to explore the option of removing Referred-By a little.
>
> As the refer-sec-options draft points out, traditional phone
> systems don't provide the kind of information Referred-By is
> trying to provide during a transfer. We've set our goals higher
> for SIP transfer, which is a good thing to do, but is it necessary
> for a base version of the application? Would a transfer application
> that didn't tell the transfer target who initiated the transfer be
> sufficient?
>
> Perhaps the Referred-By functionality should be an extension (Requires:)
> to REFER instead of part of its base definition?
>
> This would remove the security issue with base REFER (B's triggered
> request would be indistinguishable from a request that B initiated on
> its own - there is no claim in the request that the request exits
> because A asked for it to happen).
>
> The current work on securing Referred-By would continue in an extension
> draft, but advancement of REFER would not be blocked on its completion.
>
>
> RjS
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr  5 02:41:57 2002
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To: "Cullen Jennings" <fluffy@cisco.com>
Cc: "Robert Sparks" <rsparks@dynamicsoft.com>, sip@ietf.org,
        sip-admin@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Whether "traditional" phone systems provide the sort of information that 
is
carried by Referred-By or not, depends on your definition of "traditional
phone system" and the context in which REFER is being used. There are 
several examples of the availability of "Referred-by" type of information
to be found in today's telephone systems:

- ISDN's (Q.931) Redirecting Number Information Element
- QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer Initiate
  operation
- DPNSS' ....

Etc.

Leaving out all the "nice bits" may result in something that will do the
job (i.e. achieve the call transfer), but this also restricts new services
to be implemented, because there simply isn't enough information to infer
what's going on.

Regards,

Frank






Sounds reasonable to me. I think it would be possible to phrase the draft 
in
a way that when we add in a Referred-By that it does not need a Requires 
and
that REFERs without a Referred-By are just treaded as the Referred-By as
unknown or anonymous.

Cullen

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Robert
> Sparks
> Sent: Thursday, April 04, 2002 8:42 AM
> To: sip@ietf.org
> Subject: [Sip] REFER security options - removing Referred-By
>
>
> I want to explore the option of removing Referred-By a little.
>
> As the refer-sec-options draft points out, traditional phone
> systems don't provide the kind of information Referred-By is
> trying to provide during a transfer. We've set our goals higher
> for SIP transfer, which is a good thing to do, but is it necessary
> for a base version of the application? Would a transfer application
> that didn't tell the transfer target who initiated the transfer be
> sufficient?
>
> Perhaps the Referred-By functionality should be an extension (Requires:)
> to REFER instead of part of its base definition?
>
> This would remove the security issue with base REFER (B's triggered
> request would be indistinguishable from a request that B initiated on
> its own - there is no claim in the request that the request exits
> because A asked for it to happen).
>
> The current work on securing Referred-By would continue in an extension
> draft, but advancement of REFER would not be blocked on its completion.
>
>
> RjS
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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<br><font size=2 face="Courier">Whether &quot;traditional&quot; phone systems provide the sort of information that is</font>
<br><font size=2 face="Courier">carried by Referred-By or not, depends on your definition of &quot;traditional</font>
<br><font size=2 face="Courier">phone system&quot; and the context in which REFER is being used. There are </font>
<br><font size=2 face="Courier">several examples of the availability of &quot;Referred-by&quot; type of information</font>
<br><font size=2 face="Courier">to be found in today's telephone systems:</font>
<br>
<br><font size=2 face="Courier">- ISDN's (Q.931) Redirecting Number Information Element</font>
<br><font size=2 face="Courier">- QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer Initiate</font>
<br><font size=2 face="Courier">&nbsp; operation</font>
<br><font size=2 face="Courier">- DPNSS' ....</font>
<br>
<br><font size=2 face="Courier">Etc.</font>
<br>
<br><font size=2 face="Courier">Leaving out all the &quot;nice bits&quot; may result in something that will do the</font>
<br><font size=2 face="Courier">job (i.e. achieve the call transfer), but this also restricts new services</font>
<br><font size=2 face="Courier">to be implemented, because there simply isn't enough information to infer</font>
<br><font size=2 face="Courier">what's going on.</font>
<br>
<br><font size=2 face="Courier">Regards,</font>
<br>
<br><font size=2 face="Courier">Frank</font>
<br>
<br>
<br>
<br>
<br>
<br><font size=2 face="Courier"><br>
Sounds reasonable to me. I think it would be possible to phrase the draft in<br>
a way that when we add in a Referred-By that it does not need a Requires and<br>
that REFERs without a Referred-By are just treaded as the Referred-By as<br>
unknown or anonymous.<br>
<br>
Cullen<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Robert<br>
&gt; Sparks<br>
&gt; Sent: Thursday, April 04, 2002 8:42 AM<br>
&gt; To: sip@ietf.org<br>
&gt; Subject: [Sip] REFER security options - removing Referred-By<br>
&gt;<br>
&gt;<br>
&gt; I want to explore the option of removing Referred-By a little.<br>
&gt;<br>
&gt; As the refer-sec-options draft points out, traditional phone<br>
&gt; systems don't provide the kind of information Referred-By is<br>
&gt; trying to provide during a transfer. We've set our goals higher<br>
&gt; for SIP transfer, which is a good thing to do, but is it necessary<br>
&gt; for a base version of the application? Would a transfer application<br>
&gt; that didn't tell the transfer target who initiated the transfer be<br>
&gt; sufficient?<br>
&gt;<br>
&gt; Perhaps the Referred-By functionality should be an extension (Requires:)<br>
&gt; to REFER instead of part of its base definition?<br>
&gt;<br>
&gt; This would remove the security issue with base REFER (B's triggered<br>
&gt; request would be indistinguishable from a request that B initiated on<br>
&gt; its own - there is no claim in the request that the request exits<br>
&gt; because A asked for it to happen).<br>
&gt;<br>
&gt; The current work on securing Referred-By would continue in an extension<br>
&gt; draft, but advancement of REFER would not be blocked on its completion.<br>
&gt;<br>
&gt;<br>
&gt; RjS<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt;<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr  5 05:01:07 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Peterson, Jon'"
	 <jon.peterson@neustar.biz>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	 cy draft)
Date: Fri, 5 Apr 2002 10:43:31 +0100 
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> > RFC2543 clients - how can I ensure the subscribers privacy requirements
are met if the user 
> > has an RFC2543 client ? If I'm not worried about this case, then my
proxy can easily just 
> > modify the From field, since dialog matching is based only on the tags.

 

> You can't. You can NEVER prevent a client from giving away his identity.
For exampe, they might 
> add  an SMIME body conatining their name and a URL to their home-nudie-pix
site. At the very 
> best, the privacy draft documents a means by which a client requests the
server not to add
> additional private information over and above that which the client has
already seen fit to 
> disclose.
 
> The UA is reponsible for the actions of the UA. The proxy is only
responsible for the actions 
> of the proxy . . .

The implications of Data Protection in this environment are hard to fathom,
but I think it's only possible to understand them if you look at the
'service' that is being provided in a 'protocol-neutral' way. The Service
Provider is responsible for the operation of the service, full stop. - how
the implementation of the service is decomposed between devices is of not
interest to regulators etc.

If a Service Provider places certain requirements on a device, in order to
access the service, then the Service Provider is responsible in some sense
for the actions of the device in meeting those requirements - the device
could not do otherwise if it wants service. So you can't say that 'The UA is
responsible for the actions of the UA' when these are actions which are
*required* in order to access the service. In the majority case the Service
Provider provided the client anyway.

If part of the service is 'I'll carry any old piece of data you give me to
the called party' then obviously the responsibility for that data, and what
it contains, is entirely with the UA/user.

If part of the service is 'I'll carry the information you put in field X to
the called party, and by the way you should put your identity in field X',
then we are on much more dodgy ground, in particular when the identity of
the caller turns out to contain 'personal data' of someone else (e.g. the
subscriber).

In a service provider example, think of this from the point of view of the
subscriber. The subscriber has asked for all calls from this subscription to
be anonymous - they have a right to ask for this. I think the subscriber
would have a hard time if they tried to beat up the service provider because
one of the users specially configured their client to put their name in an
SMIME body - it is part of the service that bodies are carried transparently
(now if the subscriber wants a special 'body removal' service, then I'm sure
this could be arranged, at a price, though it sounds quite painful, not to
say terminal).

On the other hand, if the subscriber finds that all calls go through with
the name and SIP URL of the calling user in the From field, I think the
Service Provider would have a hard time arguing that they were taking
reasonable steps to ensure the subscribers wishes.

I'm not arguing that proxies need to be able to do this - I'm pointing out
that there are 'privacy services' which certain kinds of service provider
need to provide, which require More Than Just a Proxy, as Jon puts it. I
think it's important we understand this, as some people have the mistaken
impression that such 'privacy services' can be implemented on standard
proxies.

You seem to be arguing that privacy services in the network which modify
From/To are not required at all. This could only be the case if the
'From/To' information had the same status as, say, SMIME bodies in terms of
what the UA is required to put in there. OK, so the SIP spec does not state
a hard requirement on what should be put in there, but there is certainly a
clear implication that it should be the calling/called party identities and
indeed this *is what we want* to happen for the consistent operation of SIP
services.

We can obviously debate how responsible a Service Provider would be for
information inserted in the From/To field, when the Service Provider has not
made any hard requirements on the UA other than that they follow the SIP
spec. Service Providers in 3GPP are certainly of the view that the present
ambiguity in the SIP spec *does not* give them enough to be *sure* of
avoiding responsibility. They have two choices:
a) Accept the responsibility, and provide privacy services as I have
described
b) Decline the responsibility by explicitly requiring that the UA *does not*
put potentially private information in these fields

There is no option of 'Decline responsibility, but refer to the SIP spec for
what the UA does with these fields' - it is just not safe enough, given that
the clear implication and intention is that the UA puts identity information
into From/To. I *know now* that most clients will put personal data of the
subscriber in the From field. In 3 years time, can I really argue to a
regulator with integrity that I took reasonable steps to protect the
subscribers privacy if I do not modify this field ??

Other users of SIP will face the same dilemma - which option would you
prefer them to choose ?

...Mark



 
> --
> Dean

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP =
Priva cy draft)</TITLE>
</HEAD>
<BODY>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; &gt; RFC2543 clients - how can I ensure the =
subscribers privacy requirements are met if the user </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; has an RFC2543 client ? If I'm not worried =
about this case, then my proxy can easily just </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; modify the From field, since dialog =
matching is based only on the tags.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
</P>

<P><FONT SIZE=3D2>&gt; You can't. You can NEVER prevent a client from =
giving away his identity. For exampe, they might </FONT>
<BR><FONT SIZE=3D2>&gt; add&nbsp; an SMIME body conatining their name =
and a URL to their home-nudie-pix site. At the very </FONT>
<BR><FONT SIZE=3D2>&gt; best, the privacy draft documents a means by =
which a client requests the server not to add</FONT>
<BR><FONT SIZE=3D2>&gt; additional private information over and above =
that which the client has already seen fit to </FONT>
<BR><FONT SIZE=3D2>&gt; disclose.</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; The UA is reponsible for the actions of the UA. =
The proxy is only responsible for the actions </FONT>
<BR><FONT SIZE=3D2>&gt; of the proxy . . .</FONT>
</P>

<P><FONT SIZE=3D2>The implications of Data Protection in this =
environment are hard to fathom, but I think it's only possible to =
understand them if you look at the 'service' that is being provided in =
a 'protocol-neutral' way. The Service Provider is responsible for the =
operation of the service, full stop. - how the implementation of the =
service is decomposed between devices is of not interest to regulators =
etc.</FONT></P>

<P><FONT SIZE=3D2>If a Service Provider places certain requirements on =
a device, in order to access the service, then the Service Provider is =
responsible in some sense for the actions of the device in meeting =
those requirements - the device could not do otherwise if it wants =
service. So you can't say that 'The UA is responsible for the actions =
of the UA' when these are actions which are *required* in order to =
access the service. In the majority case the Service Provider provided =
the client anyway.</FONT></P>

<P><FONT SIZE=3D2>If part of the service is 'I'll carry any old piece =
of data you give me to the called party' then obviously the =
responsibility for that data, and what it contains, is entirely with =
the UA/user.</FONT></P>

<P><FONT SIZE=3D2>If part of the service is 'I'll carry the information =
you put in field X to the called party, and by the way you should put =
your identity in field X', then we are on much more dodgy ground, in =
particular when the identity of the caller turns out to contain =
'personal data' of someone else (e.g. the subscriber).</FONT></P>

<P><FONT SIZE=3D2>In a service provider example, think of this from the =
point of view of the subscriber. The subscriber has asked for all calls =
from this subscription to be anonymous - they have a right to ask for =
this. I think the subscriber would have a hard time if they tried to =
beat up the service provider because one of the users specially =
configured their client to put their name in an SMIME body - it is part =
of the service that bodies are carried transparently (now if the =
subscriber wants a special 'body removal' service, then I'm sure this =
could be arranged, at a price, though it sounds quite painful, not to =
say terminal).</FONT></P>

<P><FONT SIZE=3D2>On the other hand, if the subscriber finds that all =
calls go through with the name and SIP URL of the calling user in the =
From field, I think the Service Provider would have a hard time arguing =
that they were taking reasonable steps to ensure the subscribers =
wishes.</FONT></P>

<P><FONT SIZE=3D2>I'm not arguing that proxies need to be able to do =
this - I'm pointing out that there are 'privacy services' which certain =
kinds of service provider need to provide, which require More Than Just =
a Proxy, as Jon puts it. I think it's important we understand this, as =
some people have the mistaken impression that such 'privacy services' =
can be implemented on standard proxies.</FONT></P>

<P><FONT SIZE=3D2>You seem to be arguing that privacy services in the =
network which modify From/To are not required at all. This could only =
be the case if the 'From/To' information had the same status as, say, =
SMIME bodies in terms of what the UA is required to put in there. OK, =
so the SIP spec does not state a hard requirement on what should be put =
in there, but there is certainly a clear implication that it should be =
the calling/called party identities and indeed this *is what we want* =
to happen for the consistent operation of SIP services.</FONT></P>

<P><FONT SIZE=3D2>We can obviously debate how responsible a Service =
Provider would be for information inserted in the From/To field, when =
the Service Provider has not made any hard requirements on the UA other =
than that they follow the SIP spec. Service Providers in 3GPP are =
certainly of the view that the present ambiguity in the SIP spec *does =
not* give them enough to be *sure* of avoiding responsibility. They =
have two choices:</FONT></P>

<P><FONT SIZE=3D2>a) Accept the responsibility, and provide privacy =
services as I have described</FONT>
<BR><FONT SIZE=3D2>b) Decline the responsibility by explicitly =
requiring that the UA *does not* put potentially private information in =
these fields</FONT></P>

<P><FONT SIZE=3D2>There is no option of 'Decline responsibility, but =
refer to the SIP spec for what the UA does with these fields' - it is =
just not safe enough, given that the clear implication and intention is =
that the UA puts identity information into From/To. I *know now* that =
most clients will put personal data of the subscriber in the From =
field. In 3 years time, can I really argue to a regulator with =
integrity that I took reasonable steps to protect the subscribers =
privacy if I do not modify this field ??</FONT></P>

<P><FONT SIZE=3D2>Other users of SIP will face the same dilemma - which =
option would you prefer them to choose ?</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; Dean</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C1DC86.58743064--

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From daemon@optimus.ietf.org  Fri Apr  5 07:18:27 2002
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Date: Fri, 5 Apr 2002 06:59:39 -0500 (EST)
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To: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy draft)
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Mark Watson wrote:
> Dean Willis wrote:
> > You can't. You can NEVER prevent a client from giving away his identity. 
> > For exampe, they might  add  an SMIME body conatining their name and a 
> > URL to their home-nudie-pix site. At the very best, the privacy draft 
> > documents a means by which a client requests the server not to add
> > additional private information over and above that which the client has
> > already seen fit to disclose.
> >  
> > The UA is reponsible for the actions of the UA. The proxy is only
> > responsible for the actions of the proxy . . .
> 
> The implications of Data Protection in this environment are hard to fathom,
> but I think it's only possible to understand them if you look at the
> 'service' that is being provided in a 'protocol-neutral' way. The Service
> Provider is responsible for the operation of the service, full stop. - how
> the implementation of the service is decomposed between devices is of not
> interest to regulators etc.

The 'protocol-neutral' approach has certainly been the approach in the past,
and a reasonable bet for this case too.  Consider the service changes that
were required in the transition from microwave to fiber - none.

Different jurisdictions have different views on data protection, too.  For
example, CLIR is the default in California, and in other states it is not.

The interesting case, of course, is handling the jurisdictions (like 
California) where the Service Provider is expected to do something, whether
the subscriber cooperates in asking for it or not.  There are two types
of cars in the US, California-emissions and everywhere-else; I shudder
to think of the same approach to SIP User Agents.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	 cy draft)
Date: Fri, 5 Apr 2002 10:43:31 +0100 

> > RFC2543 clients - how can I ensure the subscribers privacy requirements
are met if the user 
> > has an RFC2543 client ? If I'm not worried about this case, then my
proxy can easily just 
> > modify the From field, since dialog matching is based only on the tags.

 

> You can't. You can NEVER prevent a client from giving away his identity.
For exampe, they might 
> add  an SMIME body conatining their name and a URL to their home-nudie-pix
site. At the very 
> best, the privacy draft documents a means by which a client requests the
server not to add
> additional private information over and above that which the client has
already seen fit to 
> disclose.
 
> The UA is reponsible for the actions of the UA. The proxy is only
responsible for the actions 
> of the proxy . . .

The implications of Data Protection in this environment are hard to fathom,
but I think it's only possible to understand them if you look at the
'service' that is being provided in a 'protocol-neutral' way. The Service
Provider is responsible for the operation of the service, full stop. - how
the implementation of the service is decomposed between devices is of not
interest to regulators etc.

If a Service Provider places certain requirements on a device, in order to
access the service, then the Service Provider is responsible in some sense
for the actions of the device in meeting those requirements - the device
could not do otherwise if it wants service. So you can't say that 'The UA is
responsible for the actions of the UA' when these are actions which are
*required* in order to access the service. In the majority case the Service
Provider provided the client anyway.

If part of the service is 'I'll carry any old piece of data you give me to
the called party' then obviously the responsibility for that data, and what
it contains, is entirely with the UA/user.

If part of the service is 'I'll carry the information you put in field X to
the called party, and by the way you should put your identity in field X',
then we are on much more dodgy ground, in particular when the identity of
the caller turns out to contain 'personal data' of someone else (e.g. the
subscriber).

In a service provider example, think of this from the point of view of the
subscriber. The subscriber has asked for all calls from this subscription to
be anonymous - they have a right to ask for this. I think the subscriber
would have a hard time if they tried to beat up the service provider because
one of the users specially configured their client to put their name in an
SMIME body - it is part of the service that bodies are carried transparently
(now if the subscriber wants a special 'body removal' service, then I'm sure
this could be arranged, at a price, though it sounds quite painful, not to
say terminal).

On the other hand, if the subscriber finds that all calls go through with
the name and SIP URL of the calling user in the From field, I think the
Service Provider would have a hard time arguing that they were taking
reasonable steps to ensure the subscribers wishes.

I'm not arguing that proxies need to be able to do this - I'm pointing out
that there are 'privacy services' which certain kinds of service provider
need to provide, which require More Than Just a Proxy, as Jon puts it. I
think it's important we understand this, as some people have the mistaken
impression that such 'privacy services' can be implemented on standard
proxies.

You seem to be arguing that privacy services in the network which modify
From/To are not required at all. This could only be the case if the
'From/To' information had the same status as, say, SMIME bodies in terms of
what the UA is required to put in there. OK, so the SIP spec does not state
a hard requirement on what should be put in there, but there is certainly a
clear implication that it should be the calling/called party identities and
indeed this *is what we want* to happen for the consistent operation of SIP
services.

We can obviously debate how responsible a Service Provider would be for
information inserted in the From/To field, when the Service Provider has not
made any hard requirements on the UA other than that they follow the SIP
spec. Service Providers in 3GPP are certainly of the view that the present
ambiguity in the SIP spec *does not* give them enough to be *sure* of
avoiding responsibility. They have two choices:
a) Accept the responsibility, and provide privacy services as I have
described
b) Decline the responsibility by explicitly requiring that the UA *does not*
put potentially private information in these fields

There is no option of 'Decline responsibility, but refer to the SIP spec for
what the UA does with these fields' - it is just not safe enough, given that
the clear implication and intention is that the UA puts identity information
into From/To. I *know now* that most clients will put personal data of the
subscriber in the From field. In 3 years time, can I really argue to a
regulator with integrity that I took reasonable steps to protect the
subscribers privacy if I do not modify this field ??

Other users of SIP will face the same dilemma - which option would you
prefer them to choose ?

...Mark



 
> --
> Dean

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr  5 07:38:49 2002
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From: "Mart Nurmet" <mnurmet@ev1.net>
To: "William Marshall" <wtm@research.att.com>, <sip@ietf.org>
References: <200204051159.GAA19159@fish.research.att.com>
Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy draft)
Date: Fri, 5 Apr 2002 06:27:11 -0800
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And don't forget the various national enforcement agency requirments.  Some
of which are not negotiable.


----- Original Message -----
From: "William Marshall" <wtm@research.att.com>
To: <sip@ietf.org>
Sent: Friday, April 05, 2002 3:59 AM
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy
draft)


> Mark Watson wrote:
> > Dean Willis wrote:
> > > You can't. You can NEVER prevent a client from giving away his
identity.
> > > For exampe, they might  add  an SMIME body conatining their name and a
> > > URL to their home-nudie-pix site. At the very best, the privacy draft
> > > documents a means by which a client requests the server not to add
> > > additional private information over and above that which the client
has
> > > already seen fit to disclose.
> > >
> > > The UA is reponsible for the actions of the UA. The proxy is only
> > > responsible for the actions of the proxy . . .
> >
> > The implications of Data Protection in this environment are hard to
fathom,
> > but I think it's only possible to understand them if you look at the
> > 'service' that is being provided in a 'protocol-neutral' way. The
Service
> > Provider is responsible for the operation of the service, full stop. -
how
> > the implementation of the service is decomposed between devices is of
not
> > interest to regulators etc.
>
> The 'protocol-neutral' approach has certainly been the approach in the
past,
> and a reasonable bet for this case too.  Consider the service changes that
> were required in the transition from microwave to fiber - none.
>
> Different jurisdictions have different views on data protection, too.  For
> example, CLIR is the default in California, and in other states it is not.
>
> The interesting case, of course, is handling the jurisdictions (like
> California) where the Service Provider is expected to do something,
whether
> the subscriber cooperates in asking for it or not.  There are two types
> of cars in the US, California-emissions and everywhere-else; I shudder
> to think of the same approach to SIP User Agents.
>
> Bill Marshall
> wtm@research.att.com
>
> -----original message-----
> From: "Mark Watson" <mwatson@nortelnetworks.com>
> To: "'Dean Willis'" <dean.willis@softarmor.com>,
>         "'Peterson, Jon'" <jon.peterson@neustar.biz>
> Cc: sip@ietf.org
> Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
> cy draft)
> Date: Fri, 5 Apr 2002 10:43:31 +0100
>
> > > RFC2543 clients - how can I ensure the subscribers privacy
requirements
> are met if the user
> > > has an RFC2543 client ? If I'm not worried about this case, then my
> proxy can easily just
> > > modify the From field, since dialog matching is based only on the
tags.
>
>
>
> > You can't. You can NEVER prevent a client from giving away his identity.
> For exampe, they might
> > add  an SMIME body conatining their name and a URL to their
home-nudie-pix
> site. At the very
> > best, the privacy draft documents a means by which a client requests the
> server not to add
> > additional private information over and above that which the client has
> already seen fit to
> > disclose.
>
> > The UA is reponsible for the actions of the UA. The proxy is only
> responsible for the actions
> > of the proxy . . .
>
> The implications of Data Protection in this environment are hard to
fathom,
> but I think it's only possible to understand them if you look at the
> 'service' that is being provided in a 'protocol-neutral' way. The Service
> Provider is responsible for the operation of the service, full stop. - how
> the implementation of the service is decomposed between devices is of not
> interest to regulators etc.
>
> If a Service Provider places certain requirements on a device, in order to
> access the service, then the Service Provider is responsible in some sense
> for the actions of the device in meeting those requirements - the device
> could not do otherwise if it wants service. So you can't say that 'The UA
is
> responsible for the actions of the UA' when these are actions which are
> *required* in order to access the service. In the majority case the
Service
> Provider provided the client anyway.
>
> If part of the service is 'I'll carry any old piece of data you give me to
> the called party' then obviously the responsibility for that data, and
what
> it contains, is entirely with the UA/user.
>
> If part of the service is 'I'll carry the information you put in field X
to
> the called party, and by the way you should put your identity in field X',
> then we are on much more dodgy ground, in particular when the identity of
> the caller turns out to contain 'personal data' of someone else (e.g. the
> subscriber).
>
> In a service provider example, think of this from the point of view of the
> subscriber. The subscriber has asked for all calls from this subscription
to
> be anonymous - they have a right to ask for this. I think the subscriber
> would have a hard time if they tried to beat up the service provider
because
> one of the users specially configured their client to put their name in an
> SMIME body - it is part of the service that bodies are carried
transparently
> (now if the subscriber wants a special 'body removal' service, then I'm
sure
> this could be arranged, at a price, though it sounds quite painful, not to
> say terminal).
>
> On the other hand, if the subscriber finds that all calls go through with
> the name and SIP URL of the calling user in the From field, I think the
> Service Provider would have a hard time arguing that they were taking
> reasonable steps to ensure the subscribers wishes.
>
> I'm not arguing that proxies need to be able to do this - I'm pointing out
> that there are 'privacy services' which certain kinds of service provider
> need to provide, which require More Than Just a Proxy, as Jon puts it. I
> think it's important we understand this, as some people have the mistaken
> impression that such 'privacy services' can be implemented on standard
> proxies.
>
> You seem to be arguing that privacy services in the network which modify
> From/To are not required at all. This could only be the case if the
> 'From/To' information had the same status as, say, SMIME bodies in terms
of
> what the UA is required to put in there. OK, so the SIP spec does not
state
> a hard requirement on what should be put in there, but there is certainly
a
> clear implication that it should be the calling/called party identities
and
> indeed this *is what we want* to happen for the consistent operation of
SIP
> services.
>
> We can obviously debate how responsible a Service Provider would be for
> information inserted in the From/To field, when the Service Provider has
not
> made any hard requirements on the UA other than that they follow the SIP
> spec. Service Providers in 3GPP are certainly of the view that the present
> ambiguity in the SIP spec *does not* give them enough to be *sure* of
> avoiding responsibility. They have two choices:
> a) Accept the responsibility, and provide privacy services as I have
> described
> b) Decline the responsibility by explicitly requiring that the UA *does
not*
> put potentially private information in these fields
>
> There is no option of 'Decline responsibility, but refer to the SIP spec
for
> what the UA does with these fields' - it is just not safe enough, given
that
> the clear implication and intention is that the UA puts identity
information
> into From/To. I *know now* that most clients will put personal data of the
> subscriber in the From field. In 3 years time, can I really argue to a
> regulator with integrity that I took reasonable steps to protect the
> subscribers privacy if I do not modify this field ??
>
> Other users of SIP will face the same dilemma - which option would you
> prefer them to choose ?
>
> ...Mark
>
>
>
>
> > --
> > Dean
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Mark Watson'"
	 <mwatson@nortelnetworks.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy draft)
Date: Fri, 5 Apr 2002 14:51:59 +0100 
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-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: 02 April 2002 15:55
To: 'Mark Watson'
See below

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

Cc: sip@ietf.org
Subject: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy
draft)



Mark wrote:
-----
The root problem is this: A UA cannot place 'private' or 'potentially
private' information in the From or To fields.  

This is because there is no mechanism defined to ensure that such
information is not passed to other UAs. In the absence of such a
mechanism, no requirements can be placed on what a UA does with the
From/To fields. If we do not place any requirements on these fields,
then we cannot reliably use them for anything.
------

I think this is statement of the situation clearly brings out the
misunderstanding driving this whole thing.

This logic will eventually lead us to deleting all headers or bodies
provided by the UA in at attempt to enforce "privacy" against the wishes
of the actual user. It also precludes intentially authenticating the
user to a service on the other side of the home proxy, as such
authentication by definition is an exposure of private information.

I would restate as follows:

"The SIP protocol explicitly sends the From and To fields to the
destination UA. Therefore the sending UA should only insert information
in the From and To fields that is intended to be delivered to other UAs
and potentially displayed to the user of the other UA. If the user of
the UA wishes to remain anonymous, they must not insert any information
into the From or To fields which compromises this anonymity. This same
caveat applies to every other aspect of SIP, including headers and
bodies, except for those elements which are explicitly removed or
encrypted by intermediate elements known to the UA."


[KED] I think the issues here are that the SIP specification very clearly
indicates that an identity is inserted in the From field (   The From header
field indicates the logical identity of the initiator of the request,
possibly the user's address-of-record. Like the To header field, it contains
a URI and optionally a display name. ) and therefore the protocol denies the
user privacy in this respect. Secondly, if the user has the ability to
insert information in this area, then this is not a location that can be
used for inserting a network asserted identity. I do not therefore see the
From header as a place for putting this network (third party) asserted
calling user identity.

--
Dean


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Subject: RE: [Sip] REFER security options - removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: frank.derks@philips.com
Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org, sip-admin@ietf.org
In-Reply-To: <OF9A02BEC6.C8C895D1-ONC1256B92.00275601@diamond.philips.com>
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On Fri, 2002-04-05 at 01:21, frank.derks@philips.com wrote:
> 
> Whether "traditional" phone systems provide the sort of information that
> is 
> carried by Referred-By or not, depends on your definition of
> "traditional 
> phone system" and the context in which REFER is being used. There are 
> several examples of the availability of "Referred-by" type of
> information 
> to be found in today's telephone systems: 
> 
> - ISDN's (Q.931) Redirecting Number Information Element 
> - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> Initiate 
>   operation 
> - DPNSS' .... 
> 
> Etc. 
> 
> Leaving out all the "nice bits" may result in something that will do the
> 
> job (i.e. achieve the call transfer), but this also restricts new
> services 
> to be implemented, because there simply isn't enough information to
> infer 
> what's going on. 

To be clear, I'm not suggesting we abandon the Referred-By idea. I'm
suggesting we start with a version of REFER that doesn't provide it
and bring it back in with a follow-on extension to REFER.

> 
> Regards, 
> 
> Frank 
> 
> 
> 
> 
> 
> 
> Sounds reasonable to me. I think it would be possible to phrase the
> draft in
> a way that when we add in a Referred-By that it does not need a Requires
> and
> that REFERs without a Referred-By are just treaded as the Referred-By as
> unknown or anonymous.
> 
> Cullen
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Robert
> > Sparks
> > Sent: Thursday, April 04, 2002 8:42 AM
> > To: sip@ietf.org
> > Subject: [Sip] REFER security options - removing Referred-By
> >
> >
> > I want to explore the option of removing Referred-By a little.
> >
> > As the refer-sec-options draft points out, traditional phone
> > systems don't provide the kind of information Referred-By is
> > trying to provide during a transfer. We've set our goals higher
> > for SIP transfer, which is a good thing to do, but is it necessary
> > for a base version of the application? Would a transfer application
> > that didn't tell the transfer target who initiated the transfer be
> > sufficient?
> >
> > Perhaps the Referred-By functionality should be an extension
> (Requires:)
> > to REFER instead of part of its base definition?
> >
> > This would remove the security issue with base REFER (B's triggered
> > request would be indistinguishable from a request that B initiated on
> > its own - there is no claim in the request that the request exits
> > because A asked for it to happen).
> >
> > The current work on securing Referred-By would continue in an
> extension
> > draft, but advancement of REFER would not be blocked on its
> completion.
> >
> >
> > RjS
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 



_______________________________________________
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From daemon@optimus.ietf.org  Fri Apr  5 10:34:08 2002
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Date: Fri, 05 Apr 2002 10:07:05 -0500
From: Nigel Dewdney <njdewdn@lts.ncsc.mil>
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To: sip@ietf.org
Subject: Re: [Sip] REFER security options - removing Referred-By
References: <OF9A02BEC6.C8C895D1-ONC1256B92.00275601@diamond.philips.com>
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Pardon me if I'm being obtuse, but how do I tell whether an INVITE is as
a result
of a REFER or not if Referred-By is dropped? A calls B. B REFERs A to C,
A calls
C. If C can not tell that the call is a result of a referral from A then
the REFER mechanism
has been reduced into one of "hey don't call me call them!", i.e. the
call from A to C is
disassociated from the dialog between A and B.

Dropping Referred-By could preclude many applications. Okay, an
extension to bring
it back in would enable them, but wasn't the idea of REFER that it be a
primitive to
enable such applications? The obvious example is where C wants to only
accept calls
which have been vetted by B. Irrespective of  security requirements,
this requires
that the primitive have a reference to B in A's request to C. How about
applications
where I would want to treat INVITEs differently according to where the
REFER came
from? E.g. accept calls referred from P.A. but fwd those reffered from
Marketing to my
voice-mail service.

On the question of authenticating the referral I prefer the option of C
VERIFYing with
A. Why wouldn't A put a meaningful contact address in Referred-By: and
expect C
to blindly accept the referral without the option of being able to
contact A? Thats not
to say that I don't thing the S/MIME option isn't possible - I'm just
uneasy about
man-in-the-middle.

Regards,

Nigel Dewdney
Laboratory Telecommunication Sciences


frank.derks@philips.com wrote:

>
> Whether "traditional" phone systems provide the sort of information
> that is
> carried by Referred-By or not, depends on your definition of
> "traditional
> phone system" and the context in which REFER is being used. There are
> several examples of the availability of "Referred-by" type of
> information
> to be found in today's telephone systems:
>
> - ISDN's (Q.931) Redirecting Number Information Element
> - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> Initiate
>   operation
> - DPNSS' ....
>
> Etc.
>
> Leaving out all the "nice bits" may result in something that will do
> the
> job (i.e. achieve the call transfer), but this also restricts new
> services
> to be implemented, because there simply isn't enough information to
> infer
> what's going on.
>
> Regards,
>
> Frank
>
>
>
>
>
>
> Sounds reasonable to me. I think it would be possible to phrase the
> draft in
> a way that when we add in a Referred-By that it does not need a
> Requires and
> that REFERs without a Referred-By are just treaded as the Referred-By
> as
> unknown or anonymous.
>
> Cullen
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Robert
> > Sparks
> > Sent: Thursday, April 04, 2002 8:42 AM
> > To: sip@ietf.org
> > Subject: [Sip] REFER security options - removing Referred-By
> >
> >
> > I want to explore the option of removing Referred-By a little.
> >
> > As the refer-sec-options draft points out, traditional phone
> > systems don't provide the kind of information Referred-By is
> > trying to provide during a transfer. We've set our goals higher
> > for SIP transfer, which is a good thing to do, but is it necessary
> > for a base version of the application? Would a transfer application
> > that didn't tell the transfer target who initiated the transfer be
> > sufficient?
> >
> > Perhaps the Referred-By functionality should be an extension
> (Requires:)
> > to REFER instead of part of its base definition?
> >
> > This would remove the security issue with base REFER (B's triggered
> > request would be indistinguishable from a request that B initiated
> on
> > its own - there is no claim in the request that the request exits
> > because A asked for it to happen).
> >
> > The current work on securing Referred-By would continue in an
> extension
> > draft, but advancement of REFER would not be blocked on its
> completion.
> >
> >
> > RjS
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr  5 11:00:02 2002
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Subject: Re: [Sip] REFER security options - removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Nigel Dewdney <njdewdn@lts.ncsc.mil>
Cc: sip@ietf.org
In-Reply-To: <3CADBD99.89793751@lts.ncsc.mil>
References: <OF9A02BEC6.C8C895D1-ONC1256B92.00275601@diamond.philips.com> 
	<3CADBD99.89793751@lts.ncsc.mil>
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On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote:
> Pardon me if I'm being obtuse, but how do I tell whether an INVITE is as
> a result
> of a REFER or not if Referred-By is dropped?

That's exactly the point - you wouldn't.

> A calls B. B REFERs A to C,
> A calls
> C. If C can not tell that the call is a result of a referral from A then
> the REFER mechanism
> has been reduced into one of "hey don't call me call them!", i.e. the
> call from A to C is
> disassociated from the dialog between A and B.

No - that's a separate problem. The kind of association you are
looking for here is what's being addressed through the Replaces header.

> 
> Dropping Referred-By could preclude many applications. Okay, an
> extension to bring
> it back in would enable them, but wasn't the idea of REFER that it be a
> primitive to
> enable such applications? 
> The obvious example is where C wants to only
> accept calls
> which have been vetted by B. Irrespective of  security requirements,
> this requires
> that the primitive have a reference to B in A's request to C. How about
You got your alphabet mixed up there.
> applications
> where I would want to treat INVITEs differently according to where the
> REFER came
> from? E.g. accept calls referred from P.A. but fwd those reffered from
> Marketing to my
> voice-mail service.

There are _clearly_ good applications waiting for the Referred-By
functionality. I am _NOT_ advocating we abandon providing it. What
I'm asking is whether or not the base level transfer application
really needs it. There are already fielded implementations that don't
use it providing some evidence that you have a useful service without
it.

> 
> On the question of authenticating the referral I prefer the option of C
> VERIFYing with
> A. Why wouldn't A put a meaningful contact address in Referred-By: and
> expect C
> to blindly accept the referral without the option of being able to
> contact A? Thats not
> to say that I don't thing the S/MIME option isn't possible - I'm just
> uneasy about
> man-in-the-middle.

I don't follow - you're suggesting that if C tried to contact A and
failed, it would just accept the request by default?

The real problem here is that it may be very difficult for A to
provide a contact to C that will  get back to the UA that has the
knowledge about sending the REFER. Its the same problem
we have with getting B's INVITE-Replaces to the same C we had
a consultative-hold conference with.
> 
> Regards,
> 
> Nigel Dewdney
> Laboratory Telecommunication Sciences
> 
> 
> frank.derks@philips.com wrote:
> 
> >
> > Whether "traditional" phone systems provide the sort of information
> > that is
> > carried by Referred-By or not, depends on your definition of
> > "traditional
> > phone system" and the context in which REFER is being used. There are
> > several examples of the availability of "Referred-by" type of
> > information
> > to be found in today's telephone systems:
> >
> > - ISDN's (Q.931) Redirecting Number Information Element
> > - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> > Initiate
> >   operation
> > - DPNSS' ....
> >
> > Etc.
> >
> > Leaving out all the "nice bits" may result in something that will do
> > the
> > job (i.e. achieve the call transfer), but this also restricts new
> > services
> > to be implemented, because there simply isn't enough information to
> > infer
> > what's going on.
> >
> > Regards,
> >
> > Frank
> >
> >
> >
> >
> >
> >
> > Sounds reasonable to me. I think it would be possible to phrase the
> > draft in
> > a way that when we add in a Referred-By that it does not need a
> > Requires and
> > that REFERs without a Referred-By are just treaded as the Referred-By
> > as
> > unknown or anonymous.
> >
> > Cullen
> >
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Robert
> > > Sparks
> > > Sent: Thursday, April 04, 2002 8:42 AM
> > > To: sip@ietf.org
> > > Subject: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > I want to explore the option of removing Referred-By a little.
> > >
> > > As the refer-sec-options draft points out, traditional phone
> > > systems don't provide the kind of information Referred-By is
> > > trying to provide during a transfer. We've set our goals higher
> > > for SIP transfer, which is a good thing to do, but is it necessary
> > > for a base version of the application? Would a transfer application
> > > that didn't tell the transfer target who initiated the transfer be
> > > sufficient?
> > >
> > > Perhaps the Referred-By functionality should be an extension
> > (Requires:)
> > > to REFER instead of part of its base definition?
> > >
> > > This would remove the security issue with base REFER (B's triggered
> > > request would be indistinguishable from a request that B initiated
> > on
> > > its own - there is no claim in the request that the request exits
> > > because A asked for it to happen).
> > >
> > > The current work on securing Referred-By would continue in an
> > extension
> > > draft, but advancement of REFER would not be blocked on its
> > completion.
> > >
> > >
> > > RjS
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Sriram Parameswar"<sriramp@nortelnetworks.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>,
        "'Nigel Dewdney'"
	 <njdewdn@lts.ncsc.mil>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Fri, 5 Apr 2002 10:40:44 -0600 
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Hi All,

To quickly summarize - the Referred-By is undoubtedly a good thing to put
into a request resulting out of a REFER. However to quote the REFER draft 
<<quote>>
If the URL is a SIP URL, the Referred-By header in the REFER request should
be copied into the request sent to the referred-to resource.
<<endquote>>

It seems to me that this of "SHOULD" strength, and anybody not
willing/wanting to send it - need not. I am of the opinion that we use the
appropriate mechanisms (either the SMIME/ or direct transferor to transfer
target verification method) to ensure that this Referred-By is verifiable.
Also if the transferor is wanting to protect itself it could always put in a
URI of the nature:
	sip:anonymous@anonymous.net

and allow the transfer target the choice to accept or deny the call.

Removal and converting it into Requires does not buy us much. I vote that we
continue down the path as suggested in drafts-sparks-sip-sec-options-00 and
leave the Referred-By header as is.

Thanks,

Sriram

-----Original Message-----
From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
Sent: Friday, April 05, 2002 9:41 AM
To: Nigel Dewdney
Cc: sip@ietf.org
Subject: Re: [Sip] REFER security options - removing Referred-By


On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote:
> Pardon me if I'm being obtuse, but how do I tell whether an INVITE is as
> a result
> of a REFER or not if Referred-By is dropped?

That's exactly the point - you wouldn't.

> A calls B. B REFERs A to C,
> A calls
> C. If C can not tell that the call is a result of a referral from A then
> the REFER mechanism
> has been reduced into one of "hey don't call me call them!", i.e. the
> call from A to C is
> disassociated from the dialog between A and B.

No - that's a separate problem. The kind of association you are
looking for here is what's being addressed through the Replaces header.

> 
> Dropping Referred-By could preclude many applications. Okay, an
> extension to bring
> it back in would enable them, but wasn't the idea of REFER that it be a
> primitive to
> enable such applications? 
> The obvious example is where C wants to only
> accept calls
> which have been vetted by B. Irrespective of  security requirements,
> this requires
> that the primitive have a reference to B in A's request to C. How about
You got your alphabet mixed up there.
> applications
> where I would want to treat INVITEs differently according to where the
> REFER came
> from? E.g. accept calls referred from P.A. but fwd those reffered from
> Marketing to my
> voice-mail service.

There are _clearly_ good applications waiting for the Referred-By
functionality. I am _NOT_ advocating we abandon providing it. What
I'm asking is whether or not the base level transfer application
really needs it. There are already fielded implementations that don't
use it providing some evidence that you have a useful service without
it.

> 
> On the question of authenticating the referral I prefer the option of C
> VERIFYing with
> A. Why wouldn't A put a meaningful contact address in Referred-By: and
> expect C
> to blindly accept the referral without the option of being able to
> contact A? Thats not
> to say that I don't thing the S/MIME option isn't possible - I'm just
> uneasy about
> man-in-the-middle.

I don't follow - you're suggesting that if C tried to contact A and
failed, it would just accept the request by default?

The real problem here is that it may be very difficult for A to
provide a contact to C that will  get back to the UA that has the
knowledge about sending the REFER. Its the same problem
we have with getting B's INVITE-Replaces to the same C we had
a consultative-hold conference with.
> 
> Regards,
> 
> Nigel Dewdney
> Laboratory Telecommunication Sciences
> 
> 
> frank.derks@philips.com wrote:
> 
> >
> > Whether "traditional" phone systems provide the sort of information
> > that is
> > carried by Referred-By or not, depends on your definition of
> > "traditional
> > phone system" and the context in which REFER is being used. There are
> > several examples of the availability of "Referred-by" type of
> > information
> > to be found in today's telephone systems:
> >
> > - ISDN's (Q.931) Redirecting Number Information Element
> > - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> > Initiate
> >   operation
> > - DPNSS' ....
> >
> > Etc.
> >
> > Leaving out all the "nice bits" may result in something that will do
> > the
> > job (i.e. achieve the call transfer), but this also restricts new
> > services
> > to be implemented, because there simply isn't enough information to
> > infer
> > what's going on.
> >
> > Regards,
> >
> > Frank
> >
> >
> >
> >
> >
> >
> > Sounds reasonable to me. I think it would be possible to phrase the
> > draft in
> > a way that when we add in a Referred-By that it does not need a
> > Requires and
> > that REFERs without a Referred-By are just treaded as the Referred-By
> > as
> > unknown or anonymous.
> >
> > Cullen
> >
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Robert
> > > Sparks
> > > Sent: Thursday, April 04, 2002 8:42 AM
> > > To: sip@ietf.org
> > > Subject: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > I want to explore the option of removing Referred-By a little.
> > >
> > > As the refer-sec-options draft points out, traditional phone
> > > systems don't provide the kind of information Referred-By is
> > > trying to provide during a transfer. We've set our goals higher
> > > for SIP transfer, which is a good thing to do, but is it necessary
> > > for a base version of the application? Would a transfer application
> > > that didn't tell the transfer target who initiated the transfer be
> > > sufficient?
> > >
> > > Perhaps the Referred-By functionality should be an extension
> > (Requires:)
> > > to REFER instead of part of its base definition?
> > >
> > > This would remove the security issue with base REFER (B's triggered
> > > request would be indistinguishable from a request that B initiated
> > on
> > > its own - there is no claim in the request that the request exits
> > > because A asked for it to happen).
> > >
> > > The current work on securing Referred-By would continue in an
> > extension
> > > draft, but advancement of REFER would not be blocked on its
> > completion.
> > >
> > >
> > > RjS
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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<TITLE>RE: [Sip] REFER security options - removing Referred-By</TITLE>
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<P><FONT SIZE=3D2>Hi All,</FONT>
</P>

<P><FONT SIZE=3D2>To quickly summarize - the Referred-By is undoubtedly =
a good thing to put into a request resulting out of a REFER. However to =
quote the REFER draft </FONT></P>

<P><FONT SIZE=3D2>&lt;&lt;quote&gt;&gt;</FONT>
<BR><FONT SIZE=3D2>If the URL is a SIP URL, the Referred-By header in =
the REFER request should be copied into the request sent to the =
referred-to resource.</FONT></P>

<P><FONT SIZE=3D2>&lt;&lt;endquote&gt;&gt;</FONT>
</P>

<P><FONT SIZE=3D2>It seems to me that this of &quot;SHOULD&quot; =
strength, and anybody not willing/wanting to send it - need not. I am =
of the opinion that we use the appropriate mechanisms (either the =
SMIME/ or direct transferor to transfer target verification method) to =
ensure that this Referred-By is verifiable. Also if the transferor is =
wanting to protect itself it could always put in a URI of the =
nature:</FONT></P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>sip:anonymous@anonymous.net</FONT>
</P>

<P><FONT SIZE=3D2>and allow the transfer target the choice to accept or =
deny the call.</FONT>
</P>

<P><FONT SIZE=3D2>Removal and converting it into Requires does not buy =
us much. I vote that we continue down the path as suggested in =
drafts-sparks-sip-sec-options-00 and leave the Referred-By header as =
is.</FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Robert Sparks [<A =
HREF=3D"mailto:rsparks@dynamicsoft.com">mailto:rsparks@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Friday, April 05, 2002 9:41 AM</FONT>
<BR><FONT SIZE=3D2>To: Nigel Dewdney</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] REFER security options - removing =
Referred-By</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>On Fri, 2002-04-05 at 09:07, Nigel Dewdney =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; Pardon me if I'm being obtuse, but how do I =
tell whether an INVITE is as</FONT>
<BR><FONT SIZE=3D2>&gt; a result</FONT>
<BR><FONT SIZE=3D2>&gt; of a REFER or not if Referred-By is =
dropped?</FONT>
</P>

<P><FONT SIZE=3D2>That's exactly the point - you wouldn't.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; A calls B. B REFERs A to C,</FONT>
<BR><FONT SIZE=3D2>&gt; A calls</FONT>
<BR><FONT SIZE=3D2>&gt; C. If C can not tell that the call is a result =
of a referral from A then</FONT>
<BR><FONT SIZE=3D2>&gt; the REFER mechanism</FONT>
<BR><FONT SIZE=3D2>&gt; has been reduced into one of &quot;hey don't =
call me call them!&quot;, i.e. the</FONT>
<BR><FONT SIZE=3D2>&gt; call from A to C is</FONT>
<BR><FONT SIZE=3D2>&gt; disassociated from the dialog between A and =
B.</FONT>
</P>

<P><FONT SIZE=3D2>No - that's a separate problem. The kind of =
association you are</FONT>
<BR><FONT SIZE=3D2>looking for here is what's being addressed through =
the Replaces header.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dropping Referred-By could preclude many =
applications. Okay, an</FONT>
<BR><FONT SIZE=3D2>&gt; extension to bring</FONT>
<BR><FONT SIZE=3D2>&gt; it back in would enable them, but wasn't the =
idea of REFER that it be a</FONT>
<BR><FONT SIZE=3D2>&gt; primitive to</FONT>
<BR><FONT SIZE=3D2>&gt; enable such applications? </FONT>
<BR><FONT SIZE=3D2>&gt; The obvious example is where C wants to =
only</FONT>
<BR><FONT SIZE=3D2>&gt; accept calls</FONT>
<BR><FONT SIZE=3D2>&gt; which have been vetted by B. Irrespective =
of&nbsp; security requirements,</FONT>
<BR><FONT SIZE=3D2>&gt; this requires</FONT>
<BR><FONT SIZE=3D2>&gt; that the primitive have a reference to B in A's =
request to C. How about</FONT>
<BR><FONT SIZE=3D2>You got your alphabet mixed up there.</FONT>
<BR><FONT SIZE=3D2>&gt; applications</FONT>
<BR><FONT SIZE=3D2>&gt; where I would want to treat INVITEs differently =
according to where the</FONT>
<BR><FONT SIZE=3D2>&gt; REFER came</FONT>
<BR><FONT SIZE=3D2>&gt; from? E.g. accept calls referred from P.A. but =
fwd those reffered from</FONT>
<BR><FONT SIZE=3D2>&gt; Marketing to my</FONT>
<BR><FONT SIZE=3D2>&gt; voice-mail service.</FONT>
</P>

<P><FONT SIZE=3D2>There are _clearly_ good applications waiting for the =
Referred-By</FONT>
<BR><FONT SIZE=3D2>functionality. I am _NOT_ advocating we abandon =
providing it. What</FONT>
<BR><FONT SIZE=3D2>I'm asking is whether or not the base level transfer =
application</FONT>
<BR><FONT SIZE=3D2>really needs it. There are already fielded =
implementations that don't</FONT>
<BR><FONT SIZE=3D2>use it providing some evidence that you have a =
useful service without</FONT>
<BR><FONT SIZE=3D2>it.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; On the question of authenticating the referral =
I prefer the option of C</FONT>
<BR><FONT SIZE=3D2>&gt; VERIFYing with</FONT>
<BR><FONT SIZE=3D2>&gt; A. Why wouldn't A put a meaningful contact =
address in Referred-By: and</FONT>
<BR><FONT SIZE=3D2>&gt; expect C</FONT>
<BR><FONT SIZE=3D2>&gt; to blindly accept the referral without the =
option of being able to</FONT>
<BR><FONT SIZE=3D2>&gt; contact A? Thats not</FONT>
<BR><FONT SIZE=3D2>&gt; to say that I don't thing the S/MIME option =
isn't possible - I'm just</FONT>
<BR><FONT SIZE=3D2>&gt; uneasy about</FONT>
<BR><FONT SIZE=3D2>&gt; man-in-the-middle.</FONT>
</P>

<P><FONT SIZE=3D2>I don't follow - you're suggesting that if C tried to =
contact A and</FONT>
<BR><FONT SIZE=3D2>failed, it would just accept the request by =
default?</FONT>
</P>

<P><FONT SIZE=3D2>The real problem here is that it may be very =
difficult for A to</FONT>
<BR><FONT SIZE=3D2>provide a contact to C that will&nbsp; get back to =
the UA that has the</FONT>
<BR><FONT SIZE=3D2>knowledge about sending the REFER. Its the same =
problem</FONT>
<BR><FONT SIZE=3D2>we have with getting B's INVITE-Replaces to the same =
C we had</FONT>
<BR><FONT SIZE=3D2>a consultative-hold conference with.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Nigel Dewdney</FONT>
<BR><FONT SIZE=3D2>&gt; Laboratory Telecommunication Sciences</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; frank.derks@philips.com wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Whether &quot;traditional&quot; phone =
systems provide the sort of information</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; carried by Referred-By or not, depends on =
your definition of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &quot;traditional</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; phone system&quot; and the context in =
which REFER is being used. There are</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; several examples of the availability of =
&quot;Referred-by&quot; type of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to be found in today's telephone =
systems:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; - ISDN's (Q.931) Redirecting Number =
Information Element</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; - QSIG's (ECMA-178) rerouteingNumber =
element in the Call Transfer</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Initiate</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp; operation</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; - DPNSS' ....</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Etc.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Leaving out all the &quot;nice bits&quot; =
may result in something that will do</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; job (i.e. achieve the call transfer), but =
this also restricts new</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; services</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to be implemented, because there simply =
isn't enough information to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; infer</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; what's going on.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Frank</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sounds reasonable to me. I think it would =
be possible to phrase the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; draft in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; a way that when we add in a Referred-By =
that it does not need a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Requires and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that REFERs without a Referred-By are just =
treaded as the Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; as</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; unknown or anonymous.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cullen</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From: sip-admin@ietf.org [<A =
HREF=3D"mailto:sip-admin@ietf.org">mailto:sip-admin@ietf.org</A>]On =
Behalf Of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Robert</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sparks</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sent: Thursday, April 04, 2002 8:42 =
AM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Subject: [Sip] REFER security options =
- removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I want to explore the option of =
removing Referred-By a little.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; As the refer-sec-options draft points =
out, traditional phone</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; systems don't provide the kind of =
information Referred-By is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; trying to provide during a transfer. =
We've set our goals higher</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; for SIP transfer, which is a good =
thing to do, but is it necessary</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; for a base version of the =
application? Would a transfer application</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; that didn't tell the transfer target =
who initiated the transfer be</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; sufficient?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Perhaps the Referred-By functionality =
should be an extension</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (Requires:)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; to REFER instead of part of its base =
definition?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This would remove the security issue =
with base REFER (B's triggered</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; request would be indistinguishable =
from a request that B initiated</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; on</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; its own - there is no claim in the =
request that the request exits</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; because A asked for it to =
happen).</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The current work on securing =
Referred-By would continue in an</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; extension</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; draft, but advancement of REFER would =
not be blocked on its</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; completion.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; RjS</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; ______________________________________=
_________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This list is for NEW development of =
the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sipping@ietf.org for new =
developments on the application of sip</FONT>
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From daemon@optimus.ietf.org  Fri Apr  5 14:00:08 2002
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Subject: [Sip] draft-ietf-sip-events-05 and Record-Route
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Concerning draft-ietf-sip-events-05 and 
the Record-Route, can the record-route 
portion of the Route be updated within 
an established dialog?  

 (Note: Only the Contact portion can be 
 updated within an INVITE established 
 dialog.)  

If the answer is yes, can it be updated
by SUBSCRIBE, NOTIFY, or both?


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From daemon@optimus.ietf.org  Fri Apr  5 14:00:33 2002
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content-class: urn:content-classes:message
Subject: RE: [Sip] Comment, SIP Privacy draft
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Thread-Topic: [Sip] Comment, SIP Privacy draft
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>, <Mpierce1@aol.com>,
        <fandreas@cisco.com>
Cc: <dean.willis@softarmor.com>, <sip@ietf.org>
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> If there's no "third party" then the user does not need to include any

> identity information in the message at all. The end-to-end model tips
the 
> balance completely to the side of the caller & their desire to remain 
> anonymous (or in the absence of a PKI, to mis-represent themselves).

Not really. The e2e model cuts both ways: the called party may decide to
refuse calls that do not carry some form of identification, or at least
some form of authorization. The real privacy issue is to devise a
structure in which the caller discloses enough to convince the called
party to pick up the call, but not so much as to compromise privacy. In
SIP, the caller has also to convince potential relays to forward the
call, and the same privacy issue arises: disclose enough to convince the
intermediate party to carry the call, but not so much as to compromise
privacy.

There is a big difference between private services and public services
subject to regulation. In the latter case, third parties mandate some
level of disclosure to the network, e.g. for law enforcement purposes. A
private service may be content to forward a call using for example a
role identifier instead of a user identifier; a public service may be
requested to actually identify the user, i.e. to request more
information than the caller is willing to disclose to the called party.
This is what drives us into a band-aid architecture, in which
information is supposedly first disclosed and then at a latter stage
withdrawn.

-- Christian Huitema

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From daemon@optimus.ietf.org  Fri Apr  5 14:00:57 2002
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Subject: RE: [Sip] Comment, SIP Privacy draft
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From: "Christian Huitema" <huitema@windows.microsoft.com>
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> But as has been noted before, the IP address itself may be enough to 
> violate someone's privacy. And the end user can't leave that out. If
that 
> is to be hidden in certain cases, there needs to be another device in 
> between. Likewise, if the calling party identity is not provided, in
many 
> networks the call will never be carried, due to the requirements at
the 
> terminating end. 

Actually, you can use the IPv6 "privacy address" for that purpose. For
example, you could build up an address that is only used for one
specific call.

-- Christian Huitema
 

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From daemon@optimus.ietf.org  Fri Apr  5 15:56:42 2002
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Date: Fri, 05 Apr 2002 14:36:39 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
In-reply-to: <1017938512.1187.61.camel@dhcp222.dfw.dynamicsoft.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>, sip@ietf.org
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Robert,

I have two big problems with this proposal.

The first is that a UA can treat a Referred-By header the same as it can
a From header, since both are equally unverified and authenticated.  If
a UA is doing call screening, it can't possibly use From headers, and it
can't use Referred-By - it must use some other mechanism.  If
Referred-By is "bad" then so is "From" - which of course is ridiculous.
I would support work on an extension to REFER to allow authenticated
Referred-By, but this doesn't preclude the use of an authenticated
Referred-By.

I agree that we are all working to provide "better than PSTN" services
and features.  If we strip the Referred-By header now, and instead add
it back as an extension, there will always be implementations that don't
include it, and as a result very useful "better than PSTN" functionality
will be lost.

I would suggest a explanation in the document that basically says that
Referred-By is about as trustworthy as From - no more, no less.

Thanks,
Alan Johnston
sip:alan@siptest.wcom.com
WorldCom

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Robert Sparks
> Sent: Thursday, April 04, 2002 10:42 AM
> To: sip@ietf.org
> Subject: [Sip] REFER security options - removing Referred-By
> 
> 
> I want to explore the option of removing Referred-By a little.
> 
> As the refer-sec-options draft points out, traditional phone 
> systems don't provide the kind of information Referred-By is 
> trying to provide during a transfer. We've set our goals 
> higher for SIP transfer, which is a good thing to do, but is 
> it necessary for a base version of the application? Would a 
> transfer application that didn't tell the transfer target who 
> initiated the transfer be sufficient?
> 
> Perhaps the Referred-By functionality should be an extension 
> (Requires:) to REFER instead of part of its base definition?
> 
> This would remove the security issue with base REFER (B's 
> triggered request would be indistinguishable from a request 
> that B initiated on its own - there is no claim in the 
> request that the request exits because A asked for it to happen). 
> 
> The current work on securing Referred-By would continue in an 
> extension draft, but advancement of REFER would not be 
> blocked on its completion.
> 
> 
> RjS
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


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From daemon@optimus.ietf.org  Fri Apr  5 16:16:51 2002
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Subject: RE: [Sip] REFER security options - removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Alan Johnston <alan.johnston@wcom.com>
Cc: sip@ietf.org
In-Reply-To: <000b01c1dce1$99a8bf90$c18c4041@ajohnston>
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On Fri, 2002-04-05 at 14:36, Alan Johnston wrote:
> Robert,
> 
> I have two big problems with this proposal.
> 
> The first is that a UA can treat a Referred-By header the same as it can
> a From header, since both are equally unverified and authenticated.  If
> a UA is doing call screening, it can't possibly use From headers, and it
> can't use Referred-By - it must use some other mechanism.  If
> Referred-By is "bad" then so is "From" - which of course is ridiculous.
> I would support work on an extension to REFER to allow authenticated
> Referred-By, but this doesn't preclude the use of an authenticated
> Referred-By.

You meant unauthenticated in the last phrase, right?

This is an interesting point. The basic set of "bad things" that
someone could do by providing bogus Referred-By headers can be done
by providing bogus From headers. These things are only bad if something
is basing a processing decision based on the contents of the respective
headers.

Of course, there have been people arguing that using any information
in From: to make decisions without something some proof of identity
(either PKIish or trusted-3rd-party introduction-ish) is bad...

> 
> I agree that we are all working to provide "better than PSTN" services
> and features.  If we strip the Referred-By header now, and instead add
> it back as an extension, there will always be implementations that don't
> include it, and as a result very useful "better than PSTN" functionality
> will be lost.
> 
> I would suggest a explanation in the document that basically says that
> Referred-By is about as trustworthy as From - no more, no less.
> 
> Thanks,
> Alan Johnston
> sip:alan@siptest.wcom.com
> WorldCom
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> > Behalf Of Robert Sparks
> > Sent: Thursday, April 04, 2002 10:42 AM
> > To: sip@ietf.org
> > Subject: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > I want to explore the option of removing Referred-By a little.
> > 
> > As the refer-sec-options draft points out, traditional phone 
> > systems don't provide the kind of information Referred-By is 
> > trying to provide during a transfer. We've set our goals 
> > higher for SIP transfer, which is a good thing to do, but is 
> > it necessary for a base version of the application? Would a 
> > transfer application that didn't tell the transfer target who 
> > initiated the transfer be sufficient?
> > 
> > Perhaps the Referred-By functionality should be an extension 
> > (Requires:) to REFER instead of part of its base definition?
> > 
> > This would remove the security issue with base REFER (B's 
> > triggered request would be indistinguishable from a request 
> > that B initiated on its own - there is no claim in the 
> > request that the request exits because A asked for it to happen). 
> > 
> > The current work on securing Referred-By would continue in an 
> > extension draft, but advancement of REFER would not be 
> > blocked on its completion.
> > 
> > 
> > RjS
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current 
> > sip Use sipping@ietf.org for new developments on the 
> > application of sip
> > 



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Fri Apr  5 20:37:25 2002
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Message-ID: <05f301c1dd08$946cce00$133fed0c@C1893415A>
From: "Dean Willis" <dean.willis@softarmor.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>
Cc: <sip@ietf.org>
References: <A3C2399B2FACD411A54200508BE39C74054F70CD@zwcwd00r.europe.nortel.com>
Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva cy draft)
Date: Fri, 5 Apr 2002 19:15:44 -0600
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RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva cy =
draft)
Mark wrote:
--------
You seem to be arguing that privacy services in the network which modify =
From/To are not required at all. This could only be the case if the =
'From/To' information had the same status as, say, SMIME bodies in terms =
of what the UA is required to put in there. OK, so the SIP spec does not =
state a hard requirement on what should be put in there, but there is =
certainly a clear implication that it should be the calling/called party =
identities and indeed this *is what we want* to happen for the =
consistent operation of SIP services.
--------

Yes, that's exactly what I'm saying. Do we need to spit out a new draft =
"SIP Over Regulated Networks" saying "The stuff in the To and From and =
fields are user-provided information which will be delivered without =
modification to the termination point of the dialog"? which can the be =
referenced along with bis09 by implementors of  regulated services?

--
Dean

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Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: Private Info in To/From (was RE: FW: [Sip] =
Comment, SIP Priva cy draft)</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2713.1100" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Mark wrote:</FONT></DIV>
<DIV><FONT size=3D2>--------</FONT></DIV>
<DIV><FONT size=3D2>You seem to be arguing that privacy services in the =
network=20
which modify From/To are not required at all. This could only be the =
case if the=20
'From/To' information had the same status as, say, SMIME bodies in terms =
of what=20
the UA is required to put in there. OK, so the SIP spec does not state a =
hard=20
requirement on what should be put in there, but there is certainly a =
clear=20
implication that it should be the calling/called party identities and =
indeed=20
this *is what we want* to happen for the consistent operation of SIP=20
services.</FONT></DIV>
<DIV><FONT size=3D2>--------</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Yes, that's exactly what I'm saying. Do&nbsp;we need =
to spit=20
out a new draft "SIP Over Regulated Networks" saying "The stuff in the =
To and=20
From and fields are user-provided information which will be delivered =
without=20
modification to the termination point of the dialog"? which can the be=20
referenced along with bis09 by implementors of&nbsp; regulated=20
services?</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>--<BR>Dean</FONT></DIV></BODY></HTML>

------=_NextPart_000_05F0_01C1DCD6.48C581E0--


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr  5 21:09:56 2002
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Date: Fri, 05 Apr 2002 20:54:06 -0500
From: Allison Mankin <mankin@isi.edu>
Subject: [Sip] FW: Protocol Action: DHCP Option for SIP Servers to Proposed Standard
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Another SIP spec advances!

sip-dhcp started its long review process by DHCP folks
before the sip mailing list moved to ietf.org :)


To: IETF-Announce: ;
Cc: RFC Editor <rfc-editor@ISI.EDU>, IANA <iana@iana.org>
Cc: Internet Architecture Board <iab@ISI.EDU>
Cc: sip@lists.bell-labs.com
From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: DHCP Option for SIP Servers to Proposed
	 Standard
Date: Fri, 05 Apr 2002 17:51:19 -0500



The IESG has approved the Internet-Draft 'DHCP Option for SIP Servers'
<draft-ietf-sip-dhcp-06.txt> as a Proposed Standard.  This document
is the product of the Session Initiation Protocol Working Group.
The IESG contact persons are Allison Mankin and Scott Bradner.


Technical Summary

 The Session Initiation Protocol (SIP) is an application-layer control
 protocol that can establish, modify and terminate multimedia sessions or
 calls. The SIP WG is developing its particular use for signaling of
 Internet telephony calls.  A SIP system has two components: user agents
 and servers.  The user agent is the SIP end system that acts on behalf of
 someone who wants to participate in a SIP call.

 SIP-DHCP specifies a DHCP option that allows SIP user agents (clients)
 to locate a local SIP server that is to be used for outbound SIP
 requests, the outbound proxy server.

 The SIP client obtains a DNS string via a DHCP option.  This
 string is then used by the mechanism specified by the recently
 published Proposed Standard, Locating SIP Servers, to locate
 the outbound proxy server.

 This is one of many possible solutions for locating the outbound
 SIP server.


Working Group Summary

 The SIP working group supported this proposal.  The proposal was also
 given working group last call and very carefully reviewed by the DHC
 Working Group.  Originally the DHCP and server location specifications
 were in one draft.  The IESG requested the two be separated and the
 server location draft was advanced to Proposed Standard recently.


Protocol Quality

 This document was reviewed for the IESG by Allison Mankin.  The
 DHCP usage was reviewed in tremendous detail by Thomas Narten and
 Ralph Droms, and corrections were made and Last Called to bring
 the work in line with other DHCP usage of DNS names.


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From daemon@optimus.ietf.org  Sat Apr  6 12:00:28 2002
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Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy 
 draft)
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I have a few questions here regarding what it means and what it is that
is desired of privacy in the relation to SIP.

Are we asking to mask the identity of the calling party or provide
confidentiality or both? I am going to stick with what appears to be the
focus so far in mind, as I read the list, which is masking the true
identity. I am still digesting SIPS. 

In this case my assumption is that the topic surrounds commincations
within a service provider environment/doamin not a private individuals
SIP UA and the world, and that we dont need to do anything between
service provider proxy and calling SIP UA, so why add an extension or
additional header at all? I am also going to refer to service provider
customers as enterprise customers just for this exampe, they could just
as easily be SOHO customers.

In the case of a service provider, if we are asking to mask the
identity, as in a private number/username that we dont want disclosed to
the called, then it seems that it would be a bit simpler to maintain
that which is curently implemented in SIP, and just develop the privacy
mechanism as a feature on the SIP proxy, transparently to the Caller
(which has been provisioned due to the desire of the Caller to be a
private entity).

For example if a SIP Proxy recieves a request from the calling SIP UA,
it should then act normally on the request, with the exception that, at
this point it must modify all SIP Headers that contain information such
as IP adresses and usernames of the caller, to <private@domain.com>,
instead of caller@domain.com, and as a key to getting responses back to
the SIP UA, the SIP Proxy must maintain the Call-ID with modified host
information if there is any in the Call-ID. 

This leaves incoming requests to the SIP UA's, calling parties which
want to call this particular SIP UA, with privacy enabled, will have to
be privvy to the SIP UA actual information. This implies that a request
from a SIP UA within a SIP provider domain are really all that need to
be gaurded by the service provider. 

Example Request:

 F1 INVITE User A -> User B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@100.101.102.103
   CSeq: 1 INVITE
   Contact: <sip:UserA@100.101.102.103>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


 F1 SIP Proxy -> User B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   From: BigGuy <sip:Private@here.com>
   To: LittleGuy <sip:UserB@there.com>
   Call-ID: 12345601@200.200.200.200       /*SIP Proxies public address
   CSeq: 1 INVITE
   Contact: <sip:Private@200.200.200.200>
   Content-Type: application/sdp
   Content-Length: 147

   v=0
   o=UserA 2890844526 2890844526 IN IP4 here.com
   s=Session SDP
   c=IN IP4 100.101.102.103	/*may be actual SIP UA or B2BUA ip address
   t=0 0
   m=audio 49172 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

The only shady area here is the SDP portion which depending on the
environment it is assumed may or may not be a bad thing, since this
really depends on the enterprise policies at that point, and since DHCP
or a B2BUA may be implemented in many enterprise scenarios.

This sounds too simple to me but I wanted to know to what degree I am
wrong here, if I am at all. 

As for SMIME content, if this information is provided by the user it is
then the users problem, as is the case of enterprise security, and SIP
UA mis-configuration, which is easily detectable within the SIP
messages. This actually could be an value added service, detecting these
mis-configurations.  :^) 



Chris

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To: Robert Sparks <rsparks@dynamicsoft.com>
Cc: Nigel Dewdney <njdewdn@lts.ncsc.mil>, sip@ietf.org, sip-admin@ietf.org
Subject: Re: [Sip] REFER security options - removing Referred-By
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-



>No - that's a separate problem. The kind of association you are
>looking for here is what's being addressed through the Replaces header.

I am not convinced that Replaces by itself will completely address this
issue. Adding a Replaces header still does not explicitely convey that
the reason for the replacement is a transfer. 

The difficulty we're getting into here, is that on the one hand there is 
the REFER method with its mandatory/optional headers and on the other hand
there is the application of the REFER method. Transfer is "just" an 
example
of the latter. Anything that is specific to transfer, should probably not 
be a mandatory thing for the REFER method.

>> 
>> Dropping Referred-By could preclude many applications. Okay, an
>> extension to bring
>> it back in would enable them, but wasn't the idea of REFER that it be a
>> primitive to
>> enable such applications? 
>> The obvious example is where C wants to only
>> accept calls
>> which have been vetted by B. Irrespective of  security requirements,
>> this requires
>> that the primitive have a reference to B in A's request to C. How about
>You got your alphabet mixed up there.
>> applications
>> where I would want to treat INVITEs differently according to where the
>> REFER came
>> from? E.g. accept calls referred from P.A. but fwd those reffered from
>> Marketing to my
>> voice-mail service.

>There are _clearly_ good applications waiting for the Referred-By
>functionality. I am _NOT_ advocating we abandon providing it. What
>I'm asking is whether or not the base level transfer application
>really needs it. There are already fielded implementations that don't
>use it providing some evidence that you have a useful service without
>it.

I suppose that that is correct. On my POTS telephone, transfer works like
a charm and it (the POTS phone) certainly hasn't a clue about what's going
on. However, this is more a matter of history than anything else: there
simply didn't use to be any means to convey this sort of information. 


The fact that there are applications out there, may or may not be an 
issue. 
First of all, the SIP specification is still fairly new, so how many 
applications do you expect to be adhering to the latest SIP specification 
anyway
and how many of these "fielded applications" do you expect to remain in 
existence? 
I suspect that many of the existing implementations (based on RFC2543 or 
on 
some version of the 2543bis draft) will simply disappear in the near 
future. 
In other words, it is still early days for products based on the final 
version
of RFC2543bis. Secondly, when introducing new elements in the protocol, 
what 
_is_ important is that this does not break existing implementations. 

As long as we do not specify _exactly_ what a service (transfer in this 
case) 
is supposed to deliver, this discussion will continue and go full circle 
at 
some stage. 

>> 
>> On the question of authenticating the referral I prefer the option of C
>> VERIFYing with
>> A. Why wouldn't A put a meaningful contact address in Referred-By: and
>> expect C
>> to blindly accept the referral without the option of being able to
>> contact A? Thats not
>> to say that I don't thing the S/MIME option isn't possible - I'm just
>> uneasy about
>> man-in-the-middle.

>I don't follow - you're suggesting that if C tried to contact A and
>failed, it would just accept the request by default?

>The real problem here is that it may be very difficult for A to
>provide a contact to C that will  get back to the UA that has the
>knowledge about sending the REFER. Its the same problem
>we have with getting B's INVITE-Replaces to the same C we had
>a consultative-hold conference with.
>> 
>> Regards,
>> 
>> Nigel Dewdney
>> Laboratory Telecommunication Sciences
>> 


> 
> frank.derks@philips.com wrote:
> 
> >
> > Whether "traditional" phone systems provide the sort of information
> > that is
> > carried by Referred-By or not, depends on your definition of
> > "traditional
> > phone system" and the context in which REFER is being used. There are
> > several examples of the availability of "Referred-by" type of
> > information
> > to be found in today's telephone systems:
> >
> > - ISDN's (Q.931) Redirecting Number Information Element
> > - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> > Initiate
> >   operation
> > - DPNSS' ....
> >
> > Etc.
> >
> > Leaving out all the "nice bits" may result in something that will do
> > the
> > job (i.e. achieve the call transfer), but this also restricts new
> > services
> > to be implemented, because there simply isn't enough information to
> > infer
> > what's going on.
> >
> > Regards,
> >
> > Frank
> >
> >
> >
> >
> >
> >
> > Sounds reasonable to me. I think it would be possible to phrase the
> > draft in
> > a way that when we add in a Referred-By that it does not need a
> > Requires and
> > that REFERs without a Referred-By are just treaded as the Referred-By
> > as
> > unknown or anonymous.
> >
> > Cullen
> >
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Robert
> > > Sparks
> > > Sent: Thursday, April 04, 2002 8:42 AM
> > > To: sip@ietf.org
> > > Subject: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > I want to explore the option of removing Referred-By a little.
> > >
> > > As the refer-sec-options draft points out, traditional phone
> > > systems don't provide the kind of information Referred-By is
> > > trying to provide during a transfer. We've set our goals higher
> > > for SIP transfer, which is a good thing to do, but is it necessary
> > > for a base version of the application? Would a transfer application
> > > that didn't tell the transfer target who initiated the transfer be
> > > sufficient?
> > >
> > > Perhaps the Referred-By functionality should be an extension
> > (Requires:)
> > > to REFER instead of part of its base definition?
> > >
> > > This would remove the security issue with base REFER (B's triggered
> > > request would be indistinguishable from a request that B initiated
> > on
> > > its own - there is no claim in the request that the request exits
> > > because A asked for it to happen).
> > >
> > > The current work on securing Referred-By would continue in an
> > extension
> > > draft, but advancement of REFER would not be blocked on its
> > completion.
> > >
> > >
> > > RjS
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Content-Type: text/html; charset="us-ascii"


<br><font size=2 face="sans-serif"><br>
<br>
-</font>
<br>
<br>
<br><font size=2 face="Courier New"><br>
&gt;No - that's a separate problem. The kind of association you are<br>
&gt;looking for here is what's being addressed through the Replaces header.<br>
</font>
<br><font size=2 face="Courier New">I am not convinced that Replaces by itself will completely address this</font>
<br><font size=2 face="Courier New">issue. Adding a Replaces header still does not explicitely convey that</font>
<br><font size=2 face="Courier New">the reason for the replacement is a transfer. </font>
<br>
<br><font size=2 face="Courier New">The difficulty we're getting into here, is that on the one hand there is </font>
<br><font size=2 face="Courier New">the REFER method with its mandatory/optional headers and on the other hand</font>
<br><font size=2 face="Courier New">there is the application of the REFER method. Transfer is &quot;just&quot; an example</font>
<br><font size=2 face="Courier New">of the latter. Anything that is specific to transfer, should probably not </font>
<br><font size=2 face="Courier New">be a mandatory thing for the REFER method.</font>
<br><font size=2 face="Courier New"><br>
&gt;&gt; <br>
&gt;&gt; Dropping Referred-By could preclude many applications. Okay, an<br>
&gt;&gt; extension to bring<br>
&gt;&gt; it back in would enable them, but wasn't the idea of REFER that it be a<br>
&gt;&gt; primitive to<br>
&gt;&gt; enable such applications? <br>
&gt;&gt; The obvious example is where C wants to only<br>
&gt;&gt; accept calls<br>
&gt;&gt; which have been vetted by B. Irrespective of &nbsp;security requirements,<br>
&gt;&gt; this requires<br>
&gt;&gt; that the primitive have a reference to B in A's request to C. How about<br>
&gt;You got your alphabet mixed up there.<br>
&gt;&gt; applications<br>
&gt;&gt; where I would want to treat INVITEs differently according to where the<br>
&gt;&gt; REFER came<br>
&gt;&gt; from? E.g. accept calls referred from P.A. but fwd those reffered from<br>
&gt;&gt; Marketing to my<br>
&gt;&gt; voice-mail service.<br>
<br>
&gt;There are _clearly_ good applications waiting for the Referred-By<br>
&gt;functionality. I am _NOT_ advocating we abandon providing it. What<br>
&gt;I'm asking is whether or not the base level transfer application<br>
&gt;really needs it. There are already fielded implementations that don't<br>
&gt;use it providing some evidence that you have a useful service without<br>
&gt;it.<br>
</font>
<br><font size=2 face="Courier New">I suppose that that is correct. On my POTS telephone, transfer works like</font>
<br><font size=2 face="Courier New">a charm and it (the POTS phone) certainly hasn't a clue about what's going</font>
<br><font size=2 face="Courier New">on. However, this is more a matter of history than anything else: there</font>
<br><font size=2 face="Courier New">simply didn't use to be any means to convey this sort of information. </font>
<br>
<br>
<br><font size=2 face="Courier New">The fact that there are applications out there, may or may not be an issue. </font>
<br><font size=2 face="Courier New">First of all, the SIP specification is still fairly new, so how many </font>
<br><font size=2 face="Courier New">applications do you expect to be adhering to the latest SIP specification anyway</font>
<br><font size=2 face="Courier New">and how many of these &quot;fielded applications&quot; do you expect to remain in existence? </font>
<br><font size=2 face="Courier New">I suspect that many of the existing implementations (based on RFC2543 or on </font>
<br><font size=2 face="Courier New">some version of the 2543bis draft) will simply disappear in the near future. </font>
<br><font size=2 face="Courier New">In other words, it is still early days for products based on the final version</font>
<br><font size=2 face="Courier New">of RFC2543bis. Secondly, when introducing new elements in the protocol, what </font>
<br><font size=2 face="Courier New">_is_ important is that this does not break existing implementations. </font>
<br>
<br><font size=2 face="Courier New">As long as we do not specify _exactly_ what a service (transfer in this case) </font>
<br><font size=2 face="Courier New">is supposed to deliver, this discussion will continue and go full circle at </font>
<br><font size=2 face="Courier New">some stage. <br>
</font>
<br><font size=2 face="Courier New">&gt;&gt; <br>
&gt;&gt; On the question of authenticating the referral I prefer the option of C<br>
&gt;&gt; VERIFYing with<br>
&gt;&gt; A. Why wouldn't A put a meaningful contact address in Referred-By: and<br>
&gt;&gt; expect C<br>
&gt;&gt; to blindly accept the referral without the option of being able to<br>
&gt;&gt; contact A? Thats not<br>
&gt;&gt; to say that I don't thing the S/MIME option isn't possible - I'm just<br>
&gt;&gt; uneasy about<br>
&gt;&gt; man-in-the-middle.<br>
<br>
&gt;I don't follow - you're suggesting that if C tried to contact A and<br>
&gt;failed, it would just accept the request by default?<br>
<br>
&gt;The real problem here is that it may be very difficult for A to<br>
&gt;provide a contact to C that will &nbsp;get back to the UA that has the<br>
&gt;knowledge about sending the REFER. Its the same problem<br>
&gt;we have with getting B's INVITE-Replaces to the same C we had<br>
&gt;a consultative-hold conference with.<br>
&gt;&gt; <br>
&gt;&gt; Regards,<br>
&gt;&gt; <br>
&gt;&gt; Nigel Dewdney<br>
&gt;&gt; Laboratory Telecommunication Sciences<br>
&gt;&gt; <br>
</font>
<br>
<br><font size=2 face="Courier New">&gt; <br>
&gt; frank.derks@philips.com wrote:<br>
&gt; <br>
&gt; &gt;<br>
&gt; &gt; Whether &quot;traditional&quot; phone systems provide the sort of information<br>
&gt; &gt; that is<br>
&gt; &gt; carried by Referred-By or not, depends on your definition of<br>
&gt; &gt; &quot;traditional<br>
&gt; &gt; phone system&quot; and the context in which REFER is being used. There are<br>
&gt; &gt; several examples of the availability of &quot;Referred-by&quot; type of<br>
&gt; &gt; information<br>
&gt; &gt; to be found in today's telephone systems:<br>
&gt; &gt;<br>
&gt; &gt; - ISDN's (Q.931) Redirecting Number Information Element<br>
&gt; &gt; - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer<br>
&gt; &gt; Initiate<br>
&gt; &gt; &nbsp; operation<br>
&gt; &gt; - DPNSS' ....<br>
&gt; &gt;<br>
&gt; &gt; Etc.<br>
&gt; &gt;<br>
&gt; &gt; Leaving out all the &quot;nice bits&quot; may result in something that will do<br>
&gt; &gt; the<br>
&gt; &gt; job (i.e. achieve the call transfer), but this also restricts new<br>
&gt; &gt; services<br>
&gt; &gt; to be implemented, because there simply isn't enough information to<br>
&gt; &gt; infer<br>
&gt; &gt; what's going on.<br>
&gt; &gt;<br>
&gt; &gt; Regards,<br>
&gt; &gt;<br>
&gt; &gt; Frank<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Sounds reasonable to me. I think it would be possible to phrase the<br>
&gt; &gt; draft in<br>
&gt; &gt; a way that when we add in a Referred-By that it does not need a<br>
&gt; &gt; Requires and<br>
&gt; &gt; that REFERs without a Referred-By are just treaded as the Referred-By<br>
&gt; &gt; as<br>
&gt; &gt; unknown or anonymous.<br>
&gt; &gt;<br>
&gt; &gt; Cullen<br>
&gt; &gt;<br>
&gt; &gt; &gt; -----Original Message-----<br>
&gt; &gt; &gt; From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of<br>
&gt; &gt; Robert<br>
&gt; &gt; &gt; Sparks<br>
&gt; &gt; &gt; Sent: Thursday, April 04, 2002 8:42 AM<br>
&gt; &gt; &gt; To: sip@ietf.org<br>
&gt; &gt; &gt; Subject: [Sip] REFER security options - removing Referred-By</font>
<br><font size=2 face="Courier New">&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; I want to explore the option of removing Referred-By a little.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; As the refer-sec-options draft points out, traditional phone<br>
&gt; &gt; &gt; systems don't provide the kind of information Referred-By is<br>
&gt; &gt; &gt; trying to provide during a transfer. We've set our goals higher<br>
&gt; &gt; &gt; for SIP transfer, which is a good thing to do, but is it necessary<br>
&gt; &gt; &gt; for a base version of the application? Would a transfer application<br>
&gt; &gt; &gt; that didn't tell the transfer target who initiated the transfer be<br>
&gt; &gt; &gt; sufficient?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; Perhaps the Referred-By functionality should be an extension<br>
&gt; &gt; (Requires:)<br>
&gt; &gt; &gt; to REFER instead of part of its base definition?<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; This would remove the security issue with base REFER (B's triggered<br>
&gt; &gt; &gt; request would be indistinguishable from a request that B initiated<br>
&gt; &gt; on<br>
&gt; &gt; &gt; its own - there is no claim in the request that the request exits<br>
&gt; &gt; &gt; because A asked for it to happen).<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; The current work on securing Referred-By would continue in an<br>
&gt; &gt; extension<br>
&gt; &gt; &gt; draft, but advancement of REFER would not be blocked on its<br>
&gt; &gt; completion.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; RjS<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt; _______________________________________________<br>
&gt; &gt; &gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; &gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; &gt; &gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt; &gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; &gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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To: Robert Sparks <rsparks@dynamicsoft.com>
Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org, sip-admin@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Robert,

>To be clear, I'm not suggesting we abandon the Referred-By idea. I'm
>suggesting we start with a version of REFER that doesn't provide it
>and bring it back in with a follow-on extension to REFER.

That might be an idea, but I would say that Referred-By is about the
only thing that can be used to convey some information about the "nature"
of the REFER. Furthermore, Referred-By is not specifically tied to any
application of the REFER method. If there is service specific information
to be conveyed with the REFER (e.g. for call transfer), than this is
outside the scope of the definition of the REFER method and its 
constituent headers. 
Regards,

Frank
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<br>
<br>
<br><font size=2 face="Courier New"><br>
Robert,</font>
<br>
<br><font size=2 face="Courier New">&gt;To be clear, I'm not suggesting we abandon the Referred-By idea. I'm<br>
&gt;suggesting we start with a version of REFER that doesn't provide it<br>
&gt;and bring it back in with a follow-on extension to REFER.<br>
<br>
That might be an idea, but I would say that Referred-By is about the</font>
<br><font size=2 face="Courier New">only thing that can be used to convey some information about the &quot;nature&quot;</font>
<br><font size=2 face="Courier New">of the REFER. Furthermore, Referred-By is not specifically tied to any</font>
<br><font size=2 face="Courier New">application of the REFER method. If there is service specific information</font>
<br><font size=2 face="Courier New">to be conveyed with the REFER (e.g. for call transfer), than this is</font>
<br><font size=2 face="Courier New">outside the scope of the definition of the REFER method and its </font>
<br><font size=2 face="Courier New">constituent headers. </font>
<br><font size=2 face="sans-serif">Regards,</font>
<br>
<br><font size=2 face="sans-serif">Frank</font>
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'cmartin@dasecurenetworks.com'" <cmartin@dasecurenetworks.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy  draft)
Date: Mon, 8 Apr 2002 11:13:06 +0100 
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Chris,

I am arguing that exactly the type of service you describe below will be
required to be implemented by providers of public SIP services in order to
meet Data Protection requirements.

In particular, the values in the To/From fields will need to be modified.

Others are arguing that the transport of the To/From fields without
modification is a transparent service of SIP, and that the contents of these
fields are entirely the responsibility of the calling UA i.e. the Service
Provider has no responsibility for the information in these fields.

One important point about the service you describe below is that it cannot
be implemented on a proxy. Something more that a proxy is required.

Regards...Mark


> -----Original Message-----
> From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> Sent: 06 April 2002 17:57
> Cc: sip@ietf.org
> Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> Privacy draft)
> 
> 
> I have a few questions here regarding what it means and what 
> it is that
> is desired of privacy in the relation to SIP.
> 
> Are we asking to mask the identity of the calling party or provide
> confidentiality or both? I am going to stick with what 
> appears to be the
> focus so far in mind, as I read the list, which is masking the true
> identity. I am still digesting SIPS. 
> 
> In this case my assumption is that the topic surrounds commincations
> within a service provider environment/doamin not a private individuals
> SIP UA and the world, and that we dont need to do anything between
> service provider proxy and calling SIP UA, so why add an extension or
> additional header at all? I am also going to refer to service provider
> customers as enterprise customers just for this exampe, they 
> could just
> as easily be SOHO customers.
> 
> In the case of a service provider, if we are asking to mask the
> identity, as in a private number/username that we dont want 
> disclosed to
> the called, then it seems that it would be a bit simpler to maintain
> that which is curently implemented in SIP, and just develop 
> the privacy
> mechanism as a feature on the SIP proxy, transparently to the Caller
> (which has been provisioned due to the desire of the Caller to be a
> private entity).
> 
> For example if a SIP Proxy recieves a request from the calling SIP UA,
> it should then act normally on the request, with the 
> exception that, at
> this point it must modify all SIP Headers that contain 
> information such
> as IP adresses and usernames of the caller, to <private@domain.com>,
> instead of caller@domain.com, and as a key to getting 
> responses back to
> the SIP UA, the SIP Proxy must maintain the Call-ID with modified host
> information if there is any in the Call-ID. 
> 
> This leaves incoming requests to the SIP UA's, calling parties which
> want to call this particular SIP UA, with privacy enabled, 
> will have to
> be privvy to the SIP UA actual information. This implies that 
> a request
> from a SIP UA within a SIP provider domain are really all that need to
> be gaurded by the service provider. 
> 
> Example Request:
> 
>  F1 INVITE User A -> User B
> 
>    INVITE sip:UserB@there.com SIP/2.0
>    Via: SIP/2.0/UDP here.com:5060
>    From: BigGuy <sip:UserA@here.com>
>    To: LittleGuy <sip:UserB@there.com>
>    Call-ID: 12345601@100.101.102.103
>    CSeq: 1 INVITE
>    Contact: <sip:UserA@100.101.102.103>
>    Content-Type: application/sdp
>    Content-Length: 147
> 
>    v=0
>    o=UserA 2890844526 2890844526 IN IP4 here.com
>    s=Session SDP
>    c=IN IP4 100.101.102.103
>    t=0 0
>    m=audio 49172 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
> 
> 
>  F1 SIP Proxy -> User B
> 
>    INVITE sip:UserB@there.com SIP/2.0
>    Via: SIP/2.0/UDP here.com:5060
>    From: BigGuy <sip:Private@here.com>
>    To: LittleGuy <sip:UserB@there.com>
>    Call-ID: 12345601@200.200.200.200       /*SIP Proxies 
> public address
>    CSeq: 1 INVITE
>    Contact: <sip:Private@200.200.200.200>
>    Content-Type: application/sdp
>    Content-Length: 147
> 
>    v=0
>    o=UserA 2890844526 2890844526 IN IP4 here.com
>    s=Session SDP
>    c=IN IP4 100.101.102.103	/*may be actual SIP UA or B2BUA 
> ip address
>    t=0 0
>    m=audio 49172 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
> 
> The only shady area here is the SDP portion which depending on the
> environment it is assumed may or may not be a bad thing, since this
> really depends on the enterprise policies at that point, and 
> since DHCP
> or a B2BUA may be implemented in many enterprise scenarios.
> 
> This sounds too simple to me but I wanted to know to what degree I am
> wrong here, if I am at all. 
> 
> As for SMIME content, if this information is provided by the 
> user it is
> then the users problem, as is the case of enterprise security, and SIP
> UA mis-configuration, which is easily detectable within the SIP
> messages. This actually could be an value added service, 
> detecting these
> mis-configurations.  :^) 
> 
> 
> 
> Chris
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

------_=_NextPart_001_01C1DEE5.F9F48716
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP =
Privacy  draft)</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Chris,</FONT>
</P>

<P><FONT SIZE=3D2>I am arguing that exactly the type of service you =
describe below will be required to be implemented by providers of =
public SIP services in order to meet Data Protection =
requirements.</FONT></P>

<P><FONT SIZE=3D2>In particular, the values in the To/From fields will =
need to be modified.</FONT>
</P>

<P><FONT SIZE=3D2>Others are arguing that the transport of the To/From =
fields without modification is a transparent service of SIP, and that =
the contents of these fields are entirely the responsibility of the =
calling UA i.e. the Service Provider has no responsibility for the =
information in these fields.</FONT></P>

<P><FONT SIZE=3D2>One important point about the service you describe =
below is that it cannot be implemented on a proxy. Something more that =
a proxy is required.</FONT></P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Chris Martin [<A =
HREF=3D"mailto:cmartin@dasecurenetworks.com">mailto:cmartin@dasecurenetw=
orks.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 06 April 2002 17:57</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: Private Info in To/From (was RE: =
FW: [Sip] Comment, SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Privacy draft)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I have a few questions here regarding what it =
means and what </FONT>
<BR><FONT SIZE=3D2>&gt; it is that</FONT>
<BR><FONT SIZE=3D2>&gt; is desired of privacy in the relation to =
SIP.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Are we asking to mask the identity of the =
calling party or provide</FONT>
<BR><FONT SIZE=3D2>&gt; confidentiality or both? I am going to stick =
with what </FONT>
<BR><FONT SIZE=3D2>&gt; appears to be the</FONT>
<BR><FONT SIZE=3D2>&gt; focus so far in mind, as I read the list, which =
is masking the true</FONT>
<BR><FONT SIZE=3D2>&gt; identity. I am still digesting SIPS. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In this case my assumption is that the topic =
surrounds commincations</FONT>
<BR><FONT SIZE=3D2>&gt; within a service provider environment/doamin =
not a private individuals</FONT>
<BR><FONT SIZE=3D2>&gt; SIP UA and the world, and that we dont need to =
do anything between</FONT>
<BR><FONT SIZE=3D2>&gt; service provider proxy and calling SIP UA, so =
why add an extension or</FONT>
<BR><FONT SIZE=3D2>&gt; additional header at all? I am also going to =
refer to service provider</FONT>
<BR><FONT SIZE=3D2>&gt; customers as enterprise customers just for this =
exampe, they </FONT>
<BR><FONT SIZE=3D2>&gt; could just</FONT>
<BR><FONT SIZE=3D2>&gt; as easily be SOHO customers.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In the case of a service provider, if we are =
asking to mask the</FONT>
<BR><FONT SIZE=3D2>&gt; identity, as in a private number/username that =
we dont want </FONT>
<BR><FONT SIZE=3D2>&gt; disclosed to</FONT>
<BR><FONT SIZE=3D2>&gt; the called, then it seems that it would be a =
bit simpler to maintain</FONT>
<BR><FONT SIZE=3D2>&gt; that which is curently implemented in SIP, and =
just develop </FONT>
<BR><FONT SIZE=3D2>&gt; the privacy</FONT>
<BR><FONT SIZE=3D2>&gt; mechanism as a feature on the SIP proxy, =
transparently to the Caller</FONT>
<BR><FONT SIZE=3D2>&gt; (which has been provisioned due to the desire =
of the Caller to be a</FONT>
<BR><FONT SIZE=3D2>&gt; private entity).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; For example if a SIP Proxy recieves a request =
from the calling SIP UA,</FONT>
<BR><FONT SIZE=3D2>&gt; it should then act normally on the request, =
with the </FONT>
<BR><FONT SIZE=3D2>&gt; exception that, at</FONT>
<BR><FONT SIZE=3D2>&gt; this point it must modify all SIP Headers that =
contain </FONT>
<BR><FONT SIZE=3D2>&gt; information such</FONT>
<BR><FONT SIZE=3D2>&gt; as IP adresses and usernames of the caller, to =
&lt;private@domain.com&gt;,</FONT>
<BR><FONT SIZE=3D2>&gt; instead of caller@domain.com, and as a key to =
getting </FONT>
<BR><FONT SIZE=3D2>&gt; responses back to</FONT>
<BR><FONT SIZE=3D2>&gt; the SIP UA, the SIP Proxy must maintain the =
Call-ID with modified host</FONT>
<BR><FONT SIZE=3D2>&gt; information if there is any in the Call-ID. =
</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This leaves incoming requests to the SIP UA's, =
calling parties which</FONT>
<BR><FONT SIZE=3D2>&gt; want to call this particular SIP UA, with =
privacy enabled, </FONT>
<BR><FONT SIZE=3D2>&gt; will have to</FONT>
<BR><FONT SIZE=3D2>&gt; be privvy to the SIP UA actual information. =
This implies that </FONT>
<BR><FONT SIZE=3D2>&gt; a request</FONT>
<BR><FONT SIZE=3D2>&gt; from a SIP UA within a SIP provider domain are =
really all that need to</FONT>
<BR><FONT SIZE=3D2>&gt; be gaurded by the service provider. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Example Request:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; F1 INVITE User A -&gt; User B</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; INVITE sip:UserB@there.com =
SIP/2.0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP =
here.com:5060</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; From: BigGuy =
&lt;sip:UserA@here.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; To: LittleGuy =
&lt;sip:UserB@there.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Call-ID: =
12345601@100.101.102.103</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; CSeq: 1 INVITE</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Contact: =
&lt;sip:UserA@100.101.102.103&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Content-Type: =
application/sdp</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Content-Length: 147</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; o=3DUserA 2890844526 =
2890844526 IN IP4 here.com</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; s=3DSession SDP</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
100.101.102.103</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0 PCMU/8000</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; F1 SIP Proxy -&gt; User B</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; INVITE sip:UserB@there.com =
SIP/2.0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP =
here.com:5060</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; From: BigGuy =
&lt;sip:Private@here.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; To: LittleGuy =
&lt;sip:UserB@there.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Call-ID: =
12345601@200.200.200.200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /*SIP =
Proxies </FONT>
<BR><FONT SIZE=3D2>&gt; public address</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; CSeq: 1 INVITE</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Contact: =
&lt;sip:Private@200.200.200.200&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Content-Type: =
application/sdp</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Content-Length: 147</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; o=3DUserA 2890844526 =
2890844526 IN IP4 here.com</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; s=3DSession SDP</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
100.101.102.103&nbsp;&nbsp; /*may be actual SIP UA or B2BUA </FONT>
<BR><FONT SIZE=3D2>&gt; ip address</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 RTP/AVP =
0</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0 PCMU/8000</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The only shady area here is the SDP portion =
which depending on the</FONT>
<BR><FONT SIZE=3D2>&gt; environment it is assumed may or may not be a =
bad thing, since this</FONT>
<BR><FONT SIZE=3D2>&gt; really depends on the enterprise policies at =
that point, and </FONT>
<BR><FONT SIZE=3D2>&gt; since DHCP</FONT>
<BR><FONT SIZE=3D2>&gt; or a B2BUA may be implemented in many =
enterprise scenarios.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This sounds too simple to me but I wanted to =
know to what degree I am</FONT>
<BR><FONT SIZE=3D2>&gt; wrong here, if I am at all. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; As for SMIME content, if this information is =
provided by the </FONT>
<BR><FONT SIZE=3D2>&gt; user it is</FONT>
<BR><FONT SIZE=3D2>&gt; then the users problem, as is the case of =
enterprise security, and SIP</FONT>
<BR><FONT SIZE=3D2>&gt; UA mis-configuration, which is easily =
detectable within the SIP</FONT>
<BR><FONT SIZE=3D2>&gt; messages. This actually could be an value added =
service, </FONT>
<BR><FONT SIZE=3D2>&gt; detecting these</FONT>
<BR><FONT SIZE=3D2>&gt; mis-configurations.&nbsp; :^) </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Chris</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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	 <jon.peterson@neustar.biz>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	 cy draft)
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It's not about 'regulated vs non-regulated networks' - the requirements stem
from Data Protection which applies whenever a third party gets involved in
processing personal data of an individual. The involvement of regulators
comes in terms of helping industry understand how the underlying Data
Protection legislation applies to particular services, at least this is what
has happened in Europe. Your first argument in the event of a dispute would
probably be via the regulators, but not necessarily.
 
It is not you or I who would have to defend ourselves in the event of a Data
Protection dispute - it is the Service Providers. I would be interested to
hear from any Service Providers on this list whether they feel the approach
suggested by Dean below would give them enough protection from accusations
of Data Protection breaches.
 
I can't argue with your statement below - From/To get delivered unchanged to
the terminating point of the *dialog* - My contention is that to meet Data
Protection requirements, Service Providers will need to provide services
which modify From/To in the network, necessarily involving terminating the
dialog and joining it onto a new one.
 
Finally, I'll say it again, suppose we put statements in the spec now about
From/To being end-to-end transparent fields. Suppose I offer a 'subscriber
privacy' services, which strips out what it can e.g. Call-Info, Subject
etc., but leaves From/To untouched. Now almost all calls go though with
personal data of the subscriber in the From field. Are you really saying
that this would be a satisfactory 'subscriber privacy' service ? Are there
any Service providers which agree with this ?
 
Regards...Mark
 
 

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: 06 April 2002 02:16
To: Watson, Mark [MDN05:EP10:EXCH]; 'Peterson, Jon'
Cc: sip@ietf.org
Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
cy draft)


 
Mark wrote:
--------
You seem to be arguing that privacy services in the network which modify
From/To are not required at all. This could only be the case if the
'From/To' information had the same status as, say, SMIME bodies in terms of
what the UA is required to put in there. OK, so the SIP spec does not state
a hard requirement on what should be put in there, but there is certainly a
clear implication that it should be the calling/called party identities and
indeed this *is what we want* to happen for the consistent operation of SIP
services.
--------
 
Yes, that's exactly what I'm saying. Do we need to spit out a new draft "SIP
Over Regulated Networks" saying "The stuff in the To and From and fields are
user-provided information which will be delivered without modification to
the termination point of the dialog"? which can the be referenced along with
bis09 by implementors of  regulated services?
 
--
Dean


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva cy draft)</TITLE>

<META content="MSHTML 5.00.3315.2870" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=092181710-08042002>It's 
not about 'regulated vs non-regulated networks' - the requirements stem from 
Data Protection which applies whenever a third party gets involved in processing 
personal data of an individual. The involvement of regulators comes in terms of 
helping industry understand how the underlying Data Protection legislation 
applies to particular services, at least this is what has happened in Europe. 
Your first argument in the event of a dispute would probably be via the 
regulators, but not necessarily.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=092181710-08042002>It 
is not you or I who would have to defend ourselves in the event of a Data 
Protection dispute - it is the Service Providers. I would be interested to hear 
from any Service Providers on this list whether they feel the approach suggested 
by Dean below would give them enough protection from accusations of Data 
Protection breaches.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=092181710-08042002>I 
can't argue with your statement below - From/To get delivered unchanged to the 
terminating point of the *dialog* - My contention is that to meet Data 
Protection requirements, Service Providers will need to provide services which 
modify From/To in the network, necessarily involving terminating the dialog and 
joining it onto a new one.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002>Finally, I'll say it again, suppose we put statements 
in the spec now about From/To being end-to-end transparent fields. Suppose I 
offer a 'subscriber privacy' services, which strips out what it can e.g. 
Call-Info, Subject etc., but leaves From/To untouched. Now almost all calls go 
though with personal data of the subscriber in the From field. Are you really 
saying that this would be a satisfactory 'subscriber privacy' service ? Are 
there any Service providers which agree with this ?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002>Regards...Mark</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=092181710-08042002></SPAN></FONT>&nbsp;</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Dean Willis 
  [mailto:dean.willis@softarmor.com]<BR><B>Sent:</B> 06 April 2002 
  02:16<BR><B>To:</B> Watson, Mark [MDN05:EP10:EXCH]; 'Peterson, 
  Jon'<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> Re: Private Info in To/From 
  (was RE: FW: [Sip] Comment, SIP Priva cy draft)<BR><BR></DIV></FONT>
  <DIV><FONT size=2></FONT>&nbsp;</DIV>
  <DIV><FONT size=2>Mark wrote:</FONT></DIV>
  <DIV><FONT size=2>--------</FONT></DIV>
  <DIV><FONT size=2>You seem to be arguing that privacy services in the network 
  which modify From/To are not required at all. This could only be the case if 
  the 'From/To' information had the same status as, say, SMIME bodies in terms 
  of what the UA is required to put in there. OK, so the SIP spec does not state 
  a hard requirement on what should be put in there, but there is certainly a 
  clear implication that it should be the calling/called party identities and 
  indeed this *is what we want* to happen for the consistent operation of SIP 
  services.</FONT></DIV>
  <DIV><FONT size=2>--------</FONT></DIV>
  <DIV><FONT size=2></FONT>&nbsp;</DIV>
  <DIV><FONT size=2>Yes, that's exactly what I'm saying. Do&nbsp;we need to spit 
  out a new draft "SIP Over Regulated Networks" saying "The stuff in the To and 
  From and fields are user-provided information which will be delivered without 
  modification to the termination point of the dialog"? which can the be 
  referenced along with bis09 by implementors of&nbsp; regulated 
  services?</FONT></DIV>
  <DIV><FONT size=2></FONT>&nbsp;</DIV>
  <DIV><FONT size=2>--<BR>Dean</FONT></DIV></BLOCKQUOTE></BODY></HTML>

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From daemon@ns.ietf.org  Mon Apr  8 07:22:27 2002
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Date: Mon, 08 Apr 2002 13:59:49 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Reason format: removing reason codes
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Hello,

There has been some discusion about the format of the Reason header
field. Right now this header field contains, besides a SIP status code
and the reason phrase, a reason-code. This is a new code that identifies
the reason why the request was sent. 

http://search.ietf.org/internet-drafts/draft-schulzrinne-sip-reason-01.txt

It has been proposed that we get rid of this reason code and use only
SIP status codes. I have analyzed several scenarios where the reason
header field may be useful to see if we can cover all our requirements
by using only status codes:

1) 155 responses. In this case we do not need anything besides the
status code of the final response.

2) CANCEL or BYE is sent because another branch has picked up. In this
situation a Reason header field with a 200 OK inside would do the work.

If the CANCEL or BYE have been sent because the user was tired of
waiting for the callee to answer (or hanged up in the middle of the
session), a 603 decline can be included.

This has the good property that it would work in the same way for 3pcc:

     A         Controller             B

     |              |                 |
     |              |----INV--------->|
     |              |<-200 OK SDP-----|
     |              |--ACK held------>|
     |<-----INV-----|                 |
     |-603 decline->|                 |
     |<----ACK------|                 |
     |              |-BYE Reason 603->|

In this 3pcc scenario, if A returns a 486 busy, the status code is
encapsulated in the Reason of the BYE as well.

3) BYE or CANCEL because no media was received.

We do not have a status code for this. However, we could create one,
since it may be also useful for response. For example, an application
server receives an INVITE and sends back a 183 to establish early media.
It needs to get a pin number using voive recognition. If after a while
no media has been received, the application server can respond with the
new status code 4xx No media received.

4) re-INVITE for refresh and re-INVITE to try to solve network problems.
In the first case the UAS wants to respond with the current SDP, but in
the second case the UAS might want to provide an alternative network
address, for instance. The status code defined above in 3) could be used
here.

4) When an unacceptable offer that cannot be refused (i.e, it came in
the 200 OK for a re-INVITE) is received.

  |-re-INVITE----------->|
  |<--200 OK SDP 1-------|
  |----ACK SDP 2-------->|
  |                      |
  |--BYE reason 488----->|

The SDP in the ACK is well formed, but it is useless since the SDP in
the 200 OK is unacceptable for the UAC

488 would also cover the scenario below, when a re-INVITE is refused and
the UAC does not want to continue with the session, since its re-INVITE
was refused.

  |-re-INVITE-SDP 1----->|
  |<--200 OK SDP 2-------|
  |----ACK-------------->|
  |                      |
  |--BYE reason 488----->| 



As I said in a previous mail, I believe that the reason why an initial
request (outside a dialog) has been sent is outside the scope of
"Reason". I can specify why I am sending an INVITE using the subject
header field, or the s= line of SDP (or a referred-by type of
mechanism).

So, what folks think about getting rid of the reason codes and keeping
only SIP status codes?

Best regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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From daemon@ns.ietf.org  Mon Apr  8 10:05:51 2002
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Date: Mon, 08 Apr 2002 07:58:20 -0500
From: Chris Martin <cmartin@dasecurenetworks.com>
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Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy  
 draft)
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Mark,
Commments inline

> Mark Watson wrote:
> 
> Chris,
> 
> I am arguing that exactly the type of service you describe below will
> be required to be implemented by providers of public SIP services in
> order to meet Data Protection requirements.

The features of topology hiding, could be implemented on a separate
element to provide for the protection of the IP addresses within the
SDP, etc., but I think that it still doesn't require any additional
headers to provide this functionality.  

> 
> In particular, the values in the To/From fields will need to be
> modified.

I agree with you in respect to confidentiality and topolgy hiding (IP
modification) of the internal calling domain, but also believe that the
privacy aspect can be performed on the SIP proxy for the From Headers. I
also believe that the To headers, in relation to outbound calls
originating from the calling domain of the caller, when they relate to
the called party, outside of the calling domain, doesn't really require
modification, in my mind as long as the From has been modified to
specify that the caller is calling from a private number. 

> 
> Others are arguing that the transport of the To/From fields without
> modification is a transparent service of SIP, and that the contents of
> these fields are entirely the responsibility of the calling UA i.e.
> the Service Provider has no responsibility for the information in
> these fields.

Ya, and this is where I totally agree with you, in that it must be
addressed at this level by the SIP service provider. 

The reason for this is that this is where the provisioning is performed
for customers of SIP carriers. 

Individuals can place anything they want in the From field merely by
configuring the SIP UA to do so. But in the case of a carrier much more
control and policy is required.

> 
> One important point about the service you describe below is that it
> cannot be implemented on a proxy. Something more that a proxy is
> required.

See my first comment on this one.

I have been working with many firewall vendors to assist them in their
development of both SIP enabled Firewall ALG's (SFA) and SIP enabled
Firewall Proxies (SFP)(These are basically outbound proxies). There are
several vendors now, which I will leave unnamed, that now support SIP
and other vendors still on the way, I will let them jump in and name
themselves if they so desire.  These vendors may very well provide the
functionality for the capabilities that you are looking for, in the very
near future. They have proven to do this job very nicely so far, and in
some instance provide additional security features that will also be
very beneficial in the SIP environment.


Thanks,
Chris
> 
> Regards...Mark
> 
> > -----Original Message-----
> > From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> > Sent: 06 April 2002 17:57
> > Cc: sip@ietf.org
> > Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> 
> > Privacy draft)
> >
> >
> > I have a few questions here regarding what it means and what
> > it is that
> > is desired of privacy in the relation to SIP.
> >
> > Are we asking to mask the identity of the calling party or provide
> > confidentiality or both? I am going to stick with what
> > appears to be the
> > focus so far in mind, as I read the list, which is masking the true
> > identity. I am still digesting SIPS.
> >
> > In this case my assumption is that the topic surrounds commincations
> 
> > within a service provider environment/doamin not a private
> individuals
> > SIP UA and the world, and that we dont need to do anything between
> > service provider proxy and calling SIP UA, so why add an extension
> or
> > additional header at all? I am also going to refer to service
> provider
> > customers as enterprise customers just for this exampe, they
> > could just
> > as easily be SOHO customers.
> >
> > In the case of a service provider, if we are asking to mask the
> > identity, as in a private number/username that we dont want
> > disclosed to
> > the called, then it seems that it would be a bit simpler to maintain
> 
> > that which is curently implemented in SIP, and just develop
> > the privacy
> > mechanism as a feature on the SIP proxy, transparently to the Caller
> 
> > (which has been provisioned due to the desire of the Caller to be a
> > private entity).
> >
> > For example if a SIP Proxy recieves a request from the calling SIP
> UA,
> > it should then act normally on the request, with the
> > exception that, at
> > this point it must modify all SIP Headers that contain
> > information such
> > as IP adresses and usernames of the caller, to <private@domain.com>,
> 
> > instead of caller@domain.com, and as a key to getting
> > responses back to
> > the SIP UA, the SIP Proxy must maintain the Call-ID with modified
> host
> > information if there is any in the Call-ID.
> >
> > This leaves incoming requests to the SIP UA's, calling parties which
> 
> > want to call this particular SIP UA, with privacy enabled,
> > will have to
> > be privvy to the SIP UA actual information. This implies that
> > a request
> > from a SIP UA within a SIP provider domain are really all that need
> to
> > be gaurded by the service provider.
> >
> > Example Request:
> >
> >  F1 INVITE User A -> User B
> >
> >    INVITE sip:UserB@there.com SIP/2.0
> >    Via: SIP/2.0/UDP here.com:5060
> >    From: BigGuy <sip:UserA@here.com>
> >    To: LittleGuy <sip:UserB@there.com>
> >    Call-ID: 12345601@100.101.102.103
> >    CSeq: 1 INVITE
> >    Contact: <sip:UserA@100.101.102.103>
> >    Content-Type: application/sdp
> >    Content-Length: 147
> >
> >    v=0
> >    o=UserA 2890844526 2890844526 IN IP4 here.com
> >    s=Session SDP
> >    c=IN IP4 100.101.102.103
> >    t=0 0
> >    m=audio 49172 RTP/AVP 0
> >    a=rtpmap:0 PCMU/8000
> >
> >
> >  F1 SIP Proxy -> User B
> >
> >    INVITE sip:UserB@there.com SIP/2.0
> >    Via: SIP/2.0/UDP here.com:5060
> >    From: BigGuy <sip:Private@here.com>
> >    To: LittleGuy <sip:UserB@there.com>
> >    Call-ID: 12345601@200.200.200.200       /*SIP Proxies
> > public address
> >    CSeq: 1 INVITE
> >    Contact: <sip:Private@200.200.200.200>
> >    Content-Type: application/sdp
> >    Content-Length: 147
> >
> >    v=0
> >    o=UserA 2890844526 2890844526 IN IP4 here.com
> >    s=Session SDP
> >    c=IN IP4 100.101.102.103   /*may be actual SIP UA or B2BUA
> > ip address
> >    t=0 0
> >    m=audio 49172 RTP/AVP 0
> >    a=rtpmap:0 PCMU/8000
> >
> > The only shady area here is the SDP portion which depending on the
> > environment it is assumed may or may not be a bad thing, since this
> > really depends on the enterprise policies at that point, and
> > since DHCP
> > or a B2BUA may be implemented in many enterprise scenarios.
> >
> > This sounds too simple to me but I wanted to know to what degree I
> am
> > wrong here, if I am at all.
> >
> > As for SMIME content, if this information is provided by the
> > user it is
> > then the users problem, as is the case of enterprise security, and
> SIP
> > UA mis-configuration, which is easily detectable within the SIP
> > messages. This actually could be an value added service,
> > detecting these
> > mis-configurations.  :^)
> >
> >
> >
> > Chris
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >

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From daemon@ns.ietf.org  Mon Apr  8 10:07:35 2002
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From: Nigel Dewdney <njdewdn@lts.ncsc.mil>
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To: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: Re: [Sip] REFER security options - removing Referred-By
References: <OF9A02BEC6.C8C895D1-ONC1256B92.00275601@diamond.philips.com> 
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Robert,
Inline:

Robert Sparks wrote:

> On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote:
> > Pardon me if I'm being obtuse, but how do I tell whether an INVITE is as
> > a result
> > of a REFER or not if Referred-By is dropped?
>
> That's exactly the point - you wouldn't.
>

I would argue this is a bad thing. A referral and an unsolicited invited are
conceptually two different things. The endpoint should be aware of the
type of process it is involved in. Whether it chooses to do anything
different as a result is another question.
[snip]

> > the REFER mechanism
> > has been reduced into one of "hey don't call me call them!", i.e. the
> > call from A to C is
> > disassociated from the dialog between A and B.
>
> No - that's a separate problem. The kind of association you are
> looking for here is what's being addressed through the Replaces header.
>

Sorry, I meant association at a functional or application level. It comes down
to
a question of granularity. Do you consider the referral process as one that
just
tells the referree where to go, or one that includes the point of referral?
[snip]

> > The obvious example is where C wants to only
> > accept calls
> > which have been vetted by B. Irrespective of  security requirements,
> > this requires
> > that the primitive have a reference to B in A's request to C. How about
> You got your alphabet mixed up there.

Er, no. This is just another reference to Referred-By (no pun intended).
A->B
A<-B "please A, REFER to C"
A->C
C - "hey did A go through B before contacting me?"
[snip]

>
> There are _clearly_ good applications waiting for the Referred-By
> functionality. I am _NOT_ advocating we abandon providing it. What
> I'm asking is whether or not the base level transfer application
> really needs it. There are already fielded implementations that don't
> use it providing some evidence that you have a useful service without
> it.
>

I didn't think for a moment that you were! :~) You could certainly achieve
some base functionality without Referred-By, but then you could apply
the same argument to other SIP methods. You could strip out some of the
headers in INVITE and still get some base functionality, but I don't think
anyone is arguing for that. You could remove the From header for example,
as has already been alluded to on this thread. Its another point of
granularity -
you support a lot more with it than without. You can ignore the header if
your application doesn't require it (treat it like the From: header).

regards,

Nigel Dewdney
Laboratory Telecommunication Sciences




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From daemon@ns.ietf.org  Mon Apr  8 10:24:40 2002
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Hi friends,                                                           
                                                                      
I am a graduate student at NC State University. I am required to      
design the control memory and functional specifications for SIP       
protocol looking at the RFC 2543. However the RFC does not specify the
length of header fields in the URI for the request or response        
messages.I cannot design a control memory without knowing how many    
bytes each field occupies.Can anybody help me out with this?          
                                                                      
Thanks!                                                               
VS                                                                    


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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'cmartin@dasecurenetworks.com'" <cmartin@dasecurenetworks.com>
Cc: sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy   draft)
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Chris,

Regarding whether these privacy features can be provided by a proxy,
modification of the From/To fields cannot be done by a proxy according to
the current specification. You require a device which is more clever than a
proxy, as it has to maintain the link between the From/To values of the
incoming dialog and those of the outgoing dialog, performing modification in
both directions.

Otherwise it will not work with an RFC2543 client, which matches dialogs
based on the whole From/To values. If you don't care about RFC2543 clients,
then it would work, but you would still not be a proxy according to the RFC.

I am not foolish enough to mention the name for such devices, but I expect
you can guess :-)

Regarding the To field, if A calls B and B forwards to C, then B has the
right to ask their Service Provider for the forwarding to be carried out
'anonymously'. As well as precluding forwarding based on redirection, this
would imply modification of the To field, in which B's identity is carried
(amongst other things).

Regards,

Mark

> -----Original Message-----
> From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> Sent: 08 April 2002 13:58
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: sip@ietf.org
> Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> Privacy draft)
> 
> 
> Mark,
> Commments inline
> 
> > Mark Watson wrote:
> > 
> > Chris,
> > 
> > I am arguing that exactly the type of service you describe 
> below will
> > be required to be implemented by providers of public SIP services in
> > order to meet Data Protection requirements.
> 
> The features of topology hiding, could be implemented on a separate
> element to provide for the protection of the IP addresses within the
> SDP, etc., but I think that it still doesn't require any additional
> headers to provide this functionality.  
> 
> > 
> > In particular, the values in the To/From fields will need to be
> > modified.
> 
> I agree with you in respect to confidentiality and topolgy hiding (IP
> modification) of the internal calling domain, but also 
> believe that the
> privacy aspect can be performed on the SIP proxy for the From 
> Headers. I
> also believe that the To headers, in relation to outbound calls
> originating from the calling domain of the caller, when they relate to
> the called party, outside of the calling domain, doesn't 
> really require
> modification, in my mind as long as the From has been modified to
> specify that the caller is calling from a private number. 
> 
> > 
> > Others are arguing that the transport of the To/From fields without
> > modification is a transparent service of SIP, and that the 
> contents of
> > these fields are entirely the responsibility of the calling UA i.e.
> > the Service Provider has no responsibility for the information in
> > these fields.
> 
> Ya, and this is where I totally agree with you, in that it must be
> addressed at this level by the SIP service provider. 
> 
> The reason for this is that this is where the provisioning is 
> performed
> for customers of SIP carriers. 
> 
> Individuals can place anything they want in the From field merely by
> configuring the SIP UA to do so. But in the case of a carrier 
> much more
> control and policy is required.
> 
> > 
> > One important point about the service you describe below is that it
> > cannot be implemented on a proxy. Something more that a proxy is
> > required.
> 
> See my first comment on this one.
> 
> I have been working with many firewall vendors to assist them in their
> development of both SIP enabled Firewall ALG's (SFA) and SIP enabled
> Firewall Proxies (SFP)(These are basically outbound proxies). 
> There are
> several vendors now, which I will leave unnamed, that now support SIP
> and other vendors still on the way, I will let them jump in and name
> themselves if they so desire.  These vendors may very well provide the
> functionality for the capabilities that you are looking for, 
> in the very
> near future. They have proven to do this job very nicely so 
> far, and in
> some instance provide additional security features that will also be
> very beneficial in the SIP environment.
> 
> 
> Thanks,
> Chris
> > 
> > Regards...Mark
> > 
> > > -----Original Message-----
> > > From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> > > Sent: 06 April 2002 17:57
> > > Cc: sip@ietf.org
> > > Subject: Re: Private Info in To/From (was RE: FW: [Sip] 
> Comment, SIP
> > 
> > > Privacy draft)
> > >
> > >
> > > I have a few questions here regarding what it means and what
> > > it is that
> > > is desired of privacy in the relation to SIP.
> > >
> > > Are we asking to mask the identity of the calling party or provide
> > > confidentiality or both? I am going to stick with what
> > > appears to be the
> > > focus so far in mind, as I read the list, which is 
> masking the true
> > > identity. I am still digesting SIPS.
> > >
> > > In this case my assumption is that the topic surrounds 
> commincations
> > 
> > > within a service provider environment/doamin not a private
> > individuals
> > > SIP UA and the world, and that we dont need to do anything between
> > > service provider proxy and calling SIP UA, so why add an extension
> > or
> > > additional header at all? I am also going to refer to service
> > provider
> > > customers as enterprise customers just for this exampe, they
> > > could just
> > > as easily be SOHO customers.
> > >
> > > In the case of a service provider, if we are asking to mask the
> > > identity, as in a private number/username that we dont want
> > > disclosed to
> > > the called, then it seems that it would be a bit simpler 
> to maintain
> > 
> > > that which is curently implemented in SIP, and just develop
> > > the privacy
> > > mechanism as a feature on the SIP proxy, transparently to 
> the Caller
> > 
> > > (which has been provisioned due to the desire of the 
> Caller to be a
> > > private entity).
> > >
> > > For example if a SIP Proxy recieves a request from the calling SIP
> > UA,
> > > it should then act normally on the request, with the
> > > exception that, at
> > > this point it must modify all SIP Headers that contain
> > > information such
> > > as IP adresses and usernames of the caller, to 
> <private@domain.com>,
> > 
> > > instead of caller@domain.com, and as a key to getting
> > > responses back to
> > > the SIP UA, the SIP Proxy must maintain the Call-ID with modified
> > host
> > > information if there is any in the Call-ID.
> > >
> > > This leaves incoming requests to the SIP UA's, calling 
> parties which
> > 
> > > want to call this particular SIP UA, with privacy enabled,
> > > will have to
> > > be privvy to the SIP UA actual information. This implies that
> > > a request
> > > from a SIP UA within a SIP provider domain are really all 
> that need
> > to
> > > be gaurded by the service provider.
> > >
> > > Example Request:
> > >
> > >  F1 INVITE User A -> User B
> > >
> > >    INVITE sip:UserB@there.com SIP/2.0
> > >    Via: SIP/2.0/UDP here.com:5060
> > >    From: BigGuy <sip:UserA@here.com>
> > >    To: LittleGuy <sip:UserB@there.com>
> > >    Call-ID: 12345601@100.101.102.103
> > >    CSeq: 1 INVITE
> > >    Contact: <sip:UserA@100.101.102.103>
> > >    Content-Type: application/sdp
> > >    Content-Length: 147
> > >
> > >    v=0
> > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > >    s=Session SDP
> > >    c=IN IP4 100.101.102.103
> > >    t=0 0
> > >    m=audio 49172 RTP/AVP 0
> > >    a=rtpmap:0 PCMU/8000
> > >
> > >
> > >  F1 SIP Proxy -> User B
> > >
> > >    INVITE sip:UserB@there.com SIP/2.0
> > >    Via: SIP/2.0/UDP here.com:5060
> > >    From: BigGuy <sip:Private@here.com>
> > >    To: LittleGuy <sip:UserB@there.com>
> > >    Call-ID: 12345601@200.200.200.200       /*SIP Proxies
> > > public address
> > >    CSeq: 1 INVITE
> > >    Contact: <sip:Private@200.200.200.200>
> > >    Content-Type: application/sdp
> > >    Content-Length: 147
> > >
> > >    v=0
> > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > >    s=Session SDP
> > >    c=IN IP4 100.101.102.103   /*may be actual SIP UA or B2BUA
> > > ip address
> > >    t=0 0
> > >    m=audio 49172 RTP/AVP 0
> > >    a=rtpmap:0 PCMU/8000
> > >
> > > The only shady area here is the SDP portion which depending on the
> > > environment it is assumed may or may not be a bad thing, 
> since this
> > > really depends on the enterprise policies at that point, and
> > > since DHCP
> > > or a B2BUA may be implemented in many enterprise scenarios.
> > >
> > > This sounds too simple to me but I wanted to know to what degree I
> > am
> > > wrong here, if I am at all.
> > >
> > > As for SMIME content, if this information is provided by the
> > > user it is
> > > then the users problem, as is the case of enterprise security, and
> > SIP
> > > UA mis-configuration, which is easily detectable within the SIP
> > > messages. This actually could be an value added service,
> > > detecting these
> > > mis-configurations.  :^)
> > >
> > >
> > >
> > > Chris
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> 

------_=_NextPart_001_01C1DF05.3D281B96
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP =
Privacy   draft)</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Chris,</FONT>
</P>

<P><FONT SIZE=3D2>Regarding whether these privacy features can be =
provided by a proxy, modification of the From/To fields cannot be done =
by a proxy according to the current specification. You require a device =
which is more clever than a proxy, as it has to maintain the link =
between the From/To values of the incoming dialog and those of the =
outgoing dialog, performing modification in both directions.</FONT></P>

<P><FONT SIZE=3D2>Otherwise it will not work with an RFC2543 client, =
which matches dialogs based on the whole From/To values. If you don't =
care about RFC2543 clients, then it would work, but you would still not =
be a proxy according to the RFC.</FONT></P>

<P><FONT SIZE=3D2>I am not foolish enough to mention the name for such =
devices, but I expect you can guess :-)</FONT>
</P>

<P><FONT SIZE=3D2>Regarding the To field, if A calls B and B forwards =
to C, then B has the right to ask their Service Provider for the =
forwarding to be carried out 'anonymously'. As well as precluding =
forwarding based on redirection, this would imply modification of the =
To field, in which B's identity is carried (amongst other =
things).</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Chris Martin [<A =
HREF=3D"mailto:cmartin@dasecurenetworks.com">mailto:cmartin@dasecurenetw=
orks.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 08 April 2002 13:58</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: Private Info in To/From (was RE: =
FW: [Sip] Comment, SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Privacy draft)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark,</FONT>
<BR><FONT SIZE=3D2>&gt; Commments inline</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Mark Watson wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Chris,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I am arguing that exactly the type of =
service you describe </FONT>
<BR><FONT SIZE=3D2>&gt; below will</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; be required to be implemented by providers =
of public SIP services in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; order to meet Data Protection =
requirements.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The features of topology hiding, could be =
implemented on a separate</FONT>
<BR><FONT SIZE=3D2>&gt; element to provide for the protection of the IP =
addresses within the</FONT>
<BR><FONT SIZE=3D2>&gt; SDP, etc., but I think that it still doesn't =
require any additional</FONT>
<BR><FONT SIZE=3D2>&gt; headers to provide this functionality.&nbsp; =
</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; In particular, the values in the To/From =
fields will need to be</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; modified.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I agree with you in respect to confidentiality =
and topolgy hiding (IP</FONT>
<BR><FONT SIZE=3D2>&gt; modification) of the internal calling domain, =
but also </FONT>
<BR><FONT SIZE=3D2>&gt; believe that the</FONT>
<BR><FONT SIZE=3D2>&gt; privacy aspect can be performed on the SIP =
proxy for the From </FONT>
<BR><FONT SIZE=3D2>&gt; Headers. I</FONT>
<BR><FONT SIZE=3D2>&gt; also believe that the To headers, in relation =
to outbound calls</FONT>
<BR><FONT SIZE=3D2>&gt; originating from the calling domain of the =
caller, when they relate to</FONT>
<BR><FONT SIZE=3D2>&gt; the called party, outside of the calling =
domain, doesn't </FONT>
<BR><FONT SIZE=3D2>&gt; really require</FONT>
<BR><FONT SIZE=3D2>&gt; modification, in my mind as long as the From =
has been modified to</FONT>
<BR><FONT SIZE=3D2>&gt; specify that the caller is calling from a =
private number. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Others are arguing that the transport of =
the To/From fields without</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; modification is a transparent service of =
SIP, and that the </FONT>
<BR><FONT SIZE=3D2>&gt; contents of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; these fields are entirely the =
responsibility of the calling UA i.e.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the Service Provider has no responsibility =
for the information in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; these fields.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Ya, and this is where I totally agree with you, =
in that it must be</FONT>
<BR><FONT SIZE=3D2>&gt; addressed at this level by the SIP service =
provider. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The reason for this is that this is where the =
provisioning is </FONT>
<BR><FONT SIZE=3D2>&gt; performed</FONT>
<BR><FONT SIZE=3D2>&gt; for customers of SIP carriers. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Individuals can place anything they want in the =
From field merely by</FONT>
<BR><FONT SIZE=3D2>&gt; configuring the SIP UA to do so. But in the =
case of a carrier </FONT>
<BR><FONT SIZE=3D2>&gt; much more</FONT>
<BR><FONT SIZE=3D2>&gt; control and policy is required.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; One important point about the service you =
describe below is that it</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; cannot be implemented on a proxy. =
Something more that a proxy is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; required.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; See my first comment on this one.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I have been working with many firewall vendors =
to assist them in their</FONT>
<BR><FONT SIZE=3D2>&gt; development of both SIP enabled Firewall ALG's =
(SFA) and SIP enabled</FONT>
<BR><FONT SIZE=3D2>&gt; Firewall Proxies (SFP)(These are basically =
outbound proxies). </FONT>
<BR><FONT SIZE=3D2>&gt; There are</FONT>
<BR><FONT SIZE=3D2>&gt; several vendors now, which I will leave =
unnamed, that now support SIP</FONT>
<BR><FONT SIZE=3D2>&gt; and other vendors still on the way, I will let =
them jump in and name</FONT>
<BR><FONT SIZE=3D2>&gt; themselves if they so desire.&nbsp; These =
vendors may very well provide the</FONT>
<BR><FONT SIZE=3D2>&gt; functionality for the capabilities that you are =
looking for, </FONT>
<BR><FONT SIZE=3D2>&gt; in the very</FONT>
<BR><FONT SIZE=3D2>&gt; near future. They have proven to do this job =
very nicely so </FONT>
<BR><FONT SIZE=3D2>&gt; far, and in</FONT>
<BR><FONT SIZE=3D2>&gt; some instance provide additional security =
features that will also be</FONT>
<BR><FONT SIZE=3D2>&gt; very beneficial in the SIP environment.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; Chris</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Regards...Mark</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From: Chris Martin [<A =
HREF=3D"mailto:cmartin@dasecurenetworks.com">mailto:cmartin@dasecurenetw=
orks.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sent: 06 April 2002 17:57</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Subject: Re: Private Info in To/From =
(was RE: FW: [Sip] </FONT>
<BR><FONT SIZE=3D2>&gt; Comment, SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Privacy draft)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I have a few questions here regarding =
what it means and what</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; it is that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; is desired of privacy in the relation =
to SIP.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Are we asking to mask the identity of =
the calling party or provide</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; confidentiality or both? I am going =
to stick with what</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; appears to be the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; focus so far in mind, as I read the =
list, which is </FONT>
<BR><FONT SIZE=3D2>&gt; masking the true</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; identity. I am still digesting =
SIPS.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; In this case my assumption is that =
the topic surrounds </FONT>
<BR><FONT SIZE=3D2>&gt; commincations</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; within a service provider =
environment/doamin not a private</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; individuals</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; SIP UA and the world, and that we =
dont need to do anything between</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; service provider proxy and calling =
SIP UA, so why add an extension</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; or</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; additional header at all? I am also =
going to refer to service</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; provider</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; customers as enterprise customers =
just for this exampe, they</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; could just</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; as easily be SOHO customers.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; In the case of a service provider, if =
we are asking to mask the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; identity, as in a private =
number/username that we dont want</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; disclosed to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the called, then it seems that it =
would be a bit simpler </FONT>
<BR><FONT SIZE=3D2>&gt; to maintain</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; that which is curently implemented in =
SIP, and just develop</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the privacy</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; mechanism as a feature on the SIP =
proxy, transparently to </FONT>
<BR><FONT SIZE=3D2>&gt; the Caller</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; (which has been provisioned due to =
the desire of the </FONT>
<BR><FONT SIZE=3D2>&gt; Caller to be a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; private entity).</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; For example if a SIP Proxy recieves a =
request from the calling SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; UA,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; it should then act normally on the =
request, with the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; exception that, at</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; this point it must modify all SIP =
Headers that contain</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; information such</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; as IP adresses and usernames of the =
caller, to </FONT>
<BR><FONT SIZE=3D2>&gt; &lt;private@domain.com&gt;,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; instead of caller@domain.com, and as =
a key to getting</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; responses back to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the SIP UA, the SIP Proxy must =
maintain the Call-ID with modified</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; host</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; information if there is any in the =
Call-ID.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This leaves incoming requests to the =
SIP UA's, calling </FONT>
<BR><FONT SIZE=3D2>&gt; parties which</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; want to call this particular SIP UA, =
with privacy enabled,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; will have to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; be privvy to the SIP UA actual =
information. This implies that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; a request</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; from a SIP UA within a SIP provider =
domain are really all </FONT>
<BR><FONT SIZE=3D2>&gt; that need</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; be gaurded by the service =
provider.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Example Request:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; F1 INVITE User A -&gt; User =
B</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; INVITE =
sip:UserB@there.com SIP/2.0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP =
here.com:5060</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; From: BigGuy =
&lt;sip:UserA@here.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; To: LittleGuy =
&lt;sip:UserB@there.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Call-ID: =
12345601@100.101.102.103</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; CSeq: 1 =
INVITE</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Contact: =
&lt;sip:UserA@100.101.102.103&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Content-Type: =
application/sdp</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Content-Length: =
147</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; o=3DUserA =
2890844526 2890844526 IN IP4 here.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; s=3DSession =
SDP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
100.101.102.103</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 =
RTP/AVP 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0 =
PCMU/8000</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; F1 SIP Proxy -&gt; User =
B</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; INVITE =
sip:UserB@there.com SIP/2.0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Via: SIP/2.0/UDP =
here.com:5060</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; From: BigGuy =
&lt;sip:Private@here.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; To: LittleGuy =
&lt;sip:UserB@there.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Call-ID: =
12345601@200.200.200.200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /*SIP =
Proxies</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; public address</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; CSeq: 1 =
INVITE</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Contact: =
&lt;sip:Private@200.200.200.200&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Content-Type: =
application/sdp</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; Content-Length: =
147</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; v=3D0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; o=3DUserA =
2890844526 2890844526 IN IP4 here.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; s=3DSession =
SDP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; c=3DIN IP4 =
100.101.102.103&nbsp;&nbsp; /*may be actual SIP UA or B2BUA</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; ip address</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; t=3D0 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 =
RTP/AVP 0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0 =
PCMU/8000</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The only shady area here is the SDP =
portion which depending on the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; environment it is assumed may or may =
not be a bad thing, </FONT>
<BR><FONT SIZE=3D2>&gt; since this</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; really depends on the enterprise =
policies at that point, and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; since DHCP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; or a B2BUA may be implemented in many =
enterprise scenarios.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This sounds too simple to me but I =
wanted to know to what degree I</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; am</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; wrong here, if I am at all.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; As for SMIME content, if this =
information is provided by the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; user it is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; then the users problem, as is the =
case of enterprise security, and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; UA mis-configuration, which is easily =
detectable within the SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; messages. This actually could be an =
value added service,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; detecting these</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; mis-configurations.&nbsp; :^)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Chris</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This list is for NEW development of =
the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sipping@ietf.org for new =
developments on the </FONT>
<BR><FONT SIZE=3D2>&gt; application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
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Date: Mon, 08 Apr 2002 09:59:33 -0400
From: Nigel Dewdney <njdewdn@lts.ncsc.mil>
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Subject: Re: [Sip] REFER security options - removing Referred-By
References: <OF9A02BEC6.C8C895D1-ONC1256B92.00275601@diamond.philips.com> 
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Robert,

Robert Sparks wrote:
[snip]

> >
> > On the question of authenticating the referral I prefer the option of C
> > VERIFYing with
> > A. Why wouldn't A put a meaningful contact address in Referred-By: and
> > expect C
> > to blindly accept the referral without the option of being able to
> > contact A? Thats not
> > to say that I don't thing the S/MIME option isn't possible - I'm just
> > uneasy about
> > man-in-the-middle.
>
> I don't follow - you're suggesting that if C tried to contact A and
> failed, it would just accept the request by default?
>
>

That would depend on C's policy. I would expect default to be to decline.

> The real problem here is that it may be very difficult for A to
> provide a contact to C that will  get back to the UA that has the
> knowledge about sending the REFER. Its the same problem
> we have with getting B's INVITE-Replaces to the same C we had
> a consultative-hold conference with.
> >>

I'm sorry, I can't see this yet. If B can contact A, in what situation would
C not be able to contact A (except in the case where suddenly A has an
equipment failure)? The only case I can see is if A is still in dialog with B
and only has "one line". Thus A would respond to B with "Busy". For this
case I would argue that if A is capable of using the REFER method it should
be prepared to deal with another request (without SDP) from the
referral point (C), e.g. VERIFY (if that comes to light). C doesn't have to
establish
media with A for verification.

Regards,
Nigel Dewdney
Laboratory Telecommunication Sciences

> >
> >
> > frank.derks@philips.com wrote:
> >
> > >
> > > Whether "traditional" phone systems provide the sort of information
> > > that is
> > > carried by Referred-By or not, depends on your definition of
> > > "traditional
> > > phone system" and the context in which REFER is being used. There are
> > > several examples of the availability of "Referred-by" type of
> > > information
> > > to be found in today's telephone systems:
> > >
> > > - ISDN's (Q.931) Redirecting Number Information Element
> > > - QSIG's (ECMA-178) rerouteingNumber element in the Call Transfer
> > > Initiate
> > >   operation
> > > - DPNSS' ....
> > >
> > > Etc.
> > >
> > > Leaving out all the "nice bits" may result in something that will do
> > > the
> > > job (i.e. achieve the call transfer), but this also restricts new
> > > services
> > > to be implemented, because there simply isn't enough information to
> > > infer
> > > what's going on.
> > >
> > > Regards,
> > >
> > > Frank
> > >
> > >
> > >
> > >
> > >
> > >
> > > Sounds reasonable to me. I think it would be possible to phrase the
> > > draft in
> > > a way that when we add in a Referred-By that it does not need a
> > > Requires and
> > > that REFERs without a Referred-By are just treaded as the Referred-By
> > > as
> > > unknown or anonymous.
> > >
> > > Cullen
> > >
> > > > -----Original Message-----
> > > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > > Robert
> > > > Sparks
> > > > Sent: Thursday, April 04, 2002 8:42 AM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > > I want to explore the option of removing Referred-By a little.
> > > >
> > > > As the refer-sec-options draft points out, traditional phone
> > > > systems don't provide the kind of information Referred-By is
> > > > trying to provide during a transfer. We've set our goals higher
> > > > for SIP transfer, which is a good thing to do, but is it necessary
> > > > for a base version of the application? Would a transfer application
> > > > that didn't tell the transfer target who initiated the transfer be
> > > > sufficient?
> > > >
> > > > Perhaps the Referred-By functionality should be an extension
> > > (Requires:)
> > > > to REFER instead of part of its base definition?
> > > >
> > > > This would remove the security issue with base REFER (B's triggered
> > > > request would be indistinguishable from a request that B initiated
> > > on
> > > > its own - there is no claim in the request that the request exits
> > > > because A asked for it to happen).
> > > >
> > > > The current work on securing Referred-By would continue in an
> > > extension
> > > > draft, but advancement of REFER would not be blocked on its
> > > completion.
> > > >
> > > >
> > > > RjS
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the application of sip
> > > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From daemon@ns.ietf.org  Mon Apr  8 10:41:40 2002
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Date: Mon, 08 Apr 2002 17:22:20 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Juan-Carlos.Rojas@alcatel.fr
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] Re: Offer by the callee
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Hello Juan Carlos,

I have been thinking more about your scenario, and I think that sending
back in the first answer a c=0.0.0.0 line would be the right thing to
do. 
This way, even though you have returned an answer with preconditions,
the other party will not perform resource reservation, since it does not
have an IP address to send media to.

Then, when you are ready to send an new offer, you will send it in an
UPDATE. The flow would look as follows:

                        INVITE (offer-1)
------------------------------------------------------------->
       183 (answer-to-offer-1, c=0.0.0.0)    
<-------------------------------------------------------------
                             PRACK
------------------------------------------------------------->
                             200 PRACK
<-------------------------------------------------------------
                  UPDATE (offer-2)
<-------------------------------------------------------------
            200 UPDATE (answer-to-offer-2)
------------------------------------------------------------->

        <========== Resource Reservation ==========>

Best regards,

Gonzalo

Juan-Carlos.Rojas@alcatel.fr wrote:
> 
> Hi Gonzalo,
> 
> Thank you for your answer
> You are right, but sending a held SDP does not prevent the calling party to
> start resource reservation, but only to send media.
> I agree with you that normally, a port number to zero indicates refusal of
> the media stream. Nevertheless, according to Offer/Answer draft, all port
> numbers equal to zero is used also to announce capabilities (section 9),
> and very often it is used to say just "don't use this media".
> 
> What I need here is to give to the answerer the possibility to say "Mr.the
> Offerer, I'm not rejecting your Invite, but I'd like to send you a new
> offer right now, so it is useless to start resource reservation before
> receiving this new offer"
> 
> Best regards
> Juan Carlos
> 
> Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>@ietf.org on
> 20/03/2002 05:01:02
> 
> Sent by:  sip-admin@ietf.org
> 
> To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
> cc:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
> Subject:  [Sip] Re: Offer by the callee
> 
> Hi,
> 
> Setting the port number to zero means refusal of the stream, so you are
> risking that the caller sends a CANCEL, since you have refused all the
> media streams. A better solution would be to send a held SDP
> (a=inactive).
> 
> Regards,
> 
> Gonzalo
> 
> Juan-Carlos.Rojas@alcatel.fr wrote:
> >
> > Hello,
> >
> > I'm trying to connect the offer/answer, update and manyfolks drafts for
> the
> > following scenario:
> > - Alice sends an INVITE to Bob with "offer-1"
> > - Bob wants to make an "offer-2" to Alice, before the completion of the
> > session establishment
> > - Bob would like to ask Alice to start resource reservation only based on
> > the acceptation of "offer-2", and not before (in fact Bob knows that it
> is
> > useless to start resource reservation based on offer-1, because he knows
> he
> > will send immediatly a new offer-2,  but he is obliged to wait for Prack
> > before)
> >
> > The Update draft defines the response 155 to allow the answerer (the
> callee
> > in our case) to request "Mr. the calling, please *send me* a new offer",
> > but there is no way for the same callee to announce "Mr. the calling, *I
> > want to send you* a new offer right now".
> > So, one possible way to achieve that is that the callee sends an answer
> to
> > the original offer but disabling all the media (in order to avoid to
> start
> > resource reservation), and next proposing a new offer, as described in
> the
> > call flow below (assume for the example that they are using end-to-end
> > resource reservation, but please don't generate a new debate on this
> > topic):
> >
> >    Alice
> > Bob
> >                                       INVITE (offer-1)
> >        ----------------------------------------------------------------->
> >       183 (answer-to-offer-1, all port numbers to zero)
> >       <-----------------------------------------------------------------
> >                                          PRACK
> >        ----------------------------------------------------------------->
> >                                       200 PRACK
> >       <-----------------------------------------------------------------
> >                  UPDATE (offer-2, status-type=e2e)
> >       <-----------------------------------------------------------------
> >                   200 UPDATE (answer-to-offer-2)
> >        ----------------------------------------------------------------->
> >
> >    <========== Resource Reservation ==========>
> >
> >            UPDATE (current-status = desired-status)
> >        ----------------------------------------------------------------->
> >                                       200 UPDATE
> >       <-----------------------------------------------------------------
> >                                         200 INVITE
> >       <-----------------------------------------------------------------
> >                                               ACK
> >        ----------------------------------------------------------------->
> >
> > Is this scenario compliant with the drafts Offer/Answer, Update and
> > Manyfolks ?
> >
> > Thank you for your answer
> > Best regards
> > Juan Carlos
> 
> --
> Gonzalo Camarillo                    Phone :   +1 212 939 71 71
> Columbia University                  Mobile:  +358 40 702 35 35
> 472 Computer Science Building        Fax   :  +358  9 299 30 52
> 1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
> New York, NY 10027
> USA                              Gonzalo.Camarillo@ericsson.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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From daemon@ns.ietf.org  Mon Apr  8 10:58:50 2002
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Subject: Re: [Sip] REFER security options - removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Nigel Dewdney <njdewdn@lts.ncsc.mil>
Cc: sip@ietf.org
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On Mon, 2002-04-08 at 08:30, Nigel Dewdney wrote:
> Robert,
> Inline:
> 
> Robert Sparks wrote:
> 
> > On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote:
> > > Pardon me if I'm being obtuse, but how do I tell whether an INVITE is as
> > > a result
> > > of a REFER or not if Referred-By is dropped?
> >
> > That's exactly the point - you wouldn't.
> >
> 
> I would argue this is a bad thing. A referral and an unsolicited invited are
> conceptually two different things. The endpoint should be aware of the
> type of process it is involved in. Whether it chooses to do anything
> different as a result is another question.

So, are you saying
"It will not be useful to have a version of REFER that doesn't
 make the triggered request distinct from an ordinary request."?

That's different from
"It will be useful to have a version of REFER that does make
the triggered request distinct from an ordinary request."

I agree with the second statement. I'm not convinced yet on the first.

> [snip]
> 
> > > the REFER mechanism
> > > has been reduced into one of "hey don't call me call them!", i.e. the
> > > call from A to C is
> > > disassociated from the dialog between A and B.
> >
> > No - that's a separate problem. The kind of association you are
> > looking for here is what's being addressed through the Replaces header.
> >
> 
> Sorry, I meant association at a functional or application level. It comes down
> to
> a question of granularity. Do you consider the referral process as one that
> just
> tells the referree where to go, or one that includes the point of referral?
> [snip]
> 
> > > The obvious example is where C wants to only
> > > accept calls
> > > which have been vetted by B. Irrespective of  security requirements,
> > > this requires
> > > that the primitive have a reference to B in A's request to C. How about
> > You got your alphabet mixed up there.
> 
> Er, no. This is just another reference to Referred-By (no pun intended).
> A->B
> A<-B "please A, REFER to C"
> A->C
> C - "hey did A go through B before contacting me?"
> [snip]

Ah - I was using the model in the refer-sec-options draft that has
a REFER going from A to B. 
> 
> >
> > There are _clearly_ good applications waiting for the Referred-By
> > functionality. I am _NOT_ advocating we abandon providing it. What
> > I'm asking is whether or not the base level transfer application
> > really needs it. There are already fielded implementations that don't
> > use it providing some evidence that you have a useful service without
> > it.
> >
> 
> I didn't think for a moment that you were! :~) You could certainly achieve
> some base functionality without Referred-By, but then you could apply
> the same argument to other SIP methods. You could strip out some of the
> headers in INVITE and still get some base functionality, but I don't think
> anyone is arguing for that. You could remove the From header for example,
> as has already been alluded to on this thread. Its another point of
> granularity -
> you support a lot more with it than without. You can ignore the header if
> your application doesn't require it (treat it like the From: header).

Well, the same argument _did_ get applied to other SIP methods as they
were being defined. What we're left with is what we could agree on being
the minimal set of required headers. Of course, the bar was in a little
different place before we had an agreed on Supported/Require mechanism.

Your last sentence is nearing the root of the problem though.
Are you claiming that its OK to provide Referred-By and not secure it?
(where secure it means A provides a proof "X" to C through some
mechanism. X is a proof of at least A's identity, the content of the
Refer-To header A sends to B. There may be more stuff in the proof to
help defend against replay).

Are you claiming that it is sufficient to say in the draft that
an implementation receiving any request (other than a REFER) with a
Referred-By header and no such proof SHOULD ignore the header entirely
while processing the request? (This is analogous to saying an 
implementation SHOULD NOT display the contents of the From header
while the phone is alerting unless the UA has some proof of the
contents of that From header).

> 
> regards,
> 
> Nigel Dewdney
> Laboratory Telecommunication Sciences
> 
> 



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From daemon@ns.ietf.org  Mon Apr  8 11:32:35 2002
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Subject: Addressing a particular UA instance (was Re: [Sip] REFER security
	options - removing Referred-By)
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Nigel Dewdney <njdewdn@lts.ncsc.mil>
Cc: sip@ietf.org
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> > [From RjS]
> > The real problem here is that it may be very difficult for A to
> > provide a contact to C that will  get back to the UA that has the
> > knowledge about sending the REFER. Its the same problem
> > we have with getting B's INVITE-Replaces to the same C we had
> > a consultative-hold conference with.
> > >>
> [From Nigel Dewdney]
> I'm sorry, I can't see this yet. If B can contact A, in what situation would
> C not be able to contact A (except in the case where suddenly A has an
> equipment failure)? The only case I can see is if A is still in dialog with B
> and only has "one line". Thus A would respond to B with "Busy". For this
> case I would argue that if A is capable of using the REFER method it should
> be prepared to deal with another request (without SDP) from the
> referral point (C), e.g. VERIFY (if that comes to light). C doesn't have to
> establish
> media with A for verification.
> 

The problem is not with C's ability to contact A. It's C's ability to
get a SIP message to the _exact same UA_ that A is using to talk to B.
It's no good if C's request went to some other UA for A - that other UA
has no knowledge of the REFER.

We've seen this problem several times already - there have been long
threads on it in the discussions on attended transfer. In the 
cc-transfer draft, we were bringing in "accept-contact" to try to work
around it (see the far-to-sparse section titled "Consultation hold in
the presence of forking proxies in the transfer draft).

Here's a real world instance of the problem rewritten in terms of
a VERIFY.

I have 3 SIP phones active most of the day, all registering to the
same address-of-record: sip:rsparks@dynamicsoft.com. One's on my
desk, one's on my laptop, and one's in my house.

Suppose I call Kevin using my desk phone and decide I need to transfer
Kevin to you. My desk phone knows itself (at the moment) as
sip:rsparks@dhcp204.dfw.dynamicsoft.com. Say it puts that value in the
Referred-By header and passes it to Kevin. Kevin passes it to you.
You try to send a VERIFY to to that address and it fails.
Our office is set up to allow SIP traffic only through the proxies at
sip:dynamicsoft.com. Not a problem for Kevin since we got those things
on the request path using Record-Route while setting up our dialog. Big
problem for you.

So maybe I could have set my phone up to provide
sip:rsparks@dynamicsoft.com in the Referred-By instead. Then your
request would make it into our network. Now you roll the dice to see
if my desk phone is still on the list of things that it might get
forked to. If it is, you'll probably get lucky - the 200 my desk phone
returns will beat out the other phone's non-200 final responses. But all
kinds of legitimate things can happen to cause my desk phone to not be
on that list (time-of-day routing and access control lists give you
the tip of the iceberg). Let make it worse - suppose I placed my
original call to Kevin from a phone that didn't register to any
address of record at our outgoing proxies. Your request is guaranteed to
fail. (Reread this paragraph from the perspective of an INVITE that you
really wanted to reach my desk and think about what happens if someone
picks up my house phone).

Now consider environments where the endpoints live in some variant of
a private address space (nats, split-dns, etc). What kind of identifier
can they hand you that you will find useful for contacting that 
particular endpoint?

The current discussions for solving this problem involve things like
my phone including a preloaded route for you to use in the identifier I
hand to Kevin. What's not clear yet is how my phone discovers that
route.

RjS


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From daemon@ns.ietf.org  Mon Apr  8 15:11:32 2002
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Date: Mon, 08 Apr 2002 13:41:23 -0500
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
CC: sip <sip@ietf.org>
Subject: Re: [Sip] Reason format: removing reason codes
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Gonzalo Camarillo wrote:

> Hello,


Gonzalo:

Comments inline.

> It has been proposed that we get rid of this reason code and use only
> SIP status codes. 


That's fine.

> 1) 155 responses. In this case we do not need anything besides the
> status code of the final response.


155 is probably the only response to which the Reason header field
adds any value (183 was finally rejected after a few emails on the
WG list).  I really think the I-D should specify that Reason header
makes sense for 155 responses *only*.  Otherwise we will have simply
too many ways to send responses -- normal SIP response codes, Warning
header (which also carries information about the status of a response),
and now, using Reason for sending a response.

 
> 3) BYE or CANCEL because no media was received.
> 
> We do not have a status code for this. However, we could create one,
> since it may be also useful for response. 


This is tying it to a specific application.  Could we not achieve the
same result by encapsulating a "400 No Media Received" into a Reason
header as follows:

    BYE sip:mediaserver.net SIP/2.0
    Reason: 400;text="No Media Received"
    ...

Come to think of it, do we need the "triggered" parameter.  If there
is not going to be a "SIP/3.0" (is there?) then why duplicate
the SIP version already present in the R-URI?

> As I said in a previous mail, I believe that the reason why an initial
> request (outside a dialog) has been sent is outside the scope of
> "Reason". I can specify why I am sending an INVITE using the subject
> header field, or the s= line of SDP (or a referred-by type of
> mechanism).


In that case, we should just mention that using the Reason header
outside of a dialog is outside the scope of this I-D (and that
Subject or some other means can be employed for that purpose).


Best regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@ns.ietf.org  Mon Apr  8 15:36:23 2002
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From: "Sriram Parameswar"<sriramp@nortelnetworks.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>,
        Nigel Dewdney
	 <njdewdn@lts.ncsc.mil>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Hi:

The thread "Reason format: removing reason codes" triggered this thought.
Why not use a Reason header to indicate that an INVITE is due to a REFER? We
could come up with a Reason code for this. That way the argument that there
is no way to tell a regular INVITE from one that has been caused by a REFER
goes away!

Thx,

Sriram

__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
Sent: Monday, April 08, 2002 9:23 AM
To: Nigel Dewdney
Cc: sip@ietf.org
Subject: Re: [Sip] REFER security options - removing Referred-By


On Mon, 2002-04-08 at 08:30, Nigel Dewdney wrote:
> Robert,
> Inline:
> 
> Robert Sparks wrote:
> 
> > On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote:
> > > Pardon me if I'm being obtuse, but how do I tell whether an INVITE is
as
> > > a result
> > > of a REFER or not if Referred-By is dropped?
> >
> > That's exactly the point - you wouldn't.
> >
> 
> I would argue this is a bad thing. A referral and an unsolicited invited
are
> conceptually two different things. The endpoint should be aware of the
> type of process it is involved in. Whether it chooses to do anything
> different as a result is another question.

So, are you saying
"It will not be useful to have a version of REFER that doesn't
 make the triggered request distinct from an ordinary request."?

That's different from
"It will be useful to have a version of REFER that does make
the triggered request distinct from an ordinary request."

I agree with the second statement. I'm not convinced yet on the first.

> [snip]
> 
> > > the REFER mechanism
> > > has been reduced into one of "hey don't call me call them!", i.e. the
> > > call from A to C is
> > > disassociated from the dialog between A and B.
> >
> > No - that's a separate problem. The kind of association you are
> > looking for here is what's being addressed through the Replaces header.
> >
> 
> Sorry, I meant association at a functional or application level. It comes
down
> to
> a question of granularity. Do you consider the referral process as one
that
> just
> tells the referree where to go, or one that includes the point of
referral?
> [snip]
> 
> > > The obvious example is where C wants to only
> > > accept calls
> > > which have been vetted by B. Irrespective of  security requirements,
> > > this requires
> > > that the primitive have a reference to B in A's request to C. How
about
> > You got your alphabet mixed up there.
> 
> Er, no. This is just another reference to Referred-By (no pun intended).
> A->B
> A<-B "please A, REFER to C"
> A->C
> C - "hey did A go through B before contacting me?"
> [snip]

Ah - I was using the model in the refer-sec-options draft that has
a REFER going from A to B. 
> 
> >
> > There are _clearly_ good applications waiting for the Referred-By
> > functionality. I am _NOT_ advocating we abandon providing it. What
> > I'm asking is whether or not the base level transfer application
> > really needs it. There are already fielded implementations that don't
> > use it providing some evidence that you have a useful service without
> > it.
> >
> 
> I didn't think for a moment that you were! :~) You could certainly achieve
> some base functionality without Referred-By, but then you could apply
> the same argument to other SIP methods. You could strip out some of the
> headers in INVITE and still get some base functionality, but I don't think
> anyone is arguing for that. You could remove the From header for example,
> as has already been alluded to on this thread. Its another point of
> granularity -
> you support a lot more with it than without. You can ignore the header if
> your application doesn't require it (treat it like the From: header).

Well, the same argument _did_ get applied to other SIP methods as they
were being defined. What we're left with is what we could agree on being
the minimal set of required headers. Of course, the bar was in a little
different place before we had an agreed on Supported/Require mechanism.

Your last sentence is nearing the root of the problem though.
Are you claiming that its OK to provide Referred-By and not secure it?
(where secure it means A provides a proof "X" to C through some
mechanism. X is a proof of at least A's identity, the content of the
Refer-To header A sends to B. There may be more stuff in the proof to
help defend against replay).

Are you claiming that it is sufficient to say in the draft that
an implementation receiving any request (other than a REFER) with a
Referred-By header and no such proof SHOULD ignore the header entirely
while processing the request? (This is analogous to saying an 
implementation SHOULD NOT display the contents of the From header
while the phone is alerting unless the UA has some proof of the
contents of that From header).

> 
> regards,
> 
> Nigel Dewdney
> Laboratory Telecommunication Sciences
> 
> 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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<TITLE>RE: [Sip] REFER security options - removing Referred-By</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Hi:</FONT>
</P>

<P><FONT SIZE=3D2>The thread &quot;Reason format: removing reason =
codes&quot; triggered this thought. Why not use a Reason header to =
indicate that an INVITE is due to a REFER? We could come up with a =
Reason code for this. That way the argument that there is no way to =
tell a regular INVITE from one that has been caused by a REFER goes =
away!</FONT></P>

<P><FONT SIZE=3D2>Thx,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Robert Sparks [<A =
HREF=3D"mailto:rsparks@dynamicsoft.com">mailto:rsparks@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, April 08, 2002 9:23 AM</FONT>
<BR><FONT SIZE=3D2>To: Nigel Dewdney</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sip] REFER security options - removing =
Referred-By</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>On Mon, 2002-04-08 at 08:30, Nigel Dewdney =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; Robert,</FONT>
<BR><FONT SIZE=3D2>&gt; Inline:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Robert Sparks wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; On Fri, 2002-04-05 at 09:07, Nigel Dewdney =
wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Pardon me if I'm being obtuse, but =
how do I tell whether an INVITE is as</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; a result</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; of a REFER or not if Referred-By is =
dropped?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; That's exactly the point - you =
wouldn't.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I would argue this is a bad thing. A referral =
and an unsolicited invited are</FONT>
<BR><FONT SIZE=3D2>&gt; conceptually two different things. The endpoint =
should be aware of the</FONT>
<BR><FONT SIZE=3D2>&gt; type of process it is involved in. Whether it =
chooses to do anything</FONT>
<BR><FONT SIZE=3D2>&gt; different as a result is another =
question.</FONT>
</P>

<P><FONT SIZE=3D2>So, are you saying</FONT>
<BR><FONT SIZE=3D2>&quot;It will not be useful to have a version of =
REFER that doesn't</FONT>
<BR><FONT SIZE=3D2>&nbsp;make the triggered request distinct from an =
ordinary request.&quot;?</FONT>
</P>

<P><FONT SIZE=3D2>That's different from</FONT>
<BR><FONT SIZE=3D2>&quot;It will be useful to have a version of REFER =
that does make</FONT>
<BR><FONT SIZE=3D2>the triggered request distinct from an ordinary =
request.&quot;</FONT>
</P>

<P><FONT SIZE=3D2>I agree with the second statement. I'm not convinced =
yet on the first.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; [snip]</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the REFER mechanism</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; has been reduced into one of =
&quot;hey don't call me call them!&quot;, i.e. the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; call from A to C is</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; disassociated from the dialog between =
A and B.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; No - that's a separate problem. The kind =
of association you are</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; looking for here is what's being addressed =
through the Replaces header.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Sorry, I meant association at a functional or =
application level. It comes down</FONT>
<BR><FONT SIZE=3D2>&gt; to</FONT>
<BR><FONT SIZE=3D2>&gt; a question of granularity. Do you consider the =
referral process as one that</FONT>
<BR><FONT SIZE=3D2>&gt; just</FONT>
<BR><FONT SIZE=3D2>&gt; tells the referree where to go, or one that =
includes the point of referral?</FONT>
<BR><FONT SIZE=3D2>&gt; [snip]</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The obvious example is where C wants =
to only</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; accept calls</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; which have been vetted by B. =
Irrespective of&nbsp; security requirements,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; this requires</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; that the primitive have a reference =
to B in A's request to C. How about</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; You got your alphabet mixed up =
there.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Er, no. This is just another reference to =
Referred-By (no pun intended).</FONT>
<BR><FONT SIZE=3D2>&gt; A-&gt;B</FONT>
<BR><FONT SIZE=3D2>&gt; A&lt;-B &quot;please A, REFER to C&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; A-&gt;C</FONT>
<BR><FONT SIZE=3D2>&gt; C - &quot;hey did A go through B before =
contacting me?&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; [snip]</FONT>
</P>

<P><FONT SIZE=3D2>Ah - I was using the model in the refer-sec-options =
draft that has</FONT>
<BR><FONT SIZE=3D2>a REFER going from A to B. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; There are _clearly_ good applications =
waiting for the Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; functionality. I am _NOT_ advocating we =
abandon providing it. What</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I'm asking is whether or not the base =
level transfer application</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; really needs it. There are already fielded =
implementations that don't</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; use it providing some evidence that you =
have a useful service without</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; it.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I didn't think for a moment that you were! :~) =
You could certainly achieve</FONT>
<BR><FONT SIZE=3D2>&gt; some base functionality without Referred-By, =
but then you could apply</FONT>
<BR><FONT SIZE=3D2>&gt; the same argument to other SIP methods. You =
could strip out some of the</FONT>
<BR><FONT SIZE=3D2>&gt; headers in INVITE and still get some base =
functionality, but I don't think</FONT>
<BR><FONT SIZE=3D2>&gt; anyone is arguing for that. You could remove =
the From header for example,</FONT>
<BR><FONT SIZE=3D2>&gt; as has already been alluded to on this thread. =
Its another point of</FONT>
<BR><FONT SIZE=3D2>&gt; granularity -</FONT>
<BR><FONT SIZE=3D2>&gt; you support a lot more with it than without. =
You can ignore the header if</FONT>
<BR><FONT SIZE=3D2>&gt; your application doesn't require it (treat it =
like the From: header).</FONT>
</P>

<P><FONT SIZE=3D2>Well, the same argument _did_ get applied to other =
SIP methods as they</FONT>
<BR><FONT SIZE=3D2>were being defined. What we're left with is what we =
could agree on being</FONT>
<BR><FONT SIZE=3D2>the minimal set of required headers. Of course, the =
bar was in a little</FONT>
<BR><FONT SIZE=3D2>different place before we had an agreed on =
Supported/Require mechanism.</FONT>
</P>

<P><FONT SIZE=3D2>Your last sentence is nearing the root of the problem =
though.</FONT>
<BR><FONT SIZE=3D2>Are you claiming that its OK to provide Referred-By =
and not secure it?</FONT>
<BR><FONT SIZE=3D2>(where secure it means A provides a proof =
&quot;X&quot; to C through some</FONT>
<BR><FONT SIZE=3D2>mechanism. X is a proof of at least A's identity, =
the content of the</FONT>
<BR><FONT SIZE=3D2>Refer-To header A sends to B. There may be more =
stuff in the proof to</FONT>
<BR><FONT SIZE=3D2>help defend against replay).</FONT>
</P>

<P><FONT SIZE=3D2>Are you claiming that it is sufficient to say in the =
draft that</FONT>
<BR><FONT SIZE=3D2>an implementation receiving any request (other than =
a REFER) with a</FONT>
<BR><FONT SIZE=3D2>Referred-By header and no such proof SHOULD ignore =
the header entirely</FONT>
<BR><FONT SIZE=3D2>while processing the request? (This is analogous to =
saying an </FONT>
<BR><FONT SIZE=3D2>implementation SHOULD NOT display the contents of =
the From header</FONT>
<BR><FONT SIZE=3D2>while the phone is alerting unless the UA has some =
proof of the</FONT>
<BR><FONT SIZE=3D2>contents of that From header).</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; regards,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Nigel Dewdney</FONT>
<BR><FONT SIZE=3D2>&gt; Laboratory Telecommunication Sciences</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>
<BR>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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From daemon@ns.ietf.org  Mon Apr  8 15:56:00 2002
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Subject: RE: [Sip] REFER security options - removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Sriram Parameswar <sriramp@nortelnetworks.com>
Cc: Nigel Dewdney <njdewdn@lts.ncsc.mil>, sip@ietf.org
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I think the people that want to know the INVITE was because of a REFER
are interested in more than just that one bit of information - they
are really wanting to know who sent the REFER and what that entity
actually referred to. I don't know if we could reasonably code that
into Reason:

RjS
On Mon, 2002-04-08 at 14:06, Sriram Parameswar wrote:
> Hi: 
> 
> The thread "Reason format: removing reason codes" triggered this
> thought. Why not use a Reason header to indicate that an INVITE is due
> to a REFER? We could come up with a Reason code for this. That way the
> argument that there is no way to tell a regular INVITE from one that has
> been caused by a REFER goes away!
> 
> Thx, 
> 
> Sriram 
> 
> __________________________________________ 
> Sriram Parameswar              Phone: 972-685-8540 
> Interactive Multimedia Server (IMS) Fax: 972-684-3986 
> Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 
> 
> 
> -----Original Message----- 
> From: Robert Sparks [ mailto:rsparks@dynamicsoft.com
> <mailto:rsparks@dynamicsoft.com> ] 
> Sent: Monday, April 08, 2002 9:23 AM 
> To: Nigel Dewdney 
> Cc: sip@ietf.org 
> Subject: Re: [Sip] REFER security options - removing Referred-By 
> 
> 
> On Mon, 2002-04-08 at 08:30, Nigel Dewdney wrote: 
> > Robert, 
> > Inline: 
> > 
> > Robert Sparks wrote: 
> > 
> > > On Fri, 2002-04-05 at 09:07, Nigel Dewdney wrote: 
> > > > Pardon me if I'm being obtuse, but how do I tell whether an INVITE
> is as 
> > > > a result 
> > > > of a REFER or not if Referred-By is dropped? 
> > > 
> > > That's exactly the point - you wouldn't. 
> > > 
> > 
> > I would argue this is a bad thing. A referral and an unsolicited
> invited are 
> > conceptually two different things. The endpoint should be aware of the
> 
> > type of process it is involved in. Whether it chooses to do anything 
> > different as a result is another question. 
> 
> So, are you saying 
> "It will not be useful to have a version of REFER that doesn't 
>  make the triggered request distinct from an ordinary request."? 
> 
> That's different from 
> "It will be useful to have a version of REFER that does make 
> the triggered request distinct from an ordinary request." 
> 
> I agree with the second statement. I'm not convinced yet on the first. 
> 
> > [snip] 
> > 
> > > > the REFER mechanism 
> > > > has been reduced into one of "hey don't call me call them!", i.e.
> the 
> > > > call from A to C is 
> > > > disassociated from the dialog between A and B. 
> > > 
> > > No - that's a separate problem. The kind of association you are 
> > > looking for here is what's being addressed through the Replaces
> header. 
> > > 
> > 
> > Sorry, I meant association at a functional or application level. It
> comes down 
> > to 
> > a question of granularity. Do you consider the referral process as one
> that 
> > just 
> > tells the referree where to go, or one that includes the point of
> referral? 
> > [snip] 
> > 
> > > > The obvious example is where C wants to only 
> > > > accept calls 
> > > > which have been vetted by B. Irrespective of  security
> requirements, 
> > > > this requires 
> > > > that the primitive have a reference to B in A's request to C. How
> about 
> > > You got your alphabet mixed up there. 
> > 
> > Er, no. This is just another reference to Referred-By (no pun
> intended). 
> > A->B 
> > A<-B "please A, REFER to C" 
> > A->C 
> > C - "hey did A go through B before contacting me?" 
> > [snip] 
> 
> Ah - I was using the model in the refer-sec-options draft that has 
> a REFER going from A to B. 
> > 
> > > 
> > > There are _clearly_ good applications waiting for the Referred-By 
> > > functionality. I am _NOT_ advocating we abandon providing it. What 
> > > I'm asking is whether or not the base level transfer application 
> > > really needs it. There are already fielded implementations that
> don't 
> > > use it providing some evidence that you have a useful service
> without 
> > > it. 
> > > 
> > 
> > I didn't think for a moment that you were! :~) You could certainly
> achieve 
> > some base functionality without Referred-By, but then you could apply 
> > the same argument to other SIP methods. You could strip out some of
> the 
> > headers in INVITE and still get some base functionality, but I don't
> think 
> > anyone is arguing for that. You could remove the From header for
> example, 
> > as has already been alluded to on this thread. Its another point of 
> > granularity - 
> > you support a lot more with it than without. You can ignore the header
> if 
> > your application doesn't require it (treat it like the From: header). 
> 
> Well, the same argument _did_ get applied to other SIP methods as they 
> were being defined. What we're left with is what we could agree on being
> 
> the minimal set of required headers. Of course, the bar was in a little 
> different place before we had an agreed on Supported/Require mechanism. 
> 
> Your last sentence is nearing the root of the problem though. 
> Are you claiming that its OK to provide Referred-By and not secure it? 
> (where secure it means A provides a proof "X" to C through some 
> mechanism. X is a proof of at least A's identity, the content of the 
> Refer-To header A sends to B. There may be more stuff in the proof to 
> help defend against replay). 
> 
> Are you claiming that it is sufficient to say in the draft that 
> an implementation receiving any request (other than a REFER) with a 
> Referred-By header and no such proof SHOULD ignore the header entirely 
> while processing the request? (This is analogous to saying an 
> implementation SHOULD NOT display the contents of the From header 
> while the phone is alerting unless the UA has some proof of the 
> contents of that From header). 
> 
> > 
> > regards, 
> > 
> > Nigel Dewdney 
> > Laboratory Telecommunication Sciences 
> > 
> > 
> 
> 
> 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> <https://www1.ietf.org/mailman/listinfo/sip>  
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> 



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To: Sriram Parameswar <sriramp@nortelnetworks.com>
CC: "'Robert Sparks'" <rsparks@dynamicsoft.com>,
        Nigel Dewdney <njdewdn@lts.ncsc.mil>, sip@ietf.org,
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Sriram Parameswar wrote:

> Hi:
> 
> The thread "Reason format: removing reason codes" triggered this 
> thought. Why not use a Reason header to indicate that an INVITE is due 
> to a REFER? We could come up with a Reason code for this. That way the 
> argument that there is no way to tell a regular INVITE from one that has

 > been caused by a REFER goes away!

I have been only peripherally following the "REFER security options -
removing Referred-By" thread, so aplogies if I have misunderstood the
thread discussion thus not quite answering your question.

The reason why the Reason header will not be able to do this is that
in the following scenario:

    A, B in a conversation
    A refers B to C
    B sends C an INVITE

The INVITE from B to C would be considered outside of a dialog as far
as C is concerned.

Best regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From: "Elwell, John" <John.Elwell@siemenscomms.co.uk>
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RFC2543bis09 section 8.1.2, last paragraph seems to imply that destinations
obtained through DNS look-up are tried sequentially ("trying each address
until a server is contacted") rather than in parallel. Also it refers to the
sip-srv draft, section 4.3 of which does not seem to suggest parallel
attempts. Can anyone confirm that attempts must not be tried in parallel?

> ---------------------------------------------------------------
> John Elwell (john.elwell@siemenscomms.co.uk)
> Siemens Communications Limited,
> Tel: +44 115 943 4989   Fax: +44 115 943 4969
> ---------------------------------------------------------------
> 

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From daemon@ns.ietf.org  Mon Apr  8 19:03:11 2002
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Date: Mon, 08 Apr 2002 17:12:48 -0500
From: Chris Martin <cmartin@dasecurenetworks.com>
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To: Mark Watson <mwatson@nortelnetworks.com>
CC: sip@ietf.org
Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy   
 draft)
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> Mark Watson wrote:
> 
> Chris,
> 
> Regarding whether these privacy features can be provided by a proxy,
> modification of the From/To fields cannot be done by a proxy according
> to the current specification. You require a device which is more
> clever than a proxy, as it has to maintain the link between the
> From/To values of the incoming dialog and those of the outgoing
> dialog, performing modification in both directions.
> 
> Otherwise it will not work with an RFC2543 client, which matches
> dialogs based on the whole From/To values. If you don't care about
> RFC2543 clients, then it would work, but you would still not be a
> proxy according to the RFC.

It will be acting totally in acordance with the RFC except for the fact
that it must modify the outbound packets and then maintain the original
data in order to convert the responses/requests back to the original
headers that the internal SIP UA will recognize as having sent. 


> 
> I am not foolish enough to mention the name for such devices, but I
> expect you can guess :-)
> 
> Regarding the To field, if A calls B and B forwards to C, then B has
> the right to ask their Service Provider for the forwarding to be
> carried out 'anonymously'. As well as precluding forwarding based on
> redirection, this would imply modification of the To field, in which
> B's identity is carried (amongst other things).


That is an interesting scenario, I'll need to digest this one.

I believe that we are seeking the same outcome here, so what can be done
to make it spec, do we really need additional headers, or do we rename
the "deveices" :^) .....create a privacy proxy....to go along with the
others   :^)

Thanks,
Chris

> 
> Regards,
> 
> Mark
> 
> > -----Original Message-----
> > From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> > Sent: 08 April 2002 13:58
> > To: Watson, Mark [MDN05:EP10:EXCH]
> > Cc: sip@ietf.org
> > Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> 
> > Privacy draft)
> >
> >
> > Mark,
> > Commments inline
> >
> > > Mark Watson wrote:
> > >
> > > Chris,
> > >
> > > I am arguing that exactly the type of service you describe
> > below will
> > > be required to be implemented by providers of public SIP services
> in
> > > order to meet Data Protection requirements.
> >
> > The features of topology hiding, could be implemented on a separate
> > element to provide for the protection of the IP addresses within the
> 
> > SDP, etc., but I think that it still doesn't require any additional
> > headers to provide this functionality.
> >
> > >
> > > In particular, the values in the To/From fields will need to be
> > > modified.
> >
> > I agree with you in respect to confidentiality and topolgy hiding
> (IP
> > modification) of the internal calling domain, but also
> > believe that the
> > privacy aspect can be performed on the SIP proxy for the From
> > Headers. I
> > also believe that the To headers, in relation to outbound calls
> > originating from the calling domain of the caller, when they relate
> to
> > the called party, outside of the calling domain, doesn't
> > really require
> > modification, in my mind as long as the From has been modified to
> > specify that the caller is calling from a private number.
> >
> > >
> > > Others are arguing that the transport of the To/From fields
> without
> > > modification is a transparent service of SIP, and that the
> > contents of
> > > these fields are entirely the responsibility of the calling UA
> i.e.
> > > the Service Provider has no responsibility for the information in
> > > these fields.
> >
> > Ya, and this is where I totally agree with you, in that it must be
> > addressed at this level by the SIP service provider.
> >
> > The reason for this is that this is where the provisioning is
> > performed
> > for customers of SIP carriers.
> >
> > Individuals can place anything they want in the >From field merely
> by
> > configuring the SIP UA to do so. But in the case of a carrier
> > much more
> > control and policy is required.
> >
> > >
> > > One important point about the service you describe below is that
> it
> > > cannot be implemented on a proxy. Something more that a proxy is
> > > required.
> >
> > See my first comment on this one.
> >
> > I have been working with many firewall vendors to assist them in
> their
> > development of both SIP enabled Firewall ALG's (SFA) and SIP enabled
> 
> > Firewall Proxies (SFP)(These are basically outbound proxies).
> > There are
> > several vendors now, which I will leave unnamed, that now support
> SIP
> > and other vendors still on the way, I will let them jump in and name
> 
> > themselves if they so desire.  These vendors may very well provide
> the
> > functionality for the capabilities that you are looking for,
> > in the very
> > near future. They have proven to do this job very nicely so
> > far, and in
> > some instance provide additional security features that will also be
> 
> > very beneficial in the SIP environment.
> >
> >
> > Thanks,
> > Chris
> > >
> > > Regards...Mark
> > >
> > > > -----Original Message-----
> > > > From: Chris Martin [mailto:cmartin@dasecurenetworks.com]
> > > > Sent: 06 April 2002 17:57
> > > > Cc: sip@ietf.org
> > > > Subject: Re: Private Info in To/From (was RE: FW: [Sip]
> > Comment, SIP
> > >
> > > > Privacy draft)
> > > >
> > > >
> > > > I have a few questions here regarding what it means and what
> > > > it is that
> > > > is desired of privacy in the relation to SIP.
> > > >
> > > > Are we asking to mask the identity of the calling party or
> provide
> > > > confidentiality or both? I am going to stick with what
> > > > appears to be the
> > > > focus so far in mind, as I read the list, which is
> > masking the true
> > > > identity. I am still digesting SIPS.
> > > >
> > > > In this case my assumption is that the topic surrounds
> > commincations
> > >
> > > > within a service provider environment/doamin not a private
> > > individuals
> > > > SIP UA and the world, and that we dont need to do anything
> between
> > > > service provider proxy and calling SIP UA, so why add an
> extension
> > > or
> > > > additional header at all? I am also going to refer to service
> > > provider
> > > > customers as enterprise customers just for this exampe, they
> > > > could just
> > > > as easily be SOHO customers.
> > > >
> > > > In the case of a service provider, if we are asking to mask the
> > > > identity, as in a private number/username that we dont want
> > > > disclosed to
> > > > the called, then it seems that it would be a bit simpler
> > to maintain
> > >
> > > > that which is curently implemented in SIP, and just develop
> > > > the privacy
> > > > mechanism as a feature on the SIP proxy, transparently to
> > the Caller
> > >
> > > > (which has been provisioned due to the desire of the
> > Caller to be a
> > > > private entity).
> > > >
> > > > For example if a SIP Proxy recieves a request from the calling
> SIP
> > > UA,
> > > > it should then act normally on the request, with the
> > > > exception that, at
> > > > this point it must modify all SIP Headers that contain
> > > > information such
> > > > as IP adresses and usernames of the caller, to
> > <private@domain.com>,
> > >
> > > > instead of caller@domain.com, and as a key to getting
> > > > responses back to
> > > > the SIP UA, the SIP Proxy must maintain the Call-ID with
> modified
> > > host
> > > > information if there is any in the Call-ID.
> > > >
> > > > This leaves incoming requests to the SIP UA's, calling
> > parties which
> > >
> > > > want to call this particular SIP UA, with privacy enabled,
> > > > will have to
> > > > be privvy to the SIP UA actual information. This implies that
> > > > a request
> > > > from a SIP UA within a SIP provider domain are really all
> > that need
> > > to
> > > > be gaurded by the service provider.
> > > >
> > > > Example Request:
> > > >
> > > >  F1 INVITE User A -> User B
> > > >
> > > >    INVITE sip:UserB@there.com SIP/2.0
> > > >    Via: SIP/2.0/UDP here.com:5060
> > > >    From: BigGuy <sip:UserA@here.com>
> > > >    To: LittleGuy <sip:UserB@there.com>
> > > >    Call-ID: 12345601@100.101.102.103
> > > >    CSeq: 1 INVITE
> > > >    Contact: <sip:UserA@100.101.102.103>
> > > >    Content-Type: application/sdp
> > > >    Content-Length: 147
> > > >
> > > >    v=0
> > > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > > >    s=Session SDP
> > > >    c=IN IP4 100.101.102.103
> > > >    t=0 0
> > > >    m=audio 49172 RTP/AVP 0
> > > >    a=rtpmap:0 PCMU/8000
> > > >
> > > >
> > > >  F1 SIP Proxy -> User B
> > > >
> > > >    INVITE sip:UserB@there.com SIP/2.0
> > > >    Via: SIP/2.0/UDP here.com:5060
> > > >    From: BigGuy <sip:Private@here.com>
> > > >    To: LittleGuy <sip:UserB@there.com>
> > > >    Call-ID: 12345601@200.200.200.200       /*SIP Proxies
> > > > public address
> > > >    CSeq: 1 INVITE
> > > >    Contact: <sip:Private@200.200.200.200>
> > > >    Content-Type: application/sdp
> > > >    Content-Length: 147
> > > >
> > > >    v=0
> > > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > > >    s=Session SDP
> > > >    c=IN IP4 100.101.102.103   /*may be actual SIP UA or B2BUA
> > > > ip address
> > > >    t=0 0
> > > >    m=audio 49172 RTP/AVP 0
> > > >    a=rtpmap:0 PCMU/8000
> > > >
> > > > The only shady area here is the SDP portion which depending on
> the
> > > > environment it is assumed may or may not be a bad thing,
> > since this
> > > > really depends on the enterprise policies at that point, and
> > > > since DHCP
> > > > or a B2BUA may be implemented in many enterprise scenarios.
> > > >
> > > > This sounds too simple to me but I wanted to know to what degree
> I
> > > am
> > > > wrong here, if I am at all.
> > > >
> > > > As for SMIME content, if this information is provided by the
> > > > user it is
> > > > then the users problem, as is the case of enterprise security,
> and
> > > SIP
> > > > UA mis-configuration, which is easily detectable within the SIP
> > > > messages. This actually could be an value added service,
> > > > detecting these
> > > > mis-configurations.  :^)
> > > >
> > > >
> > > >
> > > > Chris
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current
> sip
> > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > >
> >

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From daemon@optimus.ietf.org  Tue Apr  9 04:50:42 2002
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Date: Tue, 09 Apr 2002 11:16:39 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: "Vijay K. Gurbani" <vkg@lucent.com>
CC: sip <sip@ietf.org>
Subject: Re: [Sip] Reason format: removing reason codes
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Hi,

"Vijay K. Gurbani" wrote:
> 
> Gonzalo Camarillo wrote:
> 
> > Hello,
> 
> Gonzalo:
> 
> Comments inline.
> 
> > It has been proposed that we get rid of this reason code and use only
> > SIP status codes.
> 
> That's fine.
> 
> > 1) 155 responses. In this case we do not need anything besides the
> > status code of the final response.
> 
> 155 is probably the only response to which the Reason header field
> adds any value (183 was finally rejected after a few emails on the
> WG list).  I really think the I-D should specify that Reason header
> makes sense for 155 responses *only*.  Otherwise we will have simply
> too many ways to send responses -- normal SIP response codes, Warning
> header (which also carries information about the status of a response),
> and now, using Reason for sending a response.
> 

Yes, 155 is the only response that can carry a Reson header field. I
will update the draft as soon as I have the time.

> 
> > 3) BYE or CANCEL because no media was received.
> >
> > We do not have a status code for this. However, we could create one,
> > since it may be also useful for response.
> 
> This is tying it to a specific application.  Could we not achieve the
> same result by encapsulating a "400 No Media Received" into a Reason
> header as follows:

well, I would say that "No Media Received" is general enough to be a
status code on its own, but your point of view is also valid. I would
like to hear more opinions.


> 
>     BYE sip:mediaserver.net SIP/2.0
>     Reason: 400;text="No Media Received"
>     ...
> 
> Come to think of it, do we need the "triggered" parameter.  If there
> is not going to be a "SIP/3.0" (is there?) then why duplicate
> the SIP version already present in the R-URI?

the triggered parameter is useful if we want to encapsulate Q.850 or
other stuff like that. Anyway, that is a different discussion. Let's
first agree on the reason codes and then decide if we only carry SIP or
something else.

Regards,

Gonzalo

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From daemon@optimus.ietf.org  Tue Apr  9 11:20:00 2002
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Integration of Resource Management and SIP
	Author(s)	: W. Marshall, G. Camarillo, J. Rosenberg
	Filename	: draft-ietf-sip-manyfolks-resource-07.txt,.ps,.pdf
	Pages		: 30
	Date		: 08-Apr-02
	
This document defines a generic framework for preconditions which is
extensible through IANA registration. This document also discusses
how network quality of service can be made a precondition to
establishment of sessions initiated by the Session Initiation
Protocol (SIP). These preconditions require that the participant
reserve network resources before continuing with the session. We do
not define new quality of service reservation mechanisms; these
preconditions simply require a participant to use existing resource
reservation mechanisms before beginning the session

A URL for this Internet-Draft is:
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From daemon@optimus.ietf.org  Tue Apr  9 13:46:34 2002
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Subject: RE: [Sip] REFER security options - removing Referred-By
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Sirs,

I think there are plainty of cases where C :
-will not need to know that the call was started by A read in Referred-By
-or will trust B (i.e: intranet cases).

Protecting Referred-By also means protecting others headers.

This means that we should avoid all solutions that are not optional regarding C 
policy.

A VERIFYing approach starting at C is better suited; It allows C to decide to 
verify or not A identity.
If C does NOT want to or does NOT support VERIFY the call flows remains 
consistent with current behavior.

PS: Referred-By is a means for carrying some useful Information.

That's my 2 cents.

Best Regards,

Patrick


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CC: adam.roach@ericsson.com, "sip-ietf (E-mail)" <sip@ietf.org>
Subject: Re: [Sip] draft-ietf-sip-events-05 and Record-Route
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Brett Tate wrote:
> 
> Concerning draft-ietf-sip-events-05 and
> the Record-Route, can the record-route
> portion of the Route be updated within
> an established dialog?

No.

> 
>  (Note: Only the Contact portion can be
>  updated within an INVITE established
>  dialog.)

That contact portion (aka remote target URI) can be
updated for SUBSCRIBE created dialogs too.

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

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From daemon@optimus.ietf.org  Wed Apr 10 00:43:41 2002
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To: "Couillaud, Pierre" <pierre.couillaud@polycom.com>
CC: "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] How to register a gw pstn interface ?
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http://www.ietf.org/internet-drafts/draft-rs-trip-gw-03.txt

THis has become a work item of the iptel group.

-Jonathan R.

"Couillaud, Pierre" wrote:
> 
> Hello,
> 
> Here is my problem: given a small density gateway that support only up
> to a
> fully channelized T1 interface,
> that does not have any pretention to do PSTN routing, however that can
> be
> used to provide PSTN access to
> SIP phones sitting behind it.... How do I register my PSTN interface to
> my
> proxy ?
> 
> Can I register a route ?  And if so how ?  Can I just tell my proxy to
> forward to the gateway any E.164 address
> that it cannot resolve by itself ?  I do not find anything in SIP that
> could
> allow me to do something like this...
> Unless I am missing something here...
> 
> Thanks,
> 
> Pierre
> 
> --- Cheers,
> Pierre Couillaud, Polycom Canada Ltd.
> Phone: (604) 697 9346 or (604) 990 5415 x146
> "Anything one man can imagine, other men can make real", Jules Verne.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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

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From daemon@optimus.ietf.org  Wed Apr 10 01:02:15 2002
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To: salreyca@teleco.upv.es
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THe spec does not detail error handling for the infinite different ways
in which they might occur. The choice is at the discretion of the
implementation.

-Jonathan R.

Salva Rey Calatayud wrote:
> 
> Hi,
> 
>         how should a UA act when let's say the other endpoint doesn't
> behave compliantly to the O/A exchange model. For instance, we
> send an offer, and none of the associated messages that could
> contain an answer does (an UPDATE with a 200OK without
> answer, INVITE with offer and none of the provisional response nor
> final 200OK contain an answer )
> 
> thanks,
> Salva
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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

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From daemon@optimus.ietf.org  Wed Apr 10 01:15:25 2002
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Date: Wed, 10 Apr 2002 00:59:13 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: frank.derks@philips.com
CC: sip@ietf.org
Subject: Re: [Sip] Matching responses to transactions (error in 9th draft?)
References: <OFD6FC2362.BEF6E61D-ONC1256B87.0052C7C1@diamond.philips.com>
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frank.derks@philips.com wrote:
> 
> Section 17.1.3 of the 9th draft of the SIP specification states the
> following:
> 
> "A response that matches a transaction matched by a previous response is
> 
>  considered to be a restransmission of that response."
> 
> This statement is preceded by two "matching rules", one about a matching
> 
> branch parameter in the topmost Via header and one about a matching
> request method in the CSeq.
> 
> Suppose that multiple responses are sent as the result of an INVITE.
> E.g.
> first a 183 Trying, then a 180 Ringing and then a 200 OK. All of these
> would match with the request, but according to the above statement, the
> 180 Ringing and the 200 OK would seem to be regarded as retransmissions
> of the 183 Trying. This can surely not be the intention.

Nope. This is an error in the spec. I think the right thing is to strike
the sentence entirely. The state machines are correct as defined, based
on the matching rules as defined. Determining whether a response is a
retransmission is actually hard, and not needed. Its not just the same
response code, since the same response could come from different
downstream UA. Its not a combination of response code and tags, since
the same UA could generate multiple provisional responses of the same
type (two 183, each with a different RSeq). 

Thanks for pointing this out.

-Jonathan R.

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

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From daemon@optimus.ietf.org  Wed Apr 10 03:21:21 2002
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Date: Wed, 10 Apr 2002 07:50:45 +0100
From: "Elwell, John" <John.Elwell@siemenscomms.co.uk>
Subject: RE: [Sip] REFER security options - removing Referred-By
To: "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>,
        rsparks@dynamicsoft.com, sip@ietf.org
Message-id: <0E644D072B76C740BC1CBD2CBB51AEFE0312E51C@Beex50.siemenscomms.co.uk>
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I agree with what Patrick Mourot says. Let's have a basic refer capability
that provides an unsecured identity of A to C, and then leave it to C to
decide whether and how to verify the identity of A. This will depend on
whether it is attended or unattended transfer - with attended transfer the
call A-C will exist and will be identified by the Replaces header. This may
be sufficient, but if not, verification can be done in the context of that
dialog A-C.

 ---------------------------------------------------------------
 John Elwell (john.elwell@siemenscomms.co.uk)
 Siemens Communications Limited,
 ---------------------------------------------------------------
 Internet communications are not secure and therefore Siemens
 Communications Limited does not accept legal responsibility for the
 contents of this message. Any views or opinions presented are solely
 those of the author and do not necessarily represent those of Siemens
 Communications Limited unless otherwise specifically stated.
 


-----Original Message-----
From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
Sent: Tuesday, April 09, 2002 5:05 PM
To: rsparks@dynamicsoft.com; sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By



Sirs,

I think there are plainty of cases where C :
-will not need to know that the call was started by A read in Referred-By
-or will trust B (i.e: intranet cases).

Protecting Referred-By also means protecting others headers.

This means that we should avoid all solutions that are not optional
regarding C 
policy.

A VERIFYing approach starting at C is better suited; It allows C to decide
to 
verify or not A identity.
If C does NOT want to or does NOT support VERIFY the call flows remains 
consistent with current behavior.

PS: Referred-By is a means for carrying some useful Information.

That's my 2 cents.

Best Regards,

Patrick


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Apr 10 05:20:43 2002
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From: "Jesske, R" <R.Jesske@telekom.de>
To: cmartin@dasecurenetworks.com, mwatson@nortelnetworks.com,
        "Alexeitsev, D" <D.Alexeitsev@telekom.de>
Cc: sip@ietf.org, "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>,
        "Tolksdorf, Christian" <Christian.Tolksdorf@telekom.de>,
        "Pinker, Gerold" <Gerold.Pinker@telekom.de>, Robert.Benz@telekom.de
Subject: AW: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy    draft)
Date: Wed, 10 Apr 2002 10:38:33 +0200
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------_=_NextPart_000_01C1E06B.19184CF0
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Dear Chris and Mark,

regarding your discussion on privacy the privacy draft I have some =
comments:

1. From the Service Provider Point of view the mechanisms described =
within the privacy draft are needed. It is important to have a =
statement from the caller if he wants to hide his identity by the =
callee. There are many call scenarios where I would like to see this =
kind of privacy including the case of the interworking with a PSTN =
network, because there it is needed to have the clear statement if the =
call needs privacy or not.

2. For regulation, billing and other purposes the service provider =
needs an indication if the address included in the Invite is trusted or =
untrusted (screen =3D Yes/No). I know Chris does not like overregulated =
networks, but I think we need this within the SIP domain and in the =
PSTN interwoking case.=20

3. I don't know if a "privacy proxy" will solve all problems concerning =
privacy and trusted addresses/entities. I support the proposal of the =
privacy draft concerning the trust relations and the additional =
headers.

Of course the interworking with PSTN networks is not the only =
application (and of course not the most important) of the privacy =
draft. But for the Service Provider, which will have hybrid networks =
(e.g. SIP and PSTN), the privacy draft gives the tool to solve many =
problems at the network boundary.


Best Regards

Roland Jesske

Deutsche Telekom AG
Zentralbereich Netzinfrastruktur (ZB NI)
Section NI196; Signalling, Gateways and Switching Systems
Roland Jesske, NI196a12
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940
Fax:    +49 6151 83-4577
mailto: r.jesske@telekom.de



> -----Urspr=FCngliche Nachricht-----
> Von: Chris Martin [ mailto:cmartin@dasecurenetworks.com =
<mailto:cmartin@dasecurenetworks.com> ]
> Gesendet am: Dienstag, 9. April 2002 00:13
> An: Mark Watson
> Cc: sip@ietf.org
> Betreff: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> Privacy draft)
>
> > Mark Watson wrote:
> >
> > Chris,
> >
> > Regarding whether these privacy features can be provided by a =
proxy,
> > modification of the From/To fields cannot be done by a
> proxy according
> > to the current specification. You require a device which is more
> > clever than a proxy, as it has to maintain the link between the
> > From/To values of the incoming dialog and those of the outgoing
> > dialog, performing modification in both directions.
> >
> > Otherwise it will not work with an RFC2543 client, which matches
> > dialogs based on the whole From/To values. If you don't care about
> > RFC2543 clients, then it would work, but you would still not be a
> > proxy according to the RFC.
>
> It will be acting totally in acordance with the RFC except
> for the fact
> that it must modify the outbound packets and then maintain
> the original
> data in order to convert the responses/requests back to the original
> headers that the internal SIP UA will recognize as having sent.
>
>
> >
> > I am not foolish enough to mention the name for such devices, but I
> > expect you can guess :-)
> >
> > Regarding the To field, if A calls B and B forwards to C, then B =
has
> > the right to ask their Service Provider for the forwarding to be
> > carried out 'anonymously'. As well as precluding forwarding based =
on
> > redirection, this would imply modification of the To field, in =
which
> > B's identity is carried (amongst other things).
>
>
> That is an interesting scenario, I'll need to digest this one.
>
> I believe that we are seeking the same outcome here, so what
> can be done
> to make it spec, do we really need additional headers, or do we =
rename
> the "deveices" :^) .....create a privacy proxy....to go along with =
the
> others   :^)
>
> Thanks,
> Chris
>
> >
> > Regards,
> >
> > Mark
> >
> > > -----Original Message-----
> > > From: Chris Martin [ mailto:cmartin@dasecurenetworks.com =
<mailto:cmartin@dasecurenetworks.com> ]
> > > Sent: 08 April 2002 13:58
> > > To: Watson, Mark [MDN05:EP10:EXCH]
> > > Cc: sip@ietf.org
> > > Subject: Re: Private Info in To/From (was RE: FW: [Sip]
> Comment, SIP
> >
> > > Privacy draft)
> > >
> > >
> > > Mark,
> > > Commments inline
> > >
> > > > Mark Watson wrote:
> > > >
> > > > Chris,
> > > >
> > > > I am arguing that exactly the type of service you describe
> > > below will
> > > > be required to be implemented by providers of public
> SIP services
> > in
> > > > order to meet Data Protection requirements.
> > >
> > > The features of topology hiding, could be implemented on
> a separate
> > > element to provide for the protection of the IP addresses
> within the
> >
> > > SDP, etc., but I think that it still doesn't require any
> additional
> > > headers to provide this functionality.
> > >
> > > >
> > > > In particular, the values in the To/From fields will need to be
> > > > modified.
> > >
> > > I agree with you in respect to confidentiality and topolgy hiding
> > (IP
> > > modification) of the internal calling domain, but also
> > > believe that the
> > > privacy aspect can be performed on the SIP proxy for the From
> > > Headers. I
> > > also believe that the To headers, in relation to outbound calls
> > > originating from the calling domain of the caller, when
> they relate
> > to
> > > the called party, outside of the calling domain, doesn't
> > > really require
> > > modification, in my mind as long as the From has been modified to
> > > specify that the caller is calling from a private number.
> > >
> > > >
> > > > Others are arguing that the transport of the To/From fields
> > without
> > > > modification is a transparent service of SIP, and that the
> > > contents of
> > > > these fields are entirely the responsibility of the calling UA
> > i.e.
> > > > the Service Provider has no responsibility for the
> information in
> > > > these fields.
> > >
> > > Ya, and this is where I totally agree with you, in that it must =
be
> > > addressed at this level by the SIP service provider.
> > >
> > > The reason for this is that this is where the provisioning is
> > > performed
> > > for customers of SIP carriers.
> > >
> > > Individuals can place anything they want in the >From field =
merely
> > by
> > > configuring the SIP UA to do so. But in the case of a carrier
> > > much more
> > > control and policy is required.
> > >
> > > >
> > > > One important point about the service you describe below is =
that
> > it
> > > > cannot be implemented on a proxy. Something more that a proxy =
is
> > > > required.
> > >
> > > See my first comment on this one.
> > >
> > > I have been working with many firewall vendors to assist them in
> > their
> > > development of both SIP enabled Firewall ALG's (SFA) and
> SIP enabled
> >
> > > Firewall Proxies (SFP)(These are basically outbound proxies).
> > > There are
> > > several vendors now, which I will leave unnamed, that now support
> > SIP
> > > and other vendors still on the way, I will let them jump
> in and name
> >
> > > themselves if they so desire.  These vendors may very well =
provide
> > the
> > > functionality for the capabilities that you are looking for,
> > > in the very
> > > near future. They have proven to do this job very nicely so
> > > far, and in
> > > some instance provide additional security features that
> will also be
> >
> > > very beneficial in the SIP environment.
> > >
> > >
> > > Thanks,
> > > Chris
> > > >
> > > > Regards...Mark
> > > >
> > > > > -----Original Message-----
> > > > > From: Chris Martin [ mailto:cmartin@dasecurenetworks.com =
<mailto:cmartin@dasecurenetworks.com> ]
> > > > > Sent: 06 April 2002 17:57
> > > > > Cc: sip@ietf.org
> > > > > Subject: Re: Private Info in To/From (was RE: FW: [Sip]
> > > Comment, SIP
> > > >
> > > > > Privacy draft)
> > > > >
> > > > >
> > > > > I have a few questions here regarding what it means and what
> > > > > it is that
> > > > > is desired of privacy in the relation to SIP.
> > > > >
> > > > > Are we asking to mask the identity of the calling party or
> > provide
> > > > > confidentiality or both? I am going to stick with what
> > > > > appears to be the
> > > > > focus so far in mind, as I read the list, which is
> > > masking the true
> > > > > identity. I am still digesting SIPS.
> > > > >
> > > > > In this case my assumption is that the topic surrounds
> > > commincations
> > > >
> > > > > within a service provider environment/doamin not a private
> > > > individuals
> > > > > SIP UA and the world, and that we dont need to do anything
> > between
> > > > > service provider proxy and calling SIP UA, so why add an
> > extension
> > > > or
> > > > > additional header at all? I am also going to refer to service
> > > > provider
> > > > > customers as enterprise customers just for this exampe, they
> > > > > could just
> > > > > as easily be SOHO customers.
> > > > >
> > > > > In the case of a service provider, if we are asking
> to mask the
> > > > > identity, as in a private number/username that we dont want
> > > > > disclosed to
> > > > > the called, then it seems that it would be a bit simpler
> > > to maintain
> > > >
> > > > > that which is curently implemented in SIP, and just develop
> > > > > the privacy
> > > > > mechanism as a feature on the SIP proxy, transparently to
> > > the Caller
> > > >
> > > > > (which has been provisioned due to the desire of the
> > > Caller to be a
> > > > > private entity).
> > > > >
> > > > > For example if a SIP Proxy recieves a request from the =
calling
> > SIP
> > > > UA,
> > > > > it should then act normally on the request, with the
> > > > > exception that, at
> > > > > this point it must modify all SIP Headers that contain
> > > > > information such
> > > > > as IP adresses and usernames of the caller, to
> > > <private@domain.com>,
> > > >
> > > > > instead of caller@domain.com, and as a key to getting
> > > > > responses back to
> > > > > the SIP UA, the SIP Proxy must maintain the Call-ID with
> > modified
> > > > host
> > > > > information if there is any in the Call-ID.
> > > > >
> > > > > This leaves incoming requests to the SIP UA's, calling
> > > parties which
> > > >
> > > > > want to call this particular SIP UA, with privacy enabled,
> > > > > will have to
> > > > > be privvy to the SIP UA actual information. This implies that
> > > > > a request
> > > > > from a SIP UA within a SIP provider domain are really all
> > > that need
> > > > to
> > > > > be gaurded by the service provider.
> > > > >
> > > > > Example Request:
> > > > >
> > > > >  F1 INVITE User A -> User B
> > > > >
> > > > >    INVITE sip:UserB@there.com SIP/2.0
> > > > >    Via: SIP/2.0/UDP here.com:5060
> > > > >    From: BigGuy <sip:UserA@here.com>
> > > > >    To: LittleGuy <sip:UserB@there.com>
> > > > >    Call-ID: 12345601@100.101.102.103
> > > > >    CSeq: 1 INVITE
> > > > >    Contact: <sip:UserA@100.101.102.103>
> > > > >    Content-Type: application/sdp
> > > > >    Content-Length: 147
> > > > >
> > > > >    v=3D0
> > > > >    o=3DUserA 2890844526 2890844526 IN IP4 here.com
> > > > >    s=3DSession SDP
> > > > >    c=3DIN IP4 100.101.102.103
> > > > >    t=3D0 0
> > > > >    m=3Daudio 49172 RTP/AVP 0
> > > > >    a=3Drtpmap:0 PCMU/8000
> > > > >
> > > > >
> > > > >  F1 SIP Proxy -> User B
> > > > >
> > > > >    INVITE sip:UserB@there.com SIP/2.0
> > > > >    Via: SIP/2.0/UDP here.com:5060
> > > > >    From: BigGuy <sip:Private@here.com>
> > > > >    To: LittleGuy <sip:UserB@there.com>
> > > > >    Call-ID: 12345601@200.200.200.200       /*SIP Proxies
> > > > > public address
> > > > >    CSeq: 1 INVITE
> > > > >    Contact: <sip:Private@200.200.200.200>
> > > > >    Content-Type: application/sdp
> > > > >    Content-Length: 147
> > > > >
> > > > >    v=3D0
> > > > >    o=3DUserA 2890844526 2890844526 IN IP4 here.com
> > > > >    s=3DSession SDP
> > > > >    c=3DIN IP4 100.101.102.103   /*may be actual SIP UA or =
B2BUA
> > > > > ip address
> > > > >    t=3D0 0
> > > > >    m=3Daudio 49172 RTP/AVP 0
> > > > >    a=3Drtpmap:0 PCMU/8000
> > > > >
> > > > > The only shady area here is the SDP portion which depending =
on
> > the
> > > > > environment it is assumed may or may not be a bad thing,
> > > since this
> > > > > really depends on the enterprise policies at that point, and
> > > > > since DHCP
> > > > > or a B2BUA may be implemented in many enterprise scenarios.
> > > > >
> > > > > This sounds too simple to me but I wanted to know to
> what degree
> > I
> > > > am
> > > > > wrong here, if I am at all.
> > > > >
> > > > > As for SMIME content, if this information is provided by the
> > > > > user it is
> > > > > then the users problem, as is the case of enterprise =
security,
> > and
> > > > SIP
> > > > > UA mis-configuration, which is easily detectable
> within the SIP
> > > > > messages. This actually could be an value added service,
> > > > > detecting these
> > > > > mis-configurations.  :^)
> > > > >
> > > > >
> > > > >
> > > > > Chris
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip =
<https://www1.ietf.org/mailman/listinfo/sip>=20
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on current
> > sip
> > > > > Use sipping@ietf.org for new developments on the
> > > application of sip
> > > > >
> > >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip =
<https://www1.ietf.org/mailman/listinfo/sip>=20
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Wed, 10 Apr 2002 07:34:50 -0500
From: Chris Martin <cmartin@dasecurenetworks.com>
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Subject: Re: AW: Private Info in To/From (was RE: FW: [Sip] Comment, SIP 
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"Jesske, R" wrote:
> 
> Dear Chris and Mark,
> 
> regarding your discussion on privacy the privacy draft I have some comments:
> 
> 1. From the Service Provider Point of view the mechanisms described within the privacy draft are needed. It is important to have a statement from the caller if he wants to hide his identity by the callee. There are many call scenarios where I would like to see this kind of privacy including the case of the interworking with a PSTN network, because there it is needed to have the clear statement if the call needs privacy or not.

Actually I do believe that the mechanisms are required and fully support
the direction of the privacy draft. I guess what I am looking for is a
method to reduce the number of mechanisms within the protocol. Right now
I am focusing on the mechanism that basically conveys the identity of
the caller to the remote party as "private".  In the case of a service
provider or even a SIP UA for that matter, I dont see the need to place
the mechanism in the protocol, when it seems that all we need is a local
mechanism on the SIP UA itself or a provisioning option on the SIP UAS
to modify the From to "private". I am just wondering if every aspect of
SIP communications needs to be conveyed by a SIP message.

> 
> 2. For regulation, billing and other purposes the service provider needs an indication if the address included in the Invite is trusted or untrusted (screen = Yes/No). I know Chris does not like overregulated networks, but I think we need this within the SIP domain and in the PSTN interwoking case.
> 

I never said that I was against unregulated networks, they are and will
be a reality. I actually have been concerned that the regulation aspect
is being ignored, by the very fact that everyone looks at firewalls as
an evil in an IP network. It may be firewalls, SIP enabled, that provide
portions fo calling confidentiality, topology hiding, etc., and I
believe that these will aid in meeting future regulations.

> 3. I don't know if a "privacy proxy" will solve all problems concerning privacy and trusted addresses/entities. I support the proposal of the privacy draft concerning the trust relations and the additional headers.
> 

Was just kidding about privacy proxy. 

Would just like to look into the need of additional headers. As for
trust relations in a carrier network...this may be a matter of
provisioning on SIP UAS.

> Of course the interworking with PSTN networks is not the only application (and of course not the most important) of the privacy draft. But for the Service Provider, which will have hybrid networks (e.g. SIP and PSTN), the privacy draft gives the tool to solve many problems at the network boundary.
> 

Totally agree. This is a tool not only addressing future reguations at
the carrier level, but also at the enterprise boundaries.

Thanks for the excellent feedback, 
Chris

> Best Regards
> 
> Roland Jesske
> 
> Deutsche Telekom AG
> Zentralbereich Netzinfrastruktur (ZB NI)
> Section NI196; Signalling, Gateways and Switching Systems
> Roland Jesske, NI196a12
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940
> Fax:    +49 6151 83-4577
> mailto: r.jesske@telekom.de
> 
> > -----Ursprüngliche Nachricht-----
> > Von: Chris Martin [ mailto:cmartin@dasecurenetworks.com <mailto:cmartin@dasecurenetworks.com> ]
> > Gesendet am: Dienstag, 9. April 2002 00:13
> > An: Mark Watson
> > Cc: sip@ietf.org
> > Betreff: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP
> > Privacy draft)
> >
> > > Mark Watson wrote:
> > >
> > > Chris,
> > >
> > > Regarding whether these privacy features can be provided by a proxy,
> > > modification of the From/To fields cannot be done by a
> > proxy according
> > > to the current specification. You require a device which is more
> > > clever than a proxy, as it has to maintain the link between the
> > > From/To values of the incoming dialog and those of the outgoing
> > > dialog, performing modification in both directions.
> > >
> > > Otherwise it will not work with an RFC2543 client, which matches
> > > dialogs based on the whole From/To values. If you don't care about
> > > RFC2543 clients, then it would work, but you would still not be a
> > > proxy according to the RFC.
> >
> > It will be acting totally in acordance with the RFC except
> > for the fact
> > that it must modify the outbound packets and then maintain
> > the original
> > data in order to convert the responses/requests back to the original
> > headers that the internal SIP UA will recognize as having sent.
> >
> >
> > >
> > > I am not foolish enough to mention the name for such devices, but I
> > > expect you can guess :-)
> > >
> > > Regarding the To field, if A calls B and B forwards to C, then B has
> > > the right to ask their Service Provider for the forwarding to be
> > > carried out 'anonymously'. As well as precluding forwarding based on
> > > redirection, this would imply modification of the To field, in which
> > > B's identity is carried (amongst other things).
> >
> >
> > That is an interesting scenario, I'll need to digest this one.
> >
> > I believe that we are seeking the same outcome here, so what
> > can be done
> > to make it spec, do we really need additional headers, or do we rename
> > the "deveices" :^) .....create a privacy proxy....to go along with the
> > others   :^)
> >
> > Thanks,
> > Chris
> >
> > >
> > > Regards,
> > >
> > > Mark
> > >
> > > > -----Original Message-----
> > > > From: Chris Martin [ mailto:cmartin@dasecurenetworks.com <mailto:cmartin@dasecurenetworks.com> ]
> > > > Sent: 08 April 2002 13:58
> > > > To: Watson, Mark [MDN05:EP10:EXCH]
> > > > Cc: sip@ietf.org
> > > > Subject: Re: Private Info in To/From (was RE: FW: [Sip]
> > Comment, SIP
> > >
> > > > Privacy draft)
> > > >
> > > >
> > > > Mark,
> > > > Commments inline
> > > >
> > > > > Mark Watson wrote:
> > > > >
> > > > > Chris,
> > > > >
> > > > > I am arguing that exactly the type of service you describe
> > > > below will
> > > > > be required to be implemented by providers of public
> > SIP services
> > > in
> > > > > order to meet Data Protection requirements.
> > > >
> > > > The features of topology hiding, could be implemented on
> > a separate
> > > > element to provide for the protection of the IP addresses
> > within the
> > >
> > > > SDP, etc., but I think that it still doesn't require any
> > additional
> > > > headers to provide this functionality.
> > > >
> > > > >
> > > > > In particular, the values in the To/From fields will need to be
> > > > > modified.
> > > >
> > > > I agree with you in respect to confidentiality and topolgy hiding
> > > (IP
> > > > modification) of the internal calling domain, but also
> > > > believe that the
> > > > privacy aspect can be performed on the SIP proxy for the From
> > > > Headers. I
> > > > also believe that the To headers, in relation to outbound calls
> > > > originating from the calling domain of the caller, when
> > they relate
> > > to
> > > > the called party, outside of the calling domain, doesn't
> > > > really require
> > > > modification, in my mind as long as the From has been modified to
> > > > specify that the caller is calling from a private number.
> > > >
> > > > >
> > > > > Others are arguing that the transport of the To/From fields
> > > without
> > > > > modification is a transparent service of SIP, and that the
> > > > contents of
> > > > > these fields are entirely the responsibility of the calling UA
> > > i.e.
> > > > > the Service Provider has no responsibility for the
> > information in
> > > > > these fields.
> > > >
> > > > Ya, and this is where I totally agree with you, in that it must be
> > > > addressed at this level by the SIP service provider.
> > > >
> > > > The reason for this is that this is where the provisioning is
> > > > performed
> > > > for customers of SIP carriers.
> > > >
> > > > Individuals can place anything they want in the >From field merely
> > > by
> > > > configuring the SIP UA to do so. But in the case of a carrier
> > > > much more
> > > > control and policy is required.
> > > >
> > > > >
> > > > > One important point about the service you describe below is that
> > > it
> > > > > cannot be implemented on a proxy. Something more that a proxy is
> > > > > required.
> > > >
> > > > See my first comment on this one.
> > > >
> > > > I have been working with many firewall vendors to assist them in
> > > their
> > > > development of both SIP enabled Firewall ALG's (SFA) and
> > SIP enabled
> > >
> > > > Firewall Proxies (SFP)(These are basically outbound proxies).
> > > > There are
> > > > several vendors now, which I will leave unnamed, that now support
> > > SIP
> > > > and other vendors still on the way, I will let them jump
> > in and name
> > >
> > > > themselves if they so desire.  These vendors may very well provide
> > > the
> > > > functionality for the capabilities that you are looking for,
> > > > in the very
> > > > near future. They have proven to do this job very nicely so
> > > > far, and in
> > > > some instance provide additional security features that
> > will also be
> > >
> > > > very beneficial in the SIP environment.
> > > >
> > > >
> > > > Thanks,
> > > > Chris
> > > > >
> > > > > Regards...Mark
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Chris Martin [ mailto:cmartin@dasecurenetworks.com <mailto:cmartin@dasecurenetworks.com> ]
> > > > > > Sent: 06 April 2002 17:57
> > > > > > Cc: sip@ietf.org
> > > > > > Subject: Re: Private Info in To/From (was RE: FW: [Sip]
> > > > Comment, SIP
> > > > >
> > > > > > Privacy draft)
> > > > > >
> > > > > >
> > > > > > I have a few questions here regarding what it means and what
> > > > > > it is that
> > > > > > is desired of privacy in the relation to SIP.
> > > > > >
> > > > > > Are we asking to mask the identity of the calling party or
> > > provide
> > > > > > confidentiality or both? I am going to stick with what
> > > > > > appears to be the
> > > > > > focus so far in mind, as I read the list, which is
> > > > masking the true
> > > > > > identity. I am still digesting SIPS.
> > > > > >
> > > > > > In this case my assumption is that the topic surrounds
> > > > commincations
> > > > >
> > > > > > within a service provider environment/doamin not a private
> > > > > individuals
> > > > > > SIP UA and the world, and that we dont need to do anything
> > > between
> > > > > > service provider proxy and calling SIP UA, so why add an
> > > extension
> > > > > or
> > > > > > additional header at all? I am also going to refer to service
> > > > > provider
> > > > > > customers as enterprise customers just for this exampe, they
> > > > > > could just
> > > > > > as easily be SOHO customers.
> > > > > >
> > > > > > In the case of a service provider, if we are asking
> > to mask the
> > > > > > identity, as in a private number/username that we dont want
> > > > > > disclosed to
> > > > > > the called, then it seems that it would be a bit simpler
> > > > to maintain
> > > > >
> > > > > > that which is curently implemented in SIP, and just develop
> > > > > > the privacy
> > > > > > mechanism as a feature on the SIP proxy, transparently to
> > > > the Caller
> > > > >
> > > > > > (which has been provisioned due to the desire of the
> > > > Caller to be a
> > > > > > private entity).
> > > > > >
> > > > > > For example if a SIP Proxy recieves a request from the calling
> > > SIP
> > > > > UA,
> > > > > > it should then act normally on the request, with the
> > > > > > exception that, at
> > > > > > this point it must modify all SIP Headers that contain
> > > > > > information such
> > > > > > as IP adresses and usernames of the caller, to
> > > > <private@domain.com>,
> > > > >
> > > > > > instead of caller@domain.com, and as a key to getting
> > > > > > responses back to
> > > > > > the SIP UA, the SIP Proxy must maintain the Call-ID with
> > > modified
> > > > > host
> > > > > > information if there is any in the Call-ID.
> > > > > >
> > > > > > This leaves incoming requests to the SIP UA's, calling
> > > > parties which
> > > > >
> > > > > > want to call this particular SIP UA, with privacy enabled,
> > > > > > will have to
> > > > > > be privvy to the SIP UA actual information. This implies that
> > > > > > a request
> > > > > > from a SIP UA within a SIP provider domain are really all
> > > > that need
> > > > > to
> > > > > > be gaurded by the service provider.
> > > > > >
> > > > > > Example Request:
> > > > > >
> > > > > >  F1 INVITE User A -> User B
> > > > > >
> > > > > >    INVITE sip:UserB@there.com SIP/2.0
> > > > > >    Via: SIP/2.0/UDP here.com:5060
> > > > > >    From: BigGuy <sip:UserA@here.com>
> > > > > >    To: LittleGuy <sip:UserB@there.com>
> > > > > >    Call-ID: 12345601@100.101.102.103
> > > > > >    CSeq: 1 INVITE
> > > > > >    Contact: <sip:UserA@100.101.102.103>
> > > > > >    Content-Type: application/sdp
> > > > > >    Content-Length: 147
> > > > > >
> > > > > >    v=0
> > > > > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > > > > >    s=Session SDP
> > > > > >    c=IN IP4 100.101.102.103
> > > > > >    t=0 0
> > > > > >    m=audio 49172 RTP/AVP 0
> > > > > >    a=rtpmap:0 PCMU/8000
> > > > > >
> > > > > >
> > > > > >  F1 SIP Proxy -> User B
> > > > > >
> > > > > >    INVITE sip:UserB@there.com SIP/2.0
> > > > > >    Via: SIP/2.0/UDP here.com:5060
> > > > > >    From: BigGuy <sip:Private@here.com>
> > > > > >    To: LittleGuy <sip:UserB@there.com>
> > > > > >    Call-ID: 12345601@200.200.200.200       /*SIP Proxies
> > > > > > public address
> > > > > >    CSeq: 1 INVITE
> > > > > >    Contact: <sip:Private@200.200.200.200>
> > > > > >    Content-Type: application/sdp
> > > > > >    Content-Length: 147
> > > > > >
> > > > > >    v=0
> > > > > >    o=UserA 2890844526 2890844526 IN IP4 here.com
> > > > > >    s=Session SDP
> > > > > >    c=IN IP4 100.101.102.103   /*may be actual SIP UA or B2BUA
> > > > > > ip address
> > > > > >    t=0 0
> > > > > >    m=audio 49172 RTP/AVP 0
> > > > > >    a=rtpmap:0 PCMU/8000
> > > > > >
> > > > > > The only shady area here is the SDP portion which depending on
> > > the
> > > > > > environment it is assumed may or may not be a bad thing,
> > > > since this
> > > > > > really depends on the enterprise policies at that point, and
> > > > > > since DHCP
> > > > > > or a B2BUA may be implemented in many enterprise scenarios.
> > > > > >
> > > > > > This sounds too simple to me but I wanted to know to
> > what degree
> > > I
> > > > > am
> > > > > > wrong here, if I am at all.
> > > > > >
> > > > > > As for SMIME content, if this information is provided by the
> > > > > > user it is
> > > > > > then the users problem, as is the case of enterprise security,
> > > and
> > > > > SIP
> > > > > > UA mis-configuration, which is easily detectable
> > within the SIP
> > > > > > messages. This actually could be an value added service,
> > > > > > detecting these
> > > > > > mis-configurations.  :^)
> > > > > >
> > > > > >
> > > > > >
> > > > > > Chris
> > > > > >
> > > > > > _______________________________________________
> > > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip <https://www1.ietf.org/mailman/listinfo/sip>
> > > > > > This list is for NEW development of the core SIP Protocol
> > > > > > Use sip-implementors@cs.columbia.edu for questions on current
> > > sip
> > > > > > Use sipping@ietf.org for new developments on the
> > > > application of sip
> > > > > >
> > > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip <https://www1.ietf.org/mailman/listinfo/sip>
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Fuxbruner, Amihay" <Amihay_Fuxbruner@icomverse.com>
To: sip@ietf.org
Cc: "Ivanov, Elena" <Elena_Ivanov@icomverse.com>
Subject: [Sip] Question: Timer for Proceeding
Date: Wed, 10 Apr 2002 17:17:48 +0300
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Hi,

Bis 09 section 17.1.1.2 does not include timer for proceeding state.
The protocol also says one SHOULD NOT retransmit in proceeding state.
Should we need a timer or is it application decision ?

Thanks,

Amihay Fuxbruner
System Engineering - Signaling R&D
Comverse

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<P><FONT SIZE=2>Hi,</FONT>
</P>

<P><FONT SIZE=2>Bis 09 section 17.1.1.2 does not include timer for proceeding state.</FONT>
<BR><FONT SIZE=2>The protocol also says one SHOULD NOT retransmit in proceeding state.</FONT>
<BR><FONT SIZE=2>Should we need a timer or is it application decision ?</FONT>
</P>

<P><FONT SIZE=2>Thanks,</FONT>
</P>

<P><FONT SIZE=2>Amihay Fuxbruner</FONT>
<BR><FONT SIZE=2>System Engineering - Signaling R&amp;D</FONT>
<BR><FONT SIZE=2>Comverse</FONT>
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Hi Amihay,
     The client transaction does start timer C
when sending the INVITE request. This timer is
reset on receiving any provisional response. On
timeout, the client transaction moves to Terminated
state.

     In bis-05, this was described in Section 17,
and also indicated in the "INVITE client transaction"
diagram. But in bis--9, this description has been moved
to Sec 16.6, bullet 11. As an aside, it seems to have been
missed out in the INVITE client transaction diagram....


Subhash Nayak.
Hughes Software Systems.
http://www.hssworld.com






"Fuxbruner, Amihay" <Amihay_Fuxbruner@icomverse.com> on 04/10/2002 07:47:48
PM

To:   sip@ietf.org
cc:   "Ivanov, Elena" <Elena_Ivanov@icomverse.com> (bcc: Subhash Ullal
      Nayak/HSSBLR)

Subject:  [Sip] Question: Timer for Proceeding




Hi,

Bis 09 section 17.1.1.2 does not include timer for proceeding state.
The protocol also says one SHOULD NOT retransmit in proceeding state.
Should we need a timer or is it application decision ?

Thanks,

Amihay Fuxbruner
System Engineering - Signaling R&D
Comverse


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From daemon@optimus.ietf.org  Wed Apr 10 12:05:50 2002
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From: Alexandre Charest <acharest@mediatrix.com>
To: "'sunayak@hss.hns.com'" <sunayak@hss.hns.com>,
        "Fuxbruner, Amihay"
	 <Amihay_Fuxbruner@icomverse.com>
Cc: sip@ietf.org
Subject: RE: [Sip] Question: Timer for Proceeding
Date: Wed, 10 Apr 2002 11:26:56 -0400
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Timer C is for proxies.
In a UAC, if you want to wait forever in the proceeding state you are free
to do so. This is application specific.

Alexandre Charest

-----Original Message-----
From: sunayak@hss.hns.com [mailto:sunayak@hss.hns.com]
Sent: April 10, 2002 10:44 AM
To: Fuxbruner, Amihay
Cc: sip@ietf.org; Ivanov, Elena
Subject: Re: [Sip] Question: Timer for Proceeding





Hi Amihay,
     The client transaction does start timer C
when sending the INVITE request. This timer is
reset on receiving any provisional response. On
timeout, the client transaction moves to Terminated
state.

     In bis-05, this was described in Section 17,
and also indicated in the "INVITE client transaction"
diagram. But in bis--9, this description has been moved
to Sec 16.6, bullet 11. As an aside, it seems to have been
missed out in the INVITE client transaction diagram....


Subhash Nayak.
Hughes Software Systems.
http://www.hssworld.com






"Fuxbruner, Amihay" <Amihay_Fuxbruner@icomverse.com> on 04/10/2002 07:47:48
PM

To:   sip@ietf.org
cc:   "Ivanov, Elena" <Elena_Ivanov@icomverse.com> (bcc: Subhash Ullal
      Nayak/HSSBLR)

Subject:  [Sip] Question: Timer for Proceeding




Hi,

Bis 09 section 17.1.1.2 does not include timer for proceeding state.
The protocol also says one SHOULD NOT retransmit in proceeding state.
Should we need a timer or is it application decision ?

Thanks,

Amihay Fuxbruner
System Engineering - Signaling R&D
Comverse


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From daemon@optimus.ietf.org  Wed Apr 10 12:14:22 2002
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        "Isaac, Dudy"
	 <Dudy_Isaac@icomverse.com>,
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Subject: [Sip] Question: Terminate early dialog with BYE or CANCEL
Date: Wed, 10 Apr 2002 18:54:33 +0300
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Hi,

Terminating early dialog is allowed with BYE or CANCEL.
What is the best / recommended way ?
Case BYE, is it correct that we eliminate the use of ACK ? 

Regards,

Amihay Fuxbruner
System Engineering - Signaling R&D
Comverse

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<P><FONT SIZE=2>Hi,</FONT>
</P>

<P><FONT SIZE=2>Terminating early dialog is allowed with BYE or CANCEL.</FONT>
<BR><FONT SIZE=2>What is the best / recommended way ?</FONT>
<BR><FONT SIZE=2>Case BYE, is it correct that we eliminate the use of ACK ? </FONT>
</P>

<P><FONT SIZE=2>Regards,</FONT>
</P>

<P><FONT SIZE=2>Amihay Fuxbruner</FONT>
<BR><FONT SIZE=2>System Engineering - Signaling R&amp;D</FONT>
<BR><FONT SIZE=2>Comverse</FONT>
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From daemon@optimus.ietf.org  Wed Apr 10 15:49:57 2002
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Subject: [Sip] event subscription 
To: sip@ietf.org
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Hi,

Based on the sip-events draft, it seems that a subscription is only
defined/established within the context of a dialog. Is this correct? Is it
possible to have a subscription on a user/resource level outside a dialog
as long as route and contact  information of the subscriber is provisioned
for the subscription?

Thanks,
YoungSun


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From daemon@optimus.ietf.org  Wed Apr 10 17:08:03 2002
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From: "Bert Culpepper" <bert.culpepper@intervoice-brite.com>
To: <ysong@telcordia.com>, <sip@ietf.org>
Subject: RE: [Sip] event subscription 
Date: Wed, 10 Apr 2002 16:49:39 -0400
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Hi,

I read the sip-event-05 I-D as saying a subscription can create a dialog.
Also, that a subscription can exist within an established dialog.  And that
a dialog established by an INVITE transaction that also has an associated
subscription won't terminate until both the INVITE established session *and*
SUBSCRIBE established subscription have both been terminated.

An example of a non-INVITE established dialog is a phone subscribing to
message-waiting indications from a voice-mail server.

Regards,
Bert

> -----Original Message-----
> From: ysong@telcordia.com
>
> Hi,
>
> Based on the sip-events draft, it seems that a subscription is only
> defined/established within the context of a dialog. Is this
> correct? Is it
> possible to have a subscription on a user/resource level
> outside a dialog
> as long as route and contact  information of the subscriber
> is provisioned
> for the subscription?
>
> Thanks,
> YoungSun
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From daemon@optimus.ietf.org  Wed Apr 10 19:38:07 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Date: Wed, 10 Apr 2002 18:19:34 -0500
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Subject: [Sip] Slides for SIP WG, IETF 53 Posted, plase check
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I think I have posted all the slides I received for SIP working group
meetings at IETF 53 at:

http://www.softarmor.com/sipwg/meets/ietf53/slides/


Authors: Please check and make sure your slides are there and
information is correct. This has to be finalized to the proceedings team
by April 12. Nothing's certain but taxes and proceedings . . .

--
Dean


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From daemon@optimus.ietf.org  Thu Apr 11 02:31:59 2002
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Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D 
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"Drage, Keith (Keith)" wrote:
> 

> > > I would prefer to keep Path, or at least something shorter
> > > that the mouthfull that is currently proposed.
> >
> > I suspect RRR would get used a lot . . .
> >
> If it is allowed to be defined. Surely I remember a discussion on the
> list
> some time back saying that no new short form codings would be defined?
> Also
> none of them are more than one character, so do we now start inventing 3
> character ones? Certainly none of the recent headers added seem to have
> short forms.

Short forms are still allowed, but restricted to one character. THe idea
is that they are allocated to headers used frequently. It may all be
moot with compression, but I agree with Anders not to use a long name if
we can avoid it.

> > Actually, Record-Route entries are NOT changed in the response, just
> > copied back towards the source. That is, they accumulate on the INVITE
> > message, and are returned to the UA on the 200OK response. I think the
> > text in "path" now has exactly the same behaviour. If it doesn't, I
> did
> > something wrong, so please verify . . .
> >
> The piece of funtionality I was referring to is the UA handling of the
> response. At that point things do get different. Up to that point, I
> believe
> the handling is identical.

There is a major difference in handling right now. In the Path draft,
the proxy adds its URI as the BOTTOM-MOST entry in the Route header, and
the registrar inverts it before using. WIth record-route, its added as
the topmost entry. I would personally prefer to keep things consistent,
and have proxies add new values at the top. This also avoids the need to
invert them at the registrar.

-Jonathan R.


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

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From daemon@optimus.ietf.org  Thu Apr 11 04:08:15 2002
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Off to "real work" for a week or two, and its amazing the length of some
threads that amass whilst I am not actively reading the list... anyway,
my comments inline.


Mark Watson wrote:
> 
> : (1) The user inserts their identity, but the subscriber of the
> outgoing service being used requires outgoing sessions to be anonymous.
> Only the outgoing proxy knows this - well, it's a B2BUA now.
> [Peterson, Jon] One could just as easily argue that this is a
> requirement not to force such services into the local outbound proxy.
> The UA could always dynamically learn about privacy preferences from the
> service provider in some fashion, and the service provider could reject
> requests (at the local outbound) that don't conform to the privacy
> rules. And we should when possible push SIP features to the endpoints
> anyway.
> 
> MW> I'm happy with any solution which works & the above sounds
> promising. How would we get the privacy preferences to the UAC & is it
> backwards compatible ?

I agree with Jon that this is a reasonable approach. I had suggested the
same to some 3g folks in the past, to deal with the more generic
problem, which was that the user signed up for privacy services, but
"forgets" to put anonymous in the from field. 

> 
> (2) A calls B, who forwards to C. B requires privacy, but unfortunately
> A has inserted B's identity in the To field. So the proxy performing
> forwarding for B must modify the To field - OK, so it's a B2BUA too.

IMHO, the notion of a network provided anonymous forwarding service is
primarily an artifact of the PSTN, where the locations of identity
information were easy to spot and remove. In an IP network, with a
protocol like SIP, you are kidding yourself if you think To/From is
sufficient to "cleanse". Identity information could be lurking in
headers both known and unknown. Some examples:

Subject: Hey, Bob, lets go for lunch
NewHeader: Bob once again
Call-Info: http://foo.com/picture-of-bob-and-me.jpg

and let us not forget the SDP too...

The only way in SIP to offer a useful (meaning, would actually work
reliably from the subscribers perspective) anonymous forwarding service
is to use a B2BUA "with a vengeance". This B2BUA would not propagate ANY
information from the original request, stripping everything, and
reformulating the mandatory headers on its own. It would probably need
to act as a media intermediary too. I would argue that, if you want an
anonymous forwarding service to be network provided, you really DONT
want a proxy in any sense of the word. THis would destroy services, of
course, but thats the penalty for this feature.

Arguing that only the To field needs to be stripped, because that is the
only thing the sip spec defines as identity, will hardly be comforting
to a user that depends on the proper functioning of this service.






> [Peterson, Jon] If B redirects A to C instead, we might get some better
> leverage on this problem - that could be one way that B could 'require
> privacy'. Today the SIP spec RECOMMENDs that the To field in the new
> request following the redirection be the same as the To field in the
> original request (although it says UAs MAY update the To and whatnot if
> they are so inclined) - however, the spec also allows you to insert
> replacement headers into the URL, like:
> 
> Contact: sip:C@cleveland.com?To=anonymous@invalid.net
> <mailto:sip:C@cleveland.com?To=anonymous@invalid.net>
> 
> I bring these points up not because I disagree that intermediaries
> providing privacy and identity will often need to be More Than Just A
> Proxy - but I think there are reasonable attacks on the examples you
> happen to raise here that don't require resorting to a B2BUA. Let's not
> be too quick to close doors.

IMHO, it would be more proper to reject the request outright, with some
kind of new status code indicating that the called party wishes to
forward the call anonymously. However, the problem remains that B now
has to trust A to properly strip all identifying information, not jus
the To field - a risky proposition. Effectively, B needs to ask A to
perform the same functions that B2BUA above provides....

-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 11 04:15:49 2002
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Date: Thu, 11 Apr 2002 03:28:18 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Mark Watson <mwatson@nortelnetworks.com>
CC: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mark Watson wrote:
> 
> All,
> 
> I feel profoundly lucky that yesterday and Friday were public holidays
> in the UK :-)
> 
> Perhaps I can take advantage of this vantage point to offer a summary of
> the thread. Protagonists please speak up if I have mis-represented you
> below, but please don't re-open old fronts.

Thanks for the summary. It is well needed. I suspect much of the lack of
comment derives from the difficulty in following this long thread.

I would kindly request all participants to please keep the discussions
technical. A lot of the emails in this thread were personal attacks.
Many more were threats along the general line of "do X or standards body
Y will do Z". I really hate the latter ones. Let us hear no more of
either type of nonsense. It is not the IETF way.

> 
> 1) Call-Info
> 
> Long arguments about the alledged similarities/differences between
> Call-Info and RPID. Suffice to say that the point has been made that
> Call-Info (and perhaps other things) could be abused to do the things
> that RPID does, and in particular to do the 'Bad Things' that some are
> worried about with RPID. The point of disagreement was whether in the
> case of Call-Info, this would be 'major abuse', with RPID was set up to
> make these 'Bad Things' easy, or whether there was some equivalence.
> 
> The key point is that this kind of abuse of Call-Info may happen if we
> do not provide an alternative.

It was never our intent for Call-Info to provide any form of network
asserted identity that could be used in any useful way. Generally, the
idea was that insertion of Call-Info was either done by the UA, or in
the case of proxy-inserted values, was for "opt-in" services were the
user had signed up for content to be added. That said, there are
certainly cases where the caller may opt in for this service but require
privacy for a particular call, and we need a mechanism to support that.

However, in no way would I support or advocate the use of Call-Info as
an alternative for R-P-ID. 

> 
> 2) Abuse of From/To headers
> 
> Jon is concerned that 'untrusted RPID' provides a standardised
> alternative to From/To, allowing Bad Things, like never putting the real
> From/To information in the From/To fields.

I think that the proper way to handle this, as pursued in other threads,
is to provide a sane alternative to 3gpp so that they don't do this. I
believe we all agreed it would be bad.

> 
> Others have pointed out that the draft explicitly states that untrusted
> UAs MUST NOT add RPID. However the draft explicitly described proxy
> handling of RPID headers from untrusted sources (proxies OR UAs), and
> allows these to be propogated, marked as untrusted.
> 
> Therefore, a UA which DID add RPID, although non-compliant to the draft,
> would have its RPID propogated through the network.
> 
> 3) 'Untrustworthy RPID'
> 
> Flemming, Bill et al argued that RPID information is still useful when
> it is 'untrustworthy'. This can occur legitimately within the draft when
> an RPID which is not 'private' (and so not encrypted) crosses a trust
> boundary. This would represent an identity asserted by Network A and
> passed to Network B, when B has no explicit trust relationship with A.
> 
> The contention was that since this information still has some value, it
> should be passed to the UAS for display, and might also have some
> application for call trace, although obviously not as useful as a
> 'trustworthy' identity.
> 
> Jon argued that the From field provides well enough for display, and the
> value of untrustworthy information for call trace is debatable (and has
> been debated at length).

IMHO, 'untrusted RPID' is identical to the From, except that it is
purportedly inserted by a network element, whereas From is purportedly
inserted by the end user. Neither provides any guarantees. The essence
of the debate, I think, is whether you believe there is value to the end
user in a field which is the identity purportedly inserted by some
network element.  Since there is apparently value to the identity
purportedly inserted by the end user, I see no reason why we should
discriminate. 

-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 11 04:16:38 2002
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"Elwell, John" wrote:
> 
> RFC2543bis09 section 8.1.2, last paragraph seems to imply that
> destinations
> obtained through DNS look-up are tried sequentially ("trying each
> address
> until a server is contacted") rather than in parallel. Also it refers to
> the
> sip-srv draft, section 4.3 of which does not seem to suggest parallel
> attempts. Can anyone confirm that attempts must not be tried in
> parallel?

Yes. 

-Jonathan R.

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

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: DHCPv6 Options for SIP Servers
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> > : (1) The user inserts their identity, but the subscriber of the
> > outgoing service being used requires outgoing sessions to 
> be anonymous.
> > Only the outgoing proxy knows this - well, it's a B2BUA now.
> > [Peterson, Jon] One could just as easily argue that this is a
> > requirement not to force such services into the local 
> outbound proxy.
> > The UA could always dynamically learn about privacy 
> preferences from the
> > service provider in some fashion, and the service provider 
> could reject
> > requests (at the local outbound) that don't conform to the privacy
> > rules. And we should when possible push SIP features to the 
> endpoints
> > anyway.
> > 
> > MW> I'm happy with any solution which works & the above sounds
> > promising. How would we get the privacy preferences to the 
> UAC & is it
> > backwards compatible ?
> 
> I agree with Jon that this is a reasonable approach. I had 
> suggested the
> same to some 3g folks in the past, to deal with the more generic
> problem, which was that the user signed up for privacy services, but
> "forgets" to put anonymous in the from field. 
>

In principle this would work for 3GPP, if the 'xxx Privacy Required'
response were defined, because 3GPP can place new requirements on the SIP
clients in 3GPP networks. But it's no good for people wanting to use RFC2543
clients & in practice probably too late for 3GPP.

Also, the requirement for subscriber privacy could be based on some complex
algorithm in the outgoing proxy (e.g. based on call destination), so the UA
cannot expect to 'learn' when it should include its identity and when not.
This could lead to a large percentage of calls having try without privacy
and then try again. This would not be acceptable over a wireless link.


> > 
> > (2) A calls B, who forwards to C. B requires privacy, but 
> unfortunately
> > A has inserted B's identity in the To field. So the proxy performing
> > forwarding for B must modify the To field - OK, so it's a B2BUA too.
> 
> IMHO, the notion of a network provided anonymous forwarding service is
> primarily an artifact of the PSTN, where the locations of identity
> information were easy to spot and remove. In an IP network, with a
> protocol like SIP, you are kidding yourself if you think To/From is
> sufficient to "cleanse". Identity information could be lurking in
> headers both known and unknown. Some examples:
> 

I'm not arguing that To/From are the only things you would need to modify -
it's just that To/From are the problematic ones because of the link to
dialog matching in RFC2543 clients. Other things are easy (well, easier) to
modify.

> Subject: Hey, Bob, lets go for lunch
> NewHeader: Bob once again
> Call-Info: http://foo.com/picture-of-bob-and-me.jpg
> 
> and let us not forget the SDP too...
> 
> The only way in SIP to offer a useful (meaning, would actually work
> reliably from the subscribers perspective) anonymous 
> forwarding service
> is to use a B2BUA "with a vengeance". This B2BUA would not 
> propagate ANY
> information from the original request, stripping everything, and
> reformulating the mandatory headers on its own. It would probably need
> to act as a media intermediary too. I would argue that, if you want an
> anonymous forwarding service to be network provided, you really DONT
> want a proxy in any sense of the word. THis would destroy services, of
> course, but thats the penalty for this feature.
>

You can imagine a whole continuum of 'anonymity' services providing various
levels of anonymity traded off against service transparency. For example, IP
address privacy may not be required in all cases, especially with IPv6
privacy addresses.

I don't know where on this continuum you would have to be to meet the
obligation to provide an anonymous forwarding service, but I'm arguing that
even the simplest such service cannot be provided by a straightforward SIP
proxy.

Any such service would *at least* have to modify From/To. Others seemed to
be saying that since these fields were entirely a matter for the UA, then
the network has no responsibility for their contents and would never need to
modify them.

As to whether 'anonymous forwarding' would be required to be provided by
Public Service Providers or not, it's a matter of legal interpretation of
the Data Protection legislation. We can't say whether it is, or whether it
is not, although we each may have opinions.

For my part, I can't see why this requirement would not apply to Public
Telcomminications Service Providers independent of the technology they use
to offer service. Therefore I think there is *at least* a significant
*chance* that SPs will have to provide such services, and so it's important
that we understand how they might do that.


> Arguing that only the To field needs to be stripped, because 
> that is the
> only thing the sip spec defines as identity, will hardly be comforting
> to a user that depends on the proper functioning of this service.
> 

See above.

...Mark
> 

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<TITLE>RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP =
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<P><FONT SIZE=3D2>&gt; &gt; : (1) The user inserts their identity, but =
the subscriber of the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; outgoing service being used requires =
outgoing sessions to </FONT>
<BR><FONT SIZE=3D2>&gt; be anonymous.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Only the outgoing proxy knows this - well, =
it's a B2BUA now.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [Peterson, Jon] One could just as easily =
argue that this is a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; requirement not to force such services =
into the local </FONT>
<BR><FONT SIZE=3D2>&gt; outbound proxy.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; The UA could always dynamically learn =
about privacy </FONT>
<BR><FONT SIZE=3D2>&gt; preferences from the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; service provider in some fashion, and the =
service provider </FONT>
<BR><FONT SIZE=3D2>&gt; could reject</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; requests (at the local outbound) that =
don't conform to the privacy</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; rules. And we should when possible push =
SIP features to the </FONT>
<BR><FONT SIZE=3D2>&gt; endpoints</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; anyway.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; MW&gt; I'm happy with any solution which =
works &amp; the above sounds</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; promising. How would we get the privacy =
preferences to the </FONT>
<BR><FONT SIZE=3D2>&gt; UAC &amp; is it</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; backwards compatible ?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I agree with Jon that this is a reasonable =
approach. I had </FONT>
<BR><FONT SIZE=3D2>&gt; suggested the</FONT>
<BR><FONT SIZE=3D2>&gt; same to some 3g folks in the past, to deal with =
the more generic</FONT>
<BR><FONT SIZE=3D2>&gt; problem, which was that the user signed up for =
privacy services, but</FONT>
<BR><FONT SIZE=3D2>&gt; &quot;forgets&quot; to put anonymous in the =
from field. </FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>In principle this would work for 3GPP, if the 'xxx =
Privacy Required' response were defined, because 3GPP can place new =
requirements on the SIP clients in 3GPP networks. But it's no good for =
people wanting to use RFC2543 clients &amp; in practice probably too =
late for 3GPP.</FONT></P>

<P><FONT SIZE=3D2>Also, the requirement for subscriber privacy could be =
based on some complex algorithm in the outgoing proxy (e.g. based on =
call destination), so the UA cannot expect to 'learn' when it should =
include its identity and when not. This could lead to a large =
percentage of calls having try without privacy and then try again. This =
would not be acceptable over a wireless link.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (2) A calls B, who forwards to C. B =
requires privacy, but </FONT>
<BR><FONT SIZE=3D2>&gt; unfortunately</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; A has inserted B's identity in the To =
field. So the proxy performing</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; forwarding for B must modify the To field =
- OK, so it's a B2BUA too.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; IMHO, the notion of a network provided =
anonymous forwarding service is</FONT>
<BR><FONT SIZE=3D2>&gt; primarily an artifact of the PSTN, where the =
locations of identity</FONT>
<BR><FONT SIZE=3D2>&gt; information were easy to spot and remove. In an =
IP network, with a</FONT>
<BR><FONT SIZE=3D2>&gt; protocol like SIP, you are kidding yourself if =
you think To/From is</FONT>
<BR><FONT SIZE=3D2>&gt; sufficient to &quot;cleanse&quot;. Identity =
information could be lurking in</FONT>
<BR><FONT SIZE=3D2>&gt; headers both known and unknown. Some =
examples:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I'm not arguing that To/From are the only things you =
would need to modify - it's just that To/From are the problematic ones =
because of the link to dialog matching in RFC2543 clients. Other things =
are easy (well, easier) to modify.</FONT></P>

<P><FONT SIZE=3D2>&gt; Subject: Hey, Bob, lets go for lunch</FONT>
<BR><FONT SIZE=3D2>&gt; NewHeader: Bob once again</FONT>
<BR><FONT SIZE=3D2>&gt; Call-Info: <A =
HREF=3D"http://foo.com/picture-of-bob-and-me.jpg" =
TARGET=3D"_blank">http://foo.com/picture-of-bob-and-me.jpg</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; and let us not forget the SDP too...</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The only way in SIP to offer a useful (meaning, =
would actually work</FONT>
<BR><FONT SIZE=3D2>&gt; reliably from the subscribers perspective) =
anonymous </FONT>
<BR><FONT SIZE=3D2>&gt; forwarding service</FONT>
<BR><FONT SIZE=3D2>&gt; is to use a B2BUA &quot;with a vengeance&quot;. =
This B2BUA would not </FONT>
<BR><FONT SIZE=3D2>&gt; propagate ANY</FONT>
<BR><FONT SIZE=3D2>&gt; information from the original request, =
stripping everything, and</FONT>
<BR><FONT SIZE=3D2>&gt; reformulating the mandatory headers on its own. =
It would probably need</FONT>
<BR><FONT SIZE=3D2>&gt; to act as a media intermediary too. I would =
argue that, if you want an</FONT>
<BR><FONT SIZE=3D2>&gt; anonymous forwarding service to be network =
provided, you really DONT</FONT>
<BR><FONT SIZE=3D2>&gt; want a proxy in any sense of the word. THis =
would destroy services, of</FONT>
<BR><FONT SIZE=3D2>&gt; course, but thats the penalty for this =
feature.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>You can imagine a whole continuum of 'anonymity' =
services providing various levels of anonymity traded off against =
service transparency. For example, IP address privacy may not be =
required in all cases, especially with IPv6 privacy =
addresses.</FONT></P>

<P><FONT SIZE=3D2>I don't know where on this continuum you would have =
to be to meet the obligation to provide an anonymous forwarding =
service, but I'm arguing that even the simplest such service cannot be =
provided by a straightforward SIP proxy.</FONT></P>

<P><FONT SIZE=3D2>Any such service would *at least* have to modify =
From/To. Others seemed to be saying that since these fields were =
entirely a matter for the UA, then the network has no responsibility =
for their contents and would never need to modify them.</FONT></P>

<P><FONT SIZE=3D2>As to whether 'anonymous forwarding' would be =
required to be provided by Public Service Providers or not, it's a =
matter of legal interpretation of the Data Protection legislation. We =
can't say whether it is, or whether it is not, although we each may =
have opinions.</FONT></P>

<P><FONT SIZE=3D2>For my part, I can't see why this requirement would =
not apply to Public Telcomminications Service Providers independent of =
the technology they use to offer service. Therefore I think there is =
*at least* a significant *chance* that SPs will have to provide such =
services, and so it's important that we understand how they might do =
that.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; Arguing that only the To field needs to be =
stripped, because </FONT>
<BR><FONT SIZE=3D2>&gt; that is the</FONT>
<BR><FONT SIZE=3D2>&gt; only thing the sip spec defines as identity, =
will hardly be comforting</FONT>
<BR><FONT SIZE=3D2>&gt; to a user that depends on the proper =
functioning of this service.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>See above.</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy  
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> I don't know where on this continuum you would have to be to meet the
> obligation to provide an anonymous forwarding service, but I'm arguing
> that even the simplest such service cannot be provided by a
> straightforward SIP proxy.

For the privacy aspect I believe that it can be provided at least for
anonymity in the To or From, by a standard proxy, but as Johnathan
stated a B2BUA would be required to provide the topology (IP address)
hiding at all other levels, and then it would also be required to
provide the same actions for media. As for the example of content, 

> > Subject: Hey, Bob, lets go for lunch
> > NewHeader: Bob once again
> > Call-Info: http://foo.com/picture-of-bob-and-me.jpg
> >

I dont think that these fall under the same category as far as identity
is concerned, since they are not headers asserting identity, merely
topics or objects within a conversation. To me identity would be
something more tangible than that, such as the content of a From, Via,
Route, etc., or even IP addressing...but then again if UA's are behind a
B2BUA even IP addressing is less of an identifier of the originator. 

Does this seem like a correct assumption?

> 
> Any such service would *at least* have to modify From/To. Others
> seemed to be saying that since these fields were entirely a matter for
> the UA, then the network has no responsibility for their contents and
> would never need to modify them.
>

I believe that, since we have been discussing this in terms of privacy
as it relates to an SP environment, the responsibility lies with the SP.
A UA not within an SP realm can assert that they are anyone they want to
be including anonymous, right? In an authenticated SP environment the
option for a UA to muck with identity would not be permitted becuase it
wouldnt authenticate correctly/at all, would it?

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Just on this one issue of obfuscation of the From/To field....

Let's be really clear about why this has come about:

Some Service Providers *believe* in the following proposition:

(*) IF the UA is allowed to put identity information into From/To, THEN the
Service Provider will be forced (by Data Protection) to provide services
which modify these fields, in order to protect the privacy of people other
than the actual calling user - subscriber, forwarding etc.

First, set aside the question of whether to agree with (*) or not.

There are three options for people with this belief:
1) *Require* permanantly that the UA does not put identity information into
From/To
2) Reject requests where the UA puts identity information in From/To if
privacy is required
3) Build network services which provide the necessary modification
(+whatever else is needed)

I do not think there are any other options.

We all agree that (1) is bad. (2) looks attractive but is not backwards
compatible, requires a double request (not good for wireless) & has issues
as to whether you trust the UA (in redirection scenarios).

I was looking for a consensus that (3) is a possible and valid thing to do
for these Service Providers. If the IETF SIP group agree that (3) and not
(1) is _a_ valid solution to this problem, then there is more chance of
other fora adopting (3) instead of (1).

So far, people have been arguing against the belief (*). The question is not
whether (*) is demonstrably true but whether there is sufficient *chance* of
it being true to justify serious consideration. A significant group of
Service Providers obviously think it is - they are worried they will be in
this situation and want a solution defined - that should be enough for us to
discuss solutions.

So, can we agree to recommend at least that people with belief (*) adopt
solution (3) and not solution (1) ?

...Mark



> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: 11 April 2002 08:28
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon;
> 'William Marshall'
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> 
> 
> Mark Watson wrote:
> > 
> > All,
> > 
> > I feel profoundly lucky that yesterday and Friday were 
> public holidays
> > in the UK :-)
> > 
> > Perhaps I can take advantage of this vantage point to offer 
> a summary of
> > the thread. Protagonists please speak up if I have 
> mis-represented you
> > below, but please don't re-open old fronts.
> 
> Thanks for the summary. It is well needed. I suspect much of 
> the lack of
> comment derives from the difficulty in following this long thread.
> 
> I would kindly request all participants to please keep the discussions
> technical. A lot of the emails in this thread were personal attacks.
> Many more were threats along the general line of "do X or 
> standards body
> Y will do Z". I really hate the latter ones. Let us hear no more of
> either type of nonsense. It is not the IETF way.
> 
> > 
> > 1) Call-Info
> > 
> > Long arguments about the alledged similarities/differences between
> > Call-Info and RPID. Suffice to say that the point has been made that
> > Call-Info (and perhaps other things) could be abused to do 
> the things
> > that RPID does, and in particular to do the 'Bad Things' 
> that some are
> > worried about with RPID. The point of disagreement was 
> whether in the
> > case of Call-Info, this would be 'major abuse', with RPID 
> was set up to
> > make these 'Bad Things' easy, or whether there was some equivalence.
> > 
> > The key point is that this kind of abuse of Call-Info may 
> happen if we
> > do not provide an alternative.
> 
> It was never our intent for Call-Info to provide any form of network
> asserted identity that could be used in any useful way. Generally, the
> idea was that insertion of Call-Info was either done by the UA, or in
> the case of proxy-inserted values, was for "opt-in" services were the
> user had signed up for content to be added. That said, there are
> certainly cases where the caller may opt in for this service 
> but require
> privacy for a particular call, and we need a mechanism to 
> support that.
> 
> However, in no way would I support or advocate the use of Call-Info as
> an alternative for R-P-ID. 
> 
> > 
> > 2) Abuse of From/To headers
> > 
> > Jon is concerned that 'untrusted RPID' provides a standardised
> > alternative to From/To, allowing Bad Things, like never 
> putting the real
> > From/To information in the From/To fields.
> 
> I think that the proper way to handle this, as pursued in 
> other threads,
> is to provide a sane alternative to 3gpp so that they don't do this. I
> believe we all agreed it would be bad.
> 
> > 
> > Others have pointed out that the draft explicitly states 
> that untrusted
> > UAs MUST NOT add RPID. However the draft explicitly described proxy
> > handling of RPID headers from untrusted sources (proxies OR 
> UAs), and
> > allows these to be propogated, marked as untrusted.
> > 
> > Therefore, a UA which DID add RPID, although non-compliant 
> to the draft,
> > would have its RPID propogated through the network.
> > 
> > 3) 'Untrustworthy RPID'
> > 
> > Flemming, Bill et al argued that RPID information is still 
> useful when
> > it is 'untrustworthy'. This can occur legitimately within 
> the draft when
> > an RPID which is not 'private' (and so not encrypted) 
> crosses a trust
> > boundary. This would represent an identity asserted by Network A and
> > passed to Network B, when B has no explicit trust 
> relationship with A.
> > 
> > The contention was that since this information still has 
> some value, it
> > should be passed to the UAS for display, and might also have some
> > application for call trace, although obviously not as useful as a
> > 'trustworthy' identity.
> > 
> > Jon argued that the From field provides well enough for 
> display, and the
> > value of untrustworthy information for call trace is 
> debatable (and has
> > been debated at length).
> 
> IMHO, 'untrusted RPID' is identical to the From, except that it is
> purportedly inserted by a network element, whereas From is purportedly
> inserted by the end user. Neither provides any guarantees. The essence
> of the debate, I think, is whether you believe there is value 
> to the end
> user in a field which is the identity purportedly inserted by some
> network element.  Since there is apparently value to the identity
> purportedly inserted by the end user, I see no reason why we should
> discriminate. 
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> 

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<P><FONT SIZE=3D2>Just on this one issue of obfuscation of the From/To =
field....</FONT>
</P>

<P><FONT SIZE=3D2>Let's be really clear about why this has come =
about:</FONT>
</P>

<P><FONT SIZE=3D2>Some Service Providers *believe* in the following =
proposition:</FONT>
</P>

<P><FONT SIZE=3D2>(*) IF the UA is allowed to put identity information =
into From/To, THEN the Service Provider will be forced (by Data =
Protection) to provide services which modify these fields, in order to =
protect the privacy of people other than the actual calling user - =
subscriber, forwarding etc.</FONT></P>

<P><FONT SIZE=3D2>First, set aside the question of whether to agree =
with (*) or not.</FONT>
</P>

<P><FONT SIZE=3D2>There are three options for people with this =
belief:</FONT>
<BR><FONT SIZE=3D2>1) *Require* permanantly that the UA does not put =
identity information into From/To</FONT>
<BR><FONT SIZE=3D2>2) Reject requests where the UA puts identity =
information in From/To if privacy is required</FONT>
<BR><FONT SIZE=3D2>3) Build network services which provide the =
necessary modification (+whatever else is needed)</FONT>
</P>

<P><FONT SIZE=3D2>I do not think there are any other options.</FONT>
</P>

<P><FONT SIZE=3D2>We all agree that (1) is bad. (2) looks attractive =
but is not backwards compatible, requires a double request (not good =
for wireless) &amp; has issues as to whether you trust the UA (in =
redirection scenarios).</FONT></P>

<P><FONT SIZE=3D2>I was looking for a consensus that (3) is a possible =
and valid thing to do for these Service Providers. If the IETF SIP =
group agree that (3) and not (1) is _a_ valid solution to this problem, =
then there is more chance of other fora adopting (3) instead of =
(1).</FONT></P>

<P><FONT SIZE=3D2>So far, people have been arguing against the belief =
(*). The question is not whether (*) is demonstrably true but whether =
there is sufficient *chance* of it being true to justify serious =
consideration. A significant group of Service Providers obviously think =
it is - they are worried they will be in this situation and want a =
solution defined - that should be enough for us to discuss =
solutions.</FONT></P>

<P><FONT SIZE=3D2>So, can we agree to recommend at least that people =
with belief (*) adopt solution (3) and not solution (1) ?</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 11 April 2002 08:28</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org; Ben Campbell; Flemming =
Andreasen; Peterson, Jon;</FONT>
<BR><FONT SIZE=3D2>&gt; 'William Marshall'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark Watson wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; All,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I feel profoundly lucky that yesterday and =
Friday were </FONT>
<BR><FONT SIZE=3D2>&gt; public holidays</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; in the UK :-)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Perhaps I can take advantage of this =
vantage point to offer </FONT>
<BR><FONT SIZE=3D2>&gt; a summary of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the thread. Protagonists please speak up =
if I have </FONT>
<BR><FONT SIZE=3D2>&gt; mis-represented you</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; below, but please don't re-open old =
fronts.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks for the summary. It is well needed. I =
suspect much of </FONT>
<BR><FONT SIZE=3D2>&gt; the lack of</FONT>
<BR><FONT SIZE=3D2>&gt; comment derives from the difficulty in =
following this long thread.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I would kindly request all participants to =
please keep the discussions</FONT>
<BR><FONT SIZE=3D2>&gt; technical. A lot of the emails in this thread =
were personal attacks.</FONT>
<BR><FONT SIZE=3D2>&gt; Many more were threats along the general line =
of &quot;do X or </FONT>
<BR><FONT SIZE=3D2>&gt; standards body</FONT>
<BR><FONT SIZE=3D2>&gt; Y will do Z&quot;. I really hate the latter =
ones. Let us hear no more of</FONT>
<BR><FONT SIZE=3D2>&gt; either type of nonsense. It is not the IETF =
way.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 1) Call-Info</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Long arguments about the alledged =
similarities/differences between</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Call-Info and RPID. Suffice to say that =
the point has been made that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Call-Info (and perhaps other things) could =
be abused to do </FONT>
<BR><FONT SIZE=3D2>&gt; the things</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that RPID does, and in particular to do =
the 'Bad Things' </FONT>
<BR><FONT SIZE=3D2>&gt; that some are</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; worried about with RPID. The point of =
disagreement was </FONT>
<BR><FONT SIZE=3D2>&gt; whether in the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; case of Call-Info, this would be 'major =
abuse', with RPID </FONT>
<BR><FONT SIZE=3D2>&gt; was set up to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; make these 'Bad Things' easy, or whether =
there was some equivalence.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; The key point is that this kind of abuse =
of Call-Info may </FONT>
<BR><FONT SIZE=3D2>&gt; happen if we</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; do not provide an alternative.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; It was never our intent for Call-Info to =
provide any form of network</FONT>
<BR><FONT SIZE=3D2>&gt; asserted identity that could be used in any =
useful way. Generally, the</FONT>
<BR><FONT SIZE=3D2>&gt; idea was that insertion of Call-Info was either =
done by the UA, or in</FONT>
<BR><FONT SIZE=3D2>&gt; the case of proxy-inserted values, was for =
&quot;opt-in&quot; services were the</FONT>
<BR><FONT SIZE=3D2>&gt; user had signed up for content to be added. =
That said, there are</FONT>
<BR><FONT SIZE=3D2>&gt; certainly cases where the caller may opt in for =
this service </FONT>
<BR><FONT SIZE=3D2>&gt; but require</FONT>
<BR><FONT SIZE=3D2>&gt; privacy for a particular call, and we need a =
mechanism to </FONT>
<BR><FONT SIZE=3D2>&gt; support that.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; However, in no way would I support or advocate =
the use of Call-Info as</FONT>
<BR><FONT SIZE=3D2>&gt; an alternative for R-P-ID. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 2) Abuse of From/To headers</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jon is concerned that 'untrusted RPID' =
provides a standardised</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; alternative to From/To, allowing Bad =
Things, like never </FONT>
<BR><FONT SIZE=3D2>&gt; putting the real</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From/To information in the From/To =
fields.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I think that the proper way to handle this, as =
pursued in </FONT>
<BR><FONT SIZE=3D2>&gt; other threads,</FONT>
<BR><FONT SIZE=3D2>&gt; is to provide a sane alternative to 3gpp so =
that they don't do this. I</FONT>
<BR><FONT SIZE=3D2>&gt; believe we all agreed it would be bad.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Others have pointed out that the draft =
explicitly states </FONT>
<BR><FONT SIZE=3D2>&gt; that untrusted</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; UAs MUST NOT add RPID. However the draft =
explicitly described proxy</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; handling of RPID headers from untrusted =
sources (proxies OR </FONT>
<BR><FONT SIZE=3D2>&gt; UAs), and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; allows these to be propogated, marked as =
untrusted.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Therefore, a UA which DID add RPID, =
although non-compliant </FONT>
<BR><FONT SIZE=3D2>&gt; to the draft,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; would have its RPID propogated through the =
network.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 3) 'Untrustworthy RPID'</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Flemming, Bill et al argued that RPID =
information is still </FONT>
<BR><FONT SIZE=3D2>&gt; useful when</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; it is 'untrustworthy'. This can occur =
legitimately within </FONT>
<BR><FONT SIZE=3D2>&gt; the draft when</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; an RPID which is not 'private' (and so not =
encrypted) </FONT>
<BR><FONT SIZE=3D2>&gt; crosses a trust</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; boundary. This would represent an identity =
asserted by Network A and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; passed to Network B, when B has no =
explicit trust </FONT>
<BR><FONT SIZE=3D2>&gt; relationship with A.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; The contention was that since this =
information still has </FONT>
<BR><FONT SIZE=3D2>&gt; some value, it</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; should be passed to the UAS for display, =
and might also have some</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; application for call trace, although =
obviously not as useful as a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 'trustworthy' identity.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jon argued that the From field provides =
well enough for </FONT>
<BR><FONT SIZE=3D2>&gt; display, and the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; value of untrustworthy information for =
call trace is </FONT>
<BR><FONT SIZE=3D2>&gt; debatable (and has</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; been debated at length).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; IMHO, 'untrusted RPID' is identical to the =
From, except that it is</FONT>
<BR><FONT SIZE=3D2>&gt; purportedly inserted by a network element, =
whereas From is purportedly</FONT>
<BR><FONT SIZE=3D2>&gt; inserted by the end user. Neither provides any =
guarantees. The essence</FONT>
<BR><FONT SIZE=3D2>&gt; of the debate, I think, is whether you believe =
there is value </FONT>
<BR><FONT SIZE=3D2>&gt; to the end</FONT>
<BR><FONT SIZE=3D2>&gt; user in a field which is the identity =
purportedly inserted by some</FONT>
<BR><FONT SIZE=3D2>&gt; network element.&nbsp; Since there is =
apparently value to the identity</FONT>
<BR><FONT SIZE=3D2>&gt; purportedly inserted by the end user, I see no =
reason why we should</FONT>
<BR><FONT SIZE=3D2>&gt; discriminate. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
72 Eagle Rock Avenue</FONT>
<BR><FONT SIZE=3D2>&gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; East Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX: (973) =
952-5050</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; PH:&nbsp; (973) 952-5000</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'cmartin@dasecurenetworks.com'" <cmartin@dasecurenetworks.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Peterson, Jon'"
	 <jon.peterson@neustar.biz>,
        "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Priva
	cy   draft)
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> For the privacy aspect I believe that it can be provided at least for
> anonymity in the To or From, by a standard proxy, but as Johnathan
> stated a B2BUA would be required to provide the topology (IP address)
> hiding at all other levels, and then it would also be required to
> provide the same actions for media. As for the example of content, 

If you want to be compatible with RFC2543 clients, then if you muck with
From/To, you need to change them back in responses. This is not something
which a proxy, as defined in the RFC, would do. That's not to say you
couldn't build a device which looked remarkably *like* a proxy in all
respects apart from this. You might even call it a proxy, but in pedantic
specification terms that would not be correct.


> I believe that, since we have been discussing this in terms of privacy
> as it relates to an SP environment, the responsibility lies 
> with the SP.
> A UA not within an SP realm can assert that they are anyone 
> they want to
> be including anonymous, right? In an authenticated SP environment the
> option for a UA to muck with identity would not be permitted 
> becuase it
> wouldnt authenticate correctly/at all, would it?
> 

As I understand it, From is not used for authentication. A proxy which
authenticates someone uses different fields and is not required then to
check that the From field is 'correct' for the authenticated user. Of course
you *could* do that and it just might be a feature that a Service Provider
would want.

Nevertheless, I would argue that in practice the SP still has some
responsibility for this field, on the grounds that a 'subscriber privacy
service' which did not modify this field would not be regarded as actually
working by the subscriber.

...Mark

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<P><FONT SIZE=3D2>&gt; For the privacy aspect I believe that it can be =
provided at least for</FONT>
<BR><FONT SIZE=3D2>&gt; anonymity in the To or From, by a standard =
proxy, but as Johnathan</FONT>
<BR><FONT SIZE=3D2>&gt; stated a B2BUA would be required to provide the =
topology (IP address)</FONT>
<BR><FONT SIZE=3D2>&gt; hiding at all other levels, and then it would =
also be required to</FONT>
<BR><FONT SIZE=3D2>&gt; provide the same actions for media. As for the =
example of content, </FONT>
</P>

<P><FONT SIZE=3D2>If you want to be compatible with RFC2543 clients, =
then if you muck with From/To, you need to change them back in =
responses. This is not something which a proxy, as defined in the RFC, =
would do. That's not to say you couldn't build a device which looked =
remarkably *like* a proxy in all respects apart from this. You might =
even call it a proxy, but in pedantic specification terms that would =
not be correct.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; I believe that, since we have been discussing =
this in terms of privacy</FONT>
<BR><FONT SIZE=3D2>&gt; as it relates to an SP environment, the =
responsibility lies </FONT>
<BR><FONT SIZE=3D2>&gt; with the SP.</FONT>
<BR><FONT SIZE=3D2>&gt; A UA not within an SP realm can assert that =
they are anyone </FONT>
<BR><FONT SIZE=3D2>&gt; they want to</FONT>
<BR><FONT SIZE=3D2>&gt; be including anonymous, right? In an =
authenticated SP environment the</FONT>
<BR><FONT SIZE=3D2>&gt; option for a UA to muck with identity would not =
be permitted </FONT>
<BR><FONT SIZE=3D2>&gt; becuase it</FONT>
<BR><FONT SIZE=3D2>&gt; wouldnt authenticate correctly/at all, would =
it?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>As I understand it, From is not used for =
authentication. A proxy which authenticates someone uses different =
fields and is not required then to check that the From field is =
'correct' for the authenticated user. Of course you *could* do that and =
it just might be a feature that a Service Provider would =
want.</FONT></P>

<P><FONT SIZE=3D2>Nevertheless, I would argue that in practice the SP =
still has some responsibility for this field, on the grounds that a =
'subscriber privacy service' which did not modify this field would not =
be regarded as actually working by the subscriber.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Drage, Keith \(Keith\)'" <drage@lucent.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D  ACTION:draft-illis...
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> Short forms are still allowed, but restricted to one 
> character. THe idea is that they are allocated to headers 
> used frequently. It may all be moot with compression, but I 
> agree with Anders not to use a long name if we can avoid it.


So how about we just name it RRR?

> There is a major difference in handling right now. In the 
> Path draft, the proxy adds its URI as the BOTTOM-MOST entry 
> in the Route header, and the registrar inverts it before 
> using. WIth record-route, its added as the topmost entry. I 
> would personally prefer to keep things consistent, and have 
> proxies add new values at the top. This also avoids the need 
> to invert them at the registrar.

This has been pointed out to me several times, and will be fixed in next
draft rev.

Thanks!


--
Dean


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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>, "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Thu, 11 Apr 2002 13:23:12 -0500
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Mark wrote:
---------------
Some Service Providers *believe* in the following proposition: 

(*) IF the UA is allowed to put identity information into From/To, THEN
the Service Provider will be forced (by Data Protection) to provide
services which modify these fields, in order to protect the privacy of
people other than the actual calling user - subscriber, forwarding etc.

First, set aside the question of whether to agree with (*) or not. 
There are three options for people with this belief: 

1) *Require* permanantly that the UA does not put identity information
into From/To 
2) Reject requests where the UA puts identity information in From/To if
privacy is required 
3) Build network services which provide the necessary modification
(+whatever else is needed) 
I do not think there are any other options.  
----------------

Just to be a "devils advocate",  I suggest other alternatives:

4) Require that the UA anonymize the To and From headers IF requesting
CLIR on a call, and allow them to populate those headers as desired when
NOT requesting CLIR.

And the "modest suggestion" alternative:

5) Use a non-Internet protocol which completely constrains the service
functionality and user experience to a least common denominator and stop
pretending that your requirements have anything to do with the Internet.


Look, there are two reasons why we may want to use Internet protocols:

1) Because Internet protocols and approaches offer a substantially
richer, more flexible alternative which provides compelling advantages
in terms of service delivery options relative to traditional PSTN
approaches.

2) Because we're jealous of the success the Internet has had and want to
contaminate it with "telco like" requirements so that Internet people
can be miserable too.

I don't know about you, but I believe in #1, and would argue that if we
accept (*) as a requirement, that we're leading down the slippery slope
towards #2, leading to what will be a large push from certain segments
(witness Path discussion) to adopt alternative #5. Alternative #4 is a
reasonable compromise which keeps the responsibility were the Internet
says the responsibility should be, in the hands of the user.

Perhaps instead of To: and From: we should have called these headers
"What-I-will-Call-You:" and "What-I-Want-You-To-Call-Me:". These headers
have ABSOLUTELY NO SEMANTIC VALUE at the signaling and routing level.
They are HUMAN SUPPLIED values intended for HUMAN CONSUMPTION. There are
many other human-supplied values possible in the signaling, and the risk
of privacy violation is equally present on ALL of them.

If we earnestly believe that Data Protection is going to require that
we, by policy, alter these human-supplied/human-consumed values, then
we're logically going to have to do the same for every other
human-supplied/human-consumed values. I would argue that this completely
defeats the value of an Internet protocol in this context.

I believe it is significantly more effective to provide guidelines on
human-supplied/human consumed values, and implement hard requirements
for data protection only on network-supplied values.

--
Dean


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From: Vikas Jain <vikas_sip@yahoo.com>
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I'm looking for some call flow examples detailing the forking scenario. Can someone please guide me to any such ID, document or website?

Thanks,

Vikas Jain



---------------------------------
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<P>I'm looking for some call flow examples detailing the forking scenario. Can someone please guide me to any such ID, document or website?</P>
<P>Thanks,</P>
<P>Vikas Jain</P><p><br><hr size=1><b>Do You Yahoo!?</b><br>
<a href="$rd_url/welcome/?http://taxes.yahoo.com/">Yahoo! Tax Center</a> - online filing with TurboTax
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I AGREE with 3!

:^)




> Mark Watson wrote:
> 
> Just on this one issue of obfuscation of the From/To field....
> 
> Let's be really clear about why this has come about:
> 
> Some Service Providers *believe* in the following proposition:
> 
> (*) IF the UA is allowed to put identity information into From/To,
> THEN the Service Provider will be forced (by Data Protection) to
> provide services which modify these fields, in order to protect the
> privacy of people other than the actual calling user - subscriber,
> forwarding etc.
> 
> First, set aside the question of whether to agree with (*) or not.
> 
> There are three options for people with this belief:
> 1) *Require* permanantly that the UA does not put identity information
> into From/To
> 2) Reject requests where the UA puts identity information in From/To
> if privacy is required
> 3) Build network services which provide the necessary modification
> (+whatever else is needed)
> 
> I do not think there are any other options.
> 
> We all agree that (1) is bad. (2) looks attractive but is not
> backwards compatible, requires a double request (not good for
> wireless) & has issues as to whether you trust the UA (in redirection
> scenarios).
> 
> I was looking for a consensus that (3) is a possible and valid thing
> to do for these Service Providers. If the IETF SIP group agree that
> (3) and not (1) is _a_ valid solution to this problem, then there is
> more chance of other fora adopting (3) instead of (1).
> 
> So far, people have been arguing against the belief (*). The question
> is not whether (*) is demonstrably true but whether there is
> sufficient *chance* of it being true to justify serious consideration.
> A significant group of Service Providers obviously think it is - they
> are worried they will be in this situation and want a solution defined
> - that should be enough for us to discuss solutions.
> 
> So, can we agree to recommend at least that people with belief (*)
> adopt solution (3) and not solution (1) ?
> 
> ...Mark
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: 11 April 2002 08:28
> > To: Watson, Mark [MDN05:EP10:EXCH]
> > Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon;
> > 'William Marshall'
> > Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> >
> >
> >
> >
> > Mark Watson wrote:
> > >
> > > All,
> > >
> > > I feel profoundly lucky that yesterday and Friday were
> > public holidays
> > > in the UK :-)
> > >
> > > Perhaps I can take advantage of this vantage point to offer
> > a summary of
> > > the thread. Protagonists please speak up if I have
> > mis-represented you
> > > below, but please don't re-open old fronts.
> >
> > Thanks for the summary. It is well needed. I suspect much of
> > the lack of
> > comment derives from the difficulty in following this long thread.
> >
> > I would kindly request all participants to please keep the
> discussions
> > technical. A lot of the emails in this thread were personal attacks.
> 
> > Many more were threats along the general line of "do X or
> > standards body
> > Y will do Z". I really hate the latter ones. Let us hear no more of
> > either type of nonsense. It is not the IETF way.
> >
> > >
> > > 1) Call-Info
> > >
> > > Long arguments about the alledged similarities/differences between
> 
> > > Call-Info and RPID. Suffice to say that the point has been made
> that
> > > Call-Info (and perhaps other things) could be abused to do
> > the things
> > > that RPID does, and in particular to do the 'Bad Things'
> > that some are
> > > worried about with RPID. The point of disagreement was
> > whether in the
> > > case of Call-Info, this would be 'major abuse', with RPID
> > was set up to
> > > make these 'Bad Things' easy, or whether there was some
> equivalence.
> > >
> > > The key point is that this kind of abuse of Call-Info may
> > happen if we
> > > do not provide an alternative.
> >
> > It was never our intent for Call-Info to provide any form of network
> 
> > asserted identity that could be used in any useful way. Generally,
> the
> > idea was that insertion of Call-Info was either done by the UA, or
> in
> > the case of proxy-inserted values, was for "opt-in" services were
> the
> > user had signed up for content to be added. That said, there are
> > certainly cases where the caller may opt in for this service
> > but require
> > privacy for a particular call, and we need a mechanism to
> > support that.
> >
> > However, in no way would I support or advocate the use of Call-Info
> as
> > an alternative for R-P-ID.
> >
> > >
> > > 2) Abuse of From/To headers
> > >
> > > Jon is concerned that 'untrusted RPID' provides a standardised
> > > alternative to From/To, allowing Bad Things, like never
> > putting the real
> > > From/To information in the From/To fields.
> >
> > I think that the proper way to handle this, as pursued in
> > other threads,
> > is to provide a sane alternative to 3gpp so that they don't do this.
> I
> > believe we all agreed it would be bad.
> >
> > >
> > > Others have pointed out that the draft explicitly states
> > that untrusted
> > > UAs MUST NOT add RPID. However the draft explicitly described
> proxy
> > > handling of RPID headers from untrusted sources (proxies OR
> > UAs), and
> > > allows these to be propogated, marked as untrusted.
> > >
> > > Therefore, a UA which DID add RPID, although non-compliant
> > to the draft,
> > > would have its RPID propogated through the network.
> > >
> > > 3) 'Untrustworthy RPID'
> > >
> > > Flemming, Bill et al argued that RPID information is still
> > useful when
> > > it is 'untrustworthy'. This can occur legitimately within
> > the draft when
> > > an RPID which is not 'private' (and so not encrypted)
> > crosses a trust
> > > boundary. This would represent an identity asserted by Network A
> and
> > > passed to Network B, when B has no explicit trust
> > relationship with A.
> > >
> > > The contention was that since this information still has
> > some value, it
> > > should be passed to the UAS for display, and might also have some
> > > application for call trace, although obviously not as useful as a
> > > 'trustworthy' identity.
> > >
> > > Jon argued that the From field provides well enough for
> > display, and the
> > > value of untrustworthy information for call trace is
> > debatable (and has
> > > been debated at length).
> >
> > IMHO, 'untrusted RPID' is identical to the From, except that it is
> > purportedly inserted by a network element, whereas From is
> purportedly
> > inserted by the end user. Neither provides any guarantees. The
> essence
> > of the debate, I think, is whether you believe there is value
> > to the end
> > user in a field which is the identity purportedly inserted by some
> > network element.  Since there is apparently value to the identity
> > purportedly inserted by the end user, I see no reason why we should
> > discriminate.
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com
> >

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Subject: Re: Private Info in To/From (was RE: FW: [Sip] Comment, SIP Privacy   
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Inline

> Mark Watson wrote:
> 
> > For the privacy aspect I believe that it can be provided at least
> for
> > anonymity in the To or From, by a standard proxy, but as Johnathan
> > stated a B2BUA would be required to provide the topology (IP
> address)
> > hiding at all other levels, and then it would also be required to
> > provide the same actions for media. As for the example of content,
> 
> If you want to be compatible with RFC2543 clients, then if you muck
> with From/To, you need to change them back in responses. 

Right and this course would be required between client and proxy if this
function were implemented. I agree with you just trying to see if there
were a simple way to do this task.  

> This is not
> something which a proxy, as defined in the RFC, would do. That's not
> to say you couldn't build a device which looked remarkably *like* a
> proxy in all respects apart from this. You might even call it a proxy,
> but in pedantic specification terms that would not be correct.

This is true, maybe a B2BUA could have the anonamizer feature and work
within the policy of the SP realm, since a B2BUA may already performing
additional functions similiar to this. Of course then the proxy will
need to forward outbound requests through the B2BUA. 

> 
> > I believe that, since we have been discussing this in terms of
> privacy
> > as it relates to an SP environment, the responsibility lies
> > with the SP.
> > A UA not within an SP realm can assert that they are anyone
> > they want to
> > be including anonymous, right? In an authenticated SP environment
> the
> > option for a UA to muck with identity would not be permitted
> > becuase it
> > wouldnt authenticate correctly/at all, would it?
> >
> 
> As I understand it, From is not used for authentication. A proxy which
> authenticates someone uses different fields and is not required then
> to check that the From field is 'correct' for the authenticated user.
> Of course you *could* do that and it just might be a feature that a
> Service Provider would want.

Ya, I just remembered its in the proxy-auth/auth headers that the
username/pw comes in to play for authentication...

> 
> Nevertheless, I would argue that in practice the SP still has some
> responsibility for this field, on the grounds that a 'subscriber
> privacy service' which did not modify this field would not be regarded
> as actually working by the subscriber.

Agree.

> 
> ...Mark

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A few days I polled the list about opinions on adopting the Reason
header as a WG effort.

The comments I received seem to be unamimously favorable, so I would say
we have a consensus.

This seems to be covered in the callcontrol charter task, so I suspect
we just need a milestone added.

Suggestions?

--
Dean


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Please comment ASAP -- I've been lax in getting these circulated.
Finals are due tomorrow.

Revisions will be posted at:

http://www.softarmor.com/sipwg/meets/ietf53/notes/minutes.html

as they are made.

---------------------------------

Minutes edited by Dean Willis from notes taken by Vijay Gurbani, 
Joerg Ott, and Renee Cohen.

Session 1
	
19:30 CST meeting started 
     Agenda accepted 
     Chairs discuss "Note Well" 2026 notice
	

Work Plan update:

     We have revised the SIP spec; went to IESG; resulted in a "Yea"
     from IESG. SIP events spec is done

     Session timer is back to haunt us -- goes back to authors for bis
     update.

     Caller pref - pending from author for bis. 

     Precondition extensions -- in WGLC now; in IESG by April SIP 

     Privacy spec from DCS in WGLC now, IESG by April 

     REFER Method; needs bis, security updates. IESG by May? No
     complaints from authors. More important since there are many
     implementations already.

     MESSAGE method ready for WG LC -- have not done it yet. 

     PATH method needs rev from editor. Call for volunteers. 

     NAT awareness: rev pending from author. 

     SIP Privacy and Security reqs to IESG? There is a feeling that
     the SIP extensions for privacy wrt to 3GPP is not achievable --
     Jon P: some privacy (user provided privacy vs. network provided
     privacy) nuances have not been captured as of today. Henning:
     that would make a lot of sense if we know what we wanted --
     another req document is not in and of itself useful. Dean: SIP
     privacy and security reqs predate the SIPPING WG -- IESG felt
     that it is such a critical piece that it should stay in SIP WG.

     SIP over SCTP? Gonzalo: we are ready for WG LC for
     SIP/SCTP. Dean: Also in July timeframe -- have a draft standard
     version of SIP -- #1 goal in July of 2003. Original SIP spec was
     2543, we want to claim new number of 3543 :-)

     State: Charter item of pushing certain things off the SIP
     signaling state -- state or cookies specification translated to
     SIP. Did we ever come to a consensus on how to go forward? Rohan:
     Cookies I-D is good for doing what you want to do with the state
     I-D and is more general. I support it. Dean: Anyone know of
     implementations? [No implementations yet] Jonathan Rosenberg: The
     reason no one has implemented it is because everyone is using R-R
     and Contact. It is not entirely clear to me that this is even
     needed. Flemming: The difference is that R-R requires the proxy to
     be in all signaling. Andrew Zmolek: We looked at R-R issue and it
     seemed that we were going to have some trouble -- either the
     state or the cookies would do fine JDR: R-R was not sufficient
     since there was no reliable way to put something and get it
     back. With the new R-R update, this is no longer an
     issue. rjsparks: You cannot change the state in a dialog, you can
     only push and get the state. JDR: Okay, if that is the
     requirement, then fine -- if the req is to just push state for
     the purpose of a dialog, we have a mechanism already in R-R,
     Contact. Brian Rosen: we may work on a requirement document
     offline, assuming that it is useful.

     General Scheduling: Henning: meta scheduling aspect -- can you
     get people involved long enough to remember what the issue was?
     Brian: Our goal is to get a lot of stuff out that has been
     hanging around for a long time -- but we do not want to get out
     12 LC I-Ds at the same time. We will revive the LC schedule we
     had going. Henning: Do other I-Ds that are not on your list wait
     until re- chartering? Brian: No. There are some things that are
     hanging around for a long time; as long as the ADs do not breath
     down on us, we will work on them as we go along. Henning: Need a
     priority list. Jonathan: Learn from successes -- Bundle 1 was a
     success delivery to 3GPP. With the current mechanism we were
     randomly LC'ing. One of the thing the Bundle did is to focus
     people's energy on that. Brian: We can consider that; Bundle 1's
     advantage was that the drafts were related to one another. These
     do not.

     What do we do with: sipping-conferencing-models? sip-3pcc?
     app-components? sip-vxml? Jonathan: need to finish bis update;
     but done as far as I know.  Rohan: 3PCC is a very pure usage
     draft describing the usage of baseline bis offer answer
     model. Most of the stuff above is usage or framework, we should
     get it done in SIPPING. Brian Rosen: You are basically saying
     what Jonathan said: Put it in Bundles. Rohan: Sure.
 

SIP Change process, Allison Mankin:

     This is not a SIP draft; it is an individual transport area
     draft. People have said that there is a need to control SIP
     information. The Replaces header was done in a freeform manner --
     this is not good. WG discipline needed over extensions of
     SIP. RFC 3261 (new bis) has an IANA consideration which is very
     different then what you have seen before. You need a standards
     track RFC for headers, method, response codes, warning codes. One
     place you do not need to do this is for the Events RFC -- just
     need WG yes for these. serverfeatures did not go too far with
     IESG since it offered unbridled extensions. We now have P-header
     (not X-header, more constrained). Still need a RFC, but you do
     not have to have the buy in of SIPPING or SIP. The string with P-
     is reserved if they are to have a future life. Henning: while I
     agree with the notion, the naming has the same problems that X
     headers had. Attaching a meaning to "P-" is not good. Allison:
     They do not have option tags and have to have applicability
     statements. Henning: There are 2 issues: naming and
     process/applicability. If we have a header name which was
     registered (say, foo); that header has the property that as long
     as it is in the non-RFC track, it looks like a normal header. If
     it reaches standards track, it retains its name. Lets say P-bar
     header becomes popular and widely implemented. Now, if P-bar
     header goes to standards track, they will have to rename this
     header. Allison: can you make a P-header a standards track header
     -- add an option tag. Dean: The P- name will still be registered
     and be useful. Now you will basically have to track P- and non-P
     extension headers -- makes the symbol table little large --
     that's okay. Dave Oran: feel uncomfortable in mixing naming
     conventions and algorithmic behavior. I do not like the idea of
     having to parse inside header string. Jonathan Rosenberg: The
     name of the option tag is unrelated to the name of the
     header. Henning: HTTP extension model is different then SIP
     extension model. There is no correlation in header names and
     option tags. Allison: If the I-D has any implications of this
     sort, we can fix it. Gonzalo: We have option tags that have no
     headers associated with them. Keith Drage: We need to make sure
     that this I-D does not contain any requirements on SIP
     implementation -- a SIP implementer must not have to read this
     I-D.

 

The UPDATE method - Jonathan Rosenberg :

     Open issues 

     1) Glare with PRACK - UPDATE only specifies glare resolution with
        itself. You can have glare with PRACK. Rejecting PRACK is
        bad. Solution: can't send UPDATE if you have sent an answer in
        18x for which you have not gotten a PRACK. Will put some words
        with general caveats.
    
     2) Repairable response codes -- automata can fix these without
        human intervention. What about 493 Undecipherable? May require
        user intervention to fix it. Proposal: include it, add text
        saying it retries if it would otherwise retry with that
        response. [No one objected].

     3) Generate 155 instead of 4xx MAY or SHOULD -- for backward
        compatibility. SHOULD is better if the UAS supports this
        capability, the proxy may not. This makes it work at proxies
        transparently. Comment: This text is screaming for a reason
        header, otherwise the UAC will have to infer what the problem
        is. JDR: I did not talk about reason header for a reason -- it
        is on a slower track. For the basic cases we are worried about
        immediately, the UAC can infer from the headers. Going
        forward, the reason header is the way to go. Gonzalo: The last
        review of the reason header already has this, so we can use
        it. Jonathan: Does the group agree that the message sip (or
        sip frag) is the appropriate approach? Or wait for the reason
        header (which is on a slower track). Rohan: Can we pull out
        155 out of this, then? [No consensus on this]

 

Manyfolks open issues (Gonzalo Camarillo)
draft-ietf-sip-manyfolks...-05.txt 

     We are now defining a framework for preconditions of different
     types. We define the current status of the precond vs. desired
     status. We always know if current status is better or worse then
     desired status. Two status types: e2e -- always present in
     manyfolks (-04). Segmented status type introduced in -04.

     Open issues: Meaning of Require: precondition -- 2 approaches: I
     refuse everything I don't understand. Or be liberal and accept
     the offer if the preconditions can be met without your
     intervention? Which is better? 1 or 2? Comment: you can have a
     thing called "criticality" which gives a hint on what to
     do. Gonzalo: I will decide after speaking to Mark.

 

Reason code, Gonzalo Camarillo.:

     Requirements: same functionality needed in several WG items --
     why is this request (or response) being sent?

     Useful in many works: ISUP/SIP mapping, in manyfolks
     (precondition failure, unacceptable here), HERFP, 3pcc.

     Jonathan: Throw in another use: in the event that you fork the
     request to a bunch of phones and one of them picks up. The proxy
     generates CANCEL. The reason for CANCEL is not because the user
     hung up, but because 1 of N answered.

     Rohan: Lot of overlap in the reqs that generated this document
     and request history.

     Eric Burger: This is H.450 all over again.

     Jonathan R: Don't we have an enumerated list of response code in
     the bis already? This is exactly that, and then some.

     Comment: we have one address space for responses, and we have
     just added one more with the Reason header. Has a kitchen-sink
     feeling to it.

     Henning: The motivation was exactly to prevent reinventing the
     same thing every time. We are not adding new error classes that
     will have to be percolated to all existing implementation. This
     is a fine grained status code which is there if you need
     it. Example: Q.850 error code will not be pertinent to many
     implementations, but to the one that it is pertinent to, it can
     use it without too much perturbation.

     Brian Rosen: What do we do now? 2 possibilities: crisp set of
     reqs, which are clear and this is a reasonable solution. Or we do
     not have a crisp set of reqs. We need to determine this
     first. Lot of discussion on if this does or does not solve the
     job. But do we know what the job is? Should we push this back
     into sipping and generate a req document? Those who think we have
     a sensible set of reqs and we can move forward? Those who need
     more reqs? [The hum level was 50-50, no consensus by humming on
     if we have the reqs captured right.]

     Dave Oran: Need hum on slightly different -- do we get involved
     in reqs that requires identity (being able to communicate why you
     are sending it to this particular party).

     Brian Rosen: That is a reasonable suggestion -- so considered. We
     will get the reqs out before Yokohama and bring the solution out
     before then. Those of you who hummed against it should
     participate in the list when we discuss this on it.

 

 

Flemming Andreasen, SIP Extensions for Media Authorization.
draft-ietf-sip-call-auth-04.txt :

     Changes: Category is informational -- Header is now
     P-Media-Authorization. Applicability statement about appropriate
     use (SIP Proxy and Policy Server (PDP) must belong to the same
     domain) Updated rules about when to add a P-Media-Authorization
     header. Additional security considerations -- don't encrypt
     message bodies (proxies need to examine them).

     Open issues: None known (authors list need to be trimmed),
     currently in WGLC.

     Jonathan: I will send you some minor reviews. More look-see
     needed in the security section. This token is about media
     authorization -- authorization follows authentication. DOes the
     I-D point out this issue?

     Flemming: You do not necessarily need to authenticate before
     authorization. Some entity has been given an authorization token
     to access some resource.

     Jonathan: More discussion maybe needed on the security section --
     you are giving a token to a party that you may not have
     authenticated. If that is your model, fine; a couple of sentences
     would probably suffice in the I-D.

     Brian Rosen: The WGLC is going to get over, if anyone wants to
     raise more issues please do so. It fills the needs, even though
     it has a lot of limits. LC be it, we will move forward.

 

 

Ben Campbell, SIP Extensions for IM :

     -05 draft; recent changes -- remove CPIM mapping to separate
     draft. Would like to include in 3rd bundle to IESG.

     Highlights - sends IM; does not initiate a dialog, does not
     discuss message sessions; actual message in bodies.

     Open issues: No recent discussions. Needs minor editorial changes
     (forking, threading -- couple of sentences). Anything else? Is it
     ready for LC? One more revision -- no change in substance, more
     editorial.

     Brian Rosen: Ok, as soon as you have the revision, we will post
     it as LC.

     
Closing Remarks :

     Brian Rosen: Administering this list is no fun -- people forward
     their email to accounts that consistently run over quota. The
     list is setup so that only subscribers can post.

21:29 CST - WG adjourned. 



SIP Session 2, 53rd IETF 

Start 13:06 CST 

      Added AKA Digest and Path discussion to the Agenda. 
      Agenda accepted. 

Digest based authentication -- James Undery, Ubiquity 

      Quick run through of improvements to Digest
      authentication. UAC->UAS auth, UAC->Proxy authentication
      supported in the SIP spec. Our draft adds Proxy->UAS auth.,
      bid-down protection, mutual auth., integrity. Added 3 new
      headers and 1 new response (492) for Proxy->UAS authentication
      Bid-down protection: prefix added to nonces, protects scheme and
      quality of detection.

      Open issues: 1) No algorithm protection - if a hashing algo is
      broken, we need algo revocation. Proposal: make limitation
      explicit and rule that algo revocation is out of scope. 2) No
      negotiation of body integrity protection -- Proxies can't alter
      message bodies; Proposal : leave unchanged 3) No protection
      against weak passwords: Proposal - make limitation explicit, the
      solution is out of scope. 4) Client side can't initiate
      authentication 5) Forking and response collation issues - can't
      guarantee upstream entities see the challenges; response
      collation oriented towards success. Proposal: make limitations
      explicit.

      Jon: What are you trying to accomplish as a UAS by
      authenticating your upstream proxy? James: You may trust the
      proxy but not the link between the proxy and the UAS (for
      example: a radio interface). Jon: So this is the integrity
      process, not an authentication one. James: Authentication and
      integrity are closely linked. Rohan: We need to think about if
      this is needed? Looks like it is solved by TLS anyway. We need
      to understand under what circumstances we will use this
      approach. Jonathan: With this you do not know which proxy -- one
      up or 2 up -- you want to authenticate.

      Comment: You mention weak passwords; there are no strong
      passwords. You are unlikely to create a password that is not
      uncrackable, regardless of them appearing in the dictionary or
      not. Relying on a long random key is of no help. Christian: I
      would like to reinforce this point. Digest should be combined
      with strong authentication of the server, not on its
      own. Henning: when people talk about passwords, I wouldn't think
      that this is human generated; it is random string generated by
      some automata.

      Dean: Are we going to close any of these today? If not, let's
      take this to the mailing list. Brian: This has been hanging on
      for a long time; are we going to extend digest or not fortify it
      anymore. I do not know how to go ahed. People are saying that
      digest is terrible, but others are saying that it is still
      useful. I do not see other alternatives on the floor. Christian:
      2 forms with digest: 1 is if you are sending it as cleartext. 2
      is when you are doing digest with a 3rd party you have not
      authenticated. Simple thing for us is to use digest only for
      REGISTER not anything else. Brian: But there is no other
      solution on the table. Allison: The security review for bis came
      out as digest being a very lightweight way to do user
      authentication is okay. There is a good possibility of taking
      some time over this and making it better; maybe we should say
      that this document is not a WG document. We should discuss for
      the charter document something that exploits S/MIME. Rohan:
      There is some stuff in James' draft we can get consensus
      quickly; others may take some time. Allison: There is a huge
      deployment of digest. MD5 digest is known weak, but is not going
      to be thrown out. The topic we have now is not extending digest,
      but supporting some other password (a la AKA). For this
      document, consider what requirements it is meeting? Henning: One
      thing desperately needed in digest is registrar
      authentication. If we do something with digest at all, it must
      be this. Authenticating previous hops is nice, but not a known
      vulnerability we need solve now. Steven (3GPP): We do have a
      basic need to protect last hop integrity. We need to know what
      the intentions of the WG are towards digest. We need the
      direction very soon otherwise we are in a bind. Allison: Maybe,
      as Steven said, IPSec could be used for the short duration --
      could be the right way of meeting the requirement. Maybe we can
      have 5 minutes on some other agenda to talk about this. Brian:
      We are not getting anywhere -- let's move on and take it to
      list.


Digest AKA Authentication - Aki Nieme 

      AKA is a shared secret based auth that uses a smart card like
      device. Previous proposal (...-eap-01.txt) got good reception at
      SLC.

      Digest AKA reuses the digest scheme and uses the AKA parameters
      as input to the digest mechanism. AKA generates "one-time"
      passwords for Digest.

      Issues: 1) "Choke point" attack - similar to the weak password
      attack. 2) Should we adopt draf-niemi-sipping-digest-aka-00? It
      provides message integrity and is complementary to vanilla
      digest used today.

      Future: Will this become a work item for SIP WG? RFC category?
      There is some time pressure since 3GPP R5 is coming up. Can
      draft-niemi-digest-aka-00.txt be adopted as a solution?

      Allison: This does not involve any SIP extensions; just the
      extension of HTTP methods. It is good to get the SIP people's
      knowledge. There is no need for it to be a WG document; you can
      take it to RFC as an individual submission. Brian: Is there
      sufficient interest in the group to make it a WG item, or
      continue as an individual submission? [Took hum; the hum for
      people who want to make it a WG document prevailed]. Miguel
      Garcia: why use a 3G specific technology which has no broad
      impact on the Internet? Adam seconded. [The chairs agreed that
      we may have to revisit this issue again]


Security Negotiation open issues, Jari Akko 

     Presented issues, asked what to do next. Brian: Anyone object to
     NOT go forward with this? [No one objected; this is part of SIP
     WG]

SIP Extensions for Network Asserted Caller Identity and Privacy within
trusted networks - Flemming Andreasen

     Good list discussion 1 month ago; currently in WG LC. There have
     been some offline comment that have not been incorporated in the
     I-D yet.

     Overview of changes: applicability statement (only suitable in
     the same admin domain, draft is for network-asserted identity,
     not user-asserted). Anonymity header got removed. Grammar fixes
     to be consistent with -09 bis.

     Open issues: 1) Proxy handling of RP-ID received from untrusted
     entity (proxy or UA) 2 options: 1) if verifiable, set screen=yes
     2) always remove untrusted RP-ID Option 1 seems more general then
     2; Recommendation: Option 1.

     This generated a lot of discussion, mostly revising around
     policies vs. protocols, network asserted identities vs. user
     asserted identities, (in)security of this I-D and why it will not
     be acceptable to IESG, and this being more for the benefit of
     3GPP.

     Randy Bush discussed current unacceptability of draft to
     operations directorate, and agreed to Send Text.


_______________________________________________
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From daemon@optimus.ietf.org  Thu Apr 11 22:51:30 2002
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        "Peterson, Jon" <jon.peterson@neustar.biz>,
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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inline.

Mark Watson wrote:
> 
> Just on this one issue of obfuscation of the From/To field....
> 
> Let's be really clear about why this has come about:
> 
> Some Service Providers *believe* in the following proposition:
> 
> (*) IF the UA is allowed to put identity information into From/To, THEN
> the Service Provider will be forced (by Data Protection) to provide
> services which modify these fields, in order to protect the privacy of
> people other than the actual calling user - subscriber, forwarding etc.
> 
> First, set aside the question of whether to agree with (*) or not.
> 
> There are three options for people with this belief:
> 1) *Require* permanantly that the UA does not put identity information
> into From/To
> 2) Reject requests where the UA puts identity information in From/To if
> privacy is required
> 3) Build network services which provide the necessary modification
> (+whatever else is needed)
> 
> I do not think there are any other options.
> 
> We all agree that (1) is bad. (2) looks attractive but is not backwards
> compatible, requires a double request (not good for wireless) & has
> issues as to whether you trust the UA (in redirection scenarios).
> 
> I was looking for a consensus that (3) is a possible and valid thing to
> do for these Service Providers. If the IETF SIP group agree that (3) and
> not (1) is _a_ valid solution to this problem, then there is more chance
> of other fora adopting (3) instead of (1).

Honestly, I am flattered that you are hoping for the IETF blessing for
building a service. However, building such an anonyzmizer in (3) is well
within the scope of the SIP protocol, and requires no standardization
activity or consensus from IETF. I personally believe that you need a
full b2bua, since privacy, from a user perspecitve, WILL involve
stripping everything which might reveal identity, no matter whether the
header formally has identity information or not. Thus, I am not sure
what there is to do, but to emphasize to our friends in 3gpp to NOT
always put anonymous values in the To/From as this clearly will break
all reasonable interop with the outside world. (1) is BAD. Does anyone
dispute that?


-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 11 23:16:42 2002
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Date: Thu, 11 Apr 2002 21:49:57 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sip] Header name in draft-willis-sip-path-02.txt, was: I-D  
 ACTION:draft-illis...
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Dean Willis wrote:
> 
> > Short forms are still allowed, but restricted to one
> > character. THe idea is that they are allocated to headers
> > used frequently. It may all be moot with compression, but I
> > agree with Anders not to use a long name if we can avoid it.
> 
> So how about we just name it RRR?

I don't really care what we call it; not even sure why Path is no longer
OK.... 

-Jonathan R.

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

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From daemon@optimus.ietf.org  Fri Apr 12 03:22:36 2002
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	Thu, 11 Apr 2002 22:56:35 -0700
Date: Thu, 11 Apr 2002 22:56:35 -0700
From: Bobby Sardana <sardana@obsoft.com>
Message-Id: <200204120556.g3C5uZ016852@obsoft.com>
To: sip@ietf.org, vikas_sip@yahoo.com
Subject: Re: [Sip] Call Flows for Forking
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vikas_sip@yahoo.com wrote:

>I'm looking for some call flow examples detailing the forking 
>scenario. Can some one please guide me to any such ID, document 
>or website?

Try the following:

a. http://www.cs.columbia.edu/sip  <-- To find any specific drafts
b. http://www.ietf.org/internet-drafts/draft-ietf-sipping-call-flows-00.txt

regards,

Bobby Sardana.
sardana@obsoft.com

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From daemon@optimus.ietf.org  Fri Apr 12 06:39:12 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
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	 <jon.peterson@neustar.biz>,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Dean wrote:
> 
> Just to be a "devils advocate",  I suggest other alternatives:
> 
> 4) Require that the UA anonymize the To and From headers IF requesting
> CLIR on a call, and allow them to populate those headers as 
> desired when
> NOT requesting CLIR.
>

Of course we should do this anyway, but it does not solve the problem. The
proposition (*) was about personal data of people *other than the caller*
e.g. the subscriber. The UA does not necessarily have the information about
these people's privacy requirements unless we invent a way of getting it
there, and then enforcing its use.

> And the "modest suggestion" alternative:
> 
> 5) Use a non-Internet protocol which completely constrains the service
> functionality and user experience to a least common 
> denominator and stop
> pretending that your requirements have anything to do with 
> the Internet.
> 
> 
> Look, there are two reasons why we may want to use Internet protocols:
> 
> 1) Because Internet protocols and approaches offer a substantially
> richer, more flexible alternative which provides compelling advantages
> in terms of service delivery options relative to traditional PSTN
> approaches.
> 
> 2) Because we're jealous of the success the Internet has had 
> and want to
> contaminate it with "telco like" requirements so that Internet people
> can be miserable too.
> 
> I don't know about you, but I believe in #1, and would argue 
> that if we
> accept (*) as a requirement, that we're leading down the 
> slippery slope
> towards #2, leading to what will be a large push from certain segments
> (witness Path discussion) to adopt alternative #5. Alternative #4 is a
> reasonable compromise which keeps the responsibility were the Internet
> says the responsibility should be, in the hands of the user.

It's not about 'telco like' vs 'Internet like'. It's not about the
technology at all. We have to look at the *service* as it is supplied to the
user. If you have a Service Provider model at all, then the Service Provider
has responsibilities about what they do with people's Personal Data, whether
the service it implemented using IPv20 between nano-machines or over a wet
piece of string.


> Perhaps instead of To: and From: we should have called these headers
> "What-I-will-Call-You:" and "What-I-Want-You-To-Call-Me:". 
> These headers
> have ABSOLUTELY NO SEMANTIC VALUE at the signaling and routing level.
> They are HUMAN SUPPLIED values intended for HUMAN 
> CONSUMPTION. There are
> many other human-supplied values possible in the signaling, 
> and the risk
> of privacy violation is equally present on ALL of them.
> 

As I've argued before, a privacy service which pupports to protect the
subscriber's privacy, but which does not amend the
'What-I-Want-You-To-Call-Me:' field, will not be very impressive to the
subscriber. Especially if there are 'guidelines' indicating that the user
should put their identity in this field so that *in practice*, Personal Data
of the subscriber appears in this field 99% of the time.

Now I'm just putting this as a proposition. I don't know what view Data
Protection regulators would take on this. But I think from a common-sense
analysis there is a good chance that privacy services might reasonably be
expected to modify such a field, since otherwise the service is only 1%
effective in practice.

It is actually more important 'what works' than philosphical arguments about
whose responsibility it is.

The SP does have some responsibility if they have required that the UA
follow a certain spec, which recommends certain behaviour.

By contrast, there is no implication whatsoever that there should be
presonal data in the subject field. I think that a privacy service which did
not modify the subject field is less likely to cause problems, although of
course there may well be people who want that blotted out too - the
user/subscriber can choose this at the price of loss of service
transparency.


> If we earnestly believe that Data Protection is going to require that
> we, by policy, alter these human-supplied/human-consumed values, then
> we're logically going to have to do the same for every other
> human-supplied/human-consumed values. I would argue that this 
> completely
> defeats the value of an Internet protocol in this context.
> 

We don't know for certain what Data Protection will/won't require, but I
believe we have to be prepared for all reasonable eventualities. My
contention is just that it is sufficiently likely that privacy services will
be required to modify these fields that we ought to consider it. Mayby it
will never be required, in which case good, but I would like us to agree
that if it *is* required, then there is an agreed way to do it.


> I believe it is significantly more effective to provide guidelines on
> human-supplied/human consumed values, and implement hard requirements
> for data protection only on network-supplied values.
>

I agree with this, with the caveat that from a Service perspective there are
grey areas about who is responsible for a particular piece of information. A
Service Provider may have some responsibility for apparently user-supplied
information if it is implicit or required in the service that the user shall
supply a certain piece of information in that field. Whether the network
equipment checks/processes etc. that field seems irrelevant to me in this
respect.

...Mark
 
> --
> Dean
> 
> 

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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<BR>

<P><FONT SIZE=3D2>Dean wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Just to be a &quot;devils advocate&quot;,&nbsp; =
I suggest other alternatives:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 4) Require that the UA anonymize the To and =
From headers IF requesting</FONT>
<BR><FONT SIZE=3D2>&gt; CLIR on a call, and allow them to populate =
those headers as </FONT>
<BR><FONT SIZE=3D2>&gt; desired when</FONT>
<BR><FONT SIZE=3D2>&gt; NOT requesting CLIR.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>Of course we should do this anyway, but it does not =
solve the problem. The proposition (*) was about personal data of =
people *other than the caller* e.g. the subscriber. The UA does not =
necessarily have the information about these people's privacy =
requirements unless we invent a way of getting it there, and then =
enforcing its use.</FONT></P>

<P><FONT SIZE=3D2>&gt; And the &quot;modest suggestion&quot; =
alternative:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 5) Use a non-Internet protocol which completely =
constrains the service</FONT>
<BR><FONT SIZE=3D2>&gt; functionality and user experience to a least =
common </FONT>
<BR><FONT SIZE=3D2>&gt; denominator and stop</FONT>
<BR><FONT SIZE=3D2>&gt; pretending that your requirements have anything =
to do with </FONT>
<BR><FONT SIZE=3D2>&gt; the Internet.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Look, there are two reasons why we may want to =
use Internet protocols:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1) Because Internet protocols and approaches =
offer a substantially</FONT>
<BR><FONT SIZE=3D2>&gt; richer, more flexible alternative which =
provides compelling advantages</FONT>
<BR><FONT SIZE=3D2>&gt; in terms of service delivery options relative =
to traditional PSTN</FONT>
<BR><FONT SIZE=3D2>&gt; approaches.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 2) Because we're jealous of the success the =
Internet has had </FONT>
<BR><FONT SIZE=3D2>&gt; and want to</FONT>
<BR><FONT SIZE=3D2>&gt; contaminate it with &quot;telco like&quot; =
requirements so that Internet people</FONT>
<BR><FONT SIZE=3D2>&gt; can be miserable too.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I don't know about you, but I believe in #1, =
and would argue </FONT>
<BR><FONT SIZE=3D2>&gt; that if we</FONT>
<BR><FONT SIZE=3D2>&gt; accept (*) as a requirement, that we're leading =
down the </FONT>
<BR><FONT SIZE=3D2>&gt; slippery slope</FONT>
<BR><FONT SIZE=3D2>&gt; towards #2, leading to what will be a large =
push from certain segments</FONT>
<BR><FONT SIZE=3D2>&gt; (witness Path discussion) to adopt alternative =
#5. Alternative #4 is a</FONT>
<BR><FONT SIZE=3D2>&gt; reasonable compromise which keeps the =
responsibility were the Internet</FONT>
<BR><FONT SIZE=3D2>&gt; says the responsibility should be, in the hands =
of the user.</FONT>
</P>

<P><FONT SIZE=3D2>It's not about 'telco like' vs 'Internet like'. It's =
not about the technology at all. We have to look at the *service* as it =
is supplied to the user. If you have a Service Provider model at all, =
then the Service Provider has responsibilities about what they do with =
people's Personal Data, whether the service it implemented using IPv20 =
between nano-machines or over a wet piece of string.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; Perhaps instead of To: and From: we should have =
called these headers</FONT>
<BR><FONT SIZE=3D2>&gt; &quot;What-I-will-Call-You:&quot; and =
&quot;What-I-Want-You-To-Call-Me:&quot;. </FONT>
<BR><FONT SIZE=3D2>&gt; These headers</FONT>
<BR><FONT SIZE=3D2>&gt; have ABSOLUTELY NO SEMANTIC VALUE at the =
signaling and routing level.</FONT>
<BR><FONT SIZE=3D2>&gt; They are HUMAN SUPPLIED values intended for =
HUMAN </FONT>
<BR><FONT SIZE=3D2>&gt; CONSUMPTION. There are</FONT>
<BR><FONT SIZE=3D2>&gt; many other human-supplied values possible in =
the signaling, </FONT>
<BR><FONT SIZE=3D2>&gt; and the risk</FONT>
<BR><FONT SIZE=3D2>&gt; of privacy violation is equally present on ALL =
of them.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>As I've argued before, a privacy service which =
pupports to protect the subscriber's privacy, but which does not amend =
the 'What-I-Want-You-To-Call-Me:' field, will not be very impressive to =
the subscriber. Especially if there are 'guidelines' indicating that =
the user should put their identity in this field so that *in practice*, =
Personal Data of the subscriber appears in this field 99% of the =
time.</FONT></P>

<P><FONT SIZE=3D2>Now I'm just putting this as a proposition. I don't =
know what view Data Protection regulators would take on this. But I =
think from a common-sense analysis there is a good chance that privacy =
services might reasonably be expected to modify such a field, since =
otherwise the service is only 1% effective in practice.</FONT></P>

<P><FONT SIZE=3D2>It is actually more important 'what works' than =
philosphical arguments about whose responsibility it is.</FONT>
</P>

<P><FONT SIZE=3D2>The SP does have some responsibility if they have =
required that the UA follow a certain spec, which recommends certain =
behaviour.</FONT></P>

<P><FONT SIZE=3D2>By contrast, there is no implication whatsoever that =
there should be presonal data in the subject field. I think that a =
privacy service which did not modify the subject field is less likely =
to cause problems, although of course there may well be people who want =
that blotted out too - the user/subscriber can choose this at the price =
of loss of service transparency.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; If we earnestly believe that Data Protection is =
going to require that</FONT>
<BR><FONT SIZE=3D2>&gt; we, by policy, alter these =
human-supplied/human-consumed values, then</FONT>
<BR><FONT SIZE=3D2>&gt; we're logically going to have to do the same =
for every other</FONT>
<BR><FONT SIZE=3D2>&gt; human-supplied/human-consumed values. I would =
argue that this </FONT>
<BR><FONT SIZE=3D2>&gt; completely</FONT>
<BR><FONT SIZE=3D2>&gt; defeats the value of an Internet protocol in =
this context.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>We don't know for certain what Data Protection =
will/won't require, but I believe we have to be prepared for all =
reasonable eventualities. My contention is just that it is sufficiently =
likely that privacy services will be required to modify these fields =
that we ought to consider it. Mayby it will never be required, in which =
case good, but I would like us to agree that if it *is* required, then =
there is an agreed way to do it.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; I believe it is significantly more effective to =
provide guidelines on</FONT>
<BR><FONT SIZE=3D2>&gt; human-supplied/human consumed values, and =
implement hard requirements</FONT>
<BR><FONT SIZE=3D2>&gt; for data protection only on network-supplied =
values.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

<P><FONT SIZE=3D2>I agree with this, with the caveat that from a =
Service perspective there are grey areas about who is responsible for a =
particular piece of information. A Service Provider may have some =
responsibility for apparently user-supplied information if it is =
implicit or required in the service that the user shall supply a =
certain piece of information in that field. Whether the network =
equipment checks/processes etc. that field seems irrelevant to me in =
this respect.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; Dean</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>,
        "'Dean Willis'" <dean.willis@softarmor.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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> > I was looking for a consensus that (3) is a possible and 
> valid thing to
> > do for these Service Providers. If the IETF SIP group agree 
> that (3) and
> > not (1) is _a_ valid solution to this problem, then there 
> is more chance
> > of other fora adopting (3) instead of (1).
> 
> Honestly, I am flattered that you are hoping for the IETF blessing for
> building a service. However, building such an anonyzmizer in 
> (3) is well
> within the scope of the SIP protocol, and requires no standardization
> activity or consensus from IETF. I personally believe that you need a
> full b2bua, since privacy, from a user perspecitve, WILL involve
> stripping everything which might reveal identity, no matter 
> whether the
> header formally has identity information or not. 

I'm glad to hear you say this. Of course an individual could not expect free
consultancy from the IETF as to whether a given service worked, was valuable
etc. But here we are talking about cooperation between standards fora, and I
think it is valid to ask the IETF SIP community for their opinion on how a
particular problem in another forum could or ought to be solved. If there
was consensus of opinion that would be a good thing too.

Others have said, variously:
i) Privacy services SHOULD NOT require modification of these fields, because
they are end-to-end and none of the business of the network
ii) A better solution to the b2bua one is to feed back the privacy
requirements to the UA and then reject requests which do not meet the
privacy requirements

I think we can certainly say that (i) is a matter of opinion. I for one do
not feel confident that it would be safe for a Service Provider to bet on
this with respect to Data Protection.

(ii) requires standardisation and IMO doesn't quite work anyway.

> Thus, I am not sure
> what there is to do, but to emphasize to our friends in 3gpp to NOT
> always put anonymous values in the To/From as this clearly will break
> all reasonable interop with the outside world. (1) is BAD. Does anyone
> dispute that?
> 

I haven't seen anyone dispute that here. My point is that to credibly assert
that (1) is bad, we have to describe an alternative. Does anyone dispute
that (3) is a valid alternative solution to this problem ?

...Mark

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<P><FONT SIZE=3D2>&gt; &gt; I was looking for a consensus that (3) is a =
possible and </FONT>
<BR><FONT SIZE=3D2>&gt; valid thing to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; do for these Service Providers. If the =
IETF SIP group agree </FONT>
<BR><FONT SIZE=3D2>&gt; that (3) and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; not (1) is _a_ valid solution to this =
problem, then there </FONT>
<BR><FONT SIZE=3D2>&gt; is more chance</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; of other fora adopting (3) instead of =
(1).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Honestly, I am flattered that you are hoping =
for the IETF blessing for</FONT>
<BR><FONT SIZE=3D2>&gt; building a service. However, building such an =
anonyzmizer in </FONT>
<BR><FONT SIZE=3D2>&gt; (3) is well</FONT>
<BR><FONT SIZE=3D2>&gt; within the scope of the SIP protocol, and =
requires no standardization</FONT>
<BR><FONT SIZE=3D2>&gt; activity or consensus from IETF. I personally =
believe that you need a</FONT>
<BR><FONT SIZE=3D2>&gt; full b2bua, since privacy, from a user =
perspecitve, WILL involve</FONT>
<BR><FONT SIZE=3D2>&gt; stripping everything which might reveal =
identity, no matter </FONT>
<BR><FONT SIZE=3D2>&gt; whether the</FONT>
<BR><FONT SIZE=3D2>&gt; header formally has identity information or =
not. </FONT>
</P>

<P><FONT SIZE=3D2>I'm glad to hear you say this. Of course an =
individual could not expect free consultancy from the IETF as to =
whether a given service worked, was valuable etc. But here we are =
talking about cooperation between standards fora, and I think it is =
valid to ask the IETF SIP community for their opinion on how a =
particular problem in another forum could or ought to be solved. If =
there was consensus of opinion that would be a good thing =
too.</FONT></P>

<P><FONT SIZE=3D2>Others have said, variously:</FONT>
<BR><FONT SIZE=3D2>i) Privacy services SHOULD NOT require modification =
of these fields, because they are end-to-end and none of the business =
of the network</FONT></P>

<P><FONT SIZE=3D2>ii) A better solution to the b2bua one is to feed =
back the privacy requirements to the UA and then reject requests which =
do not meet the privacy requirements</FONT></P>

<P><FONT SIZE=3D2>I think we can certainly say that (i) is a matter of =
opinion. I for one do not feel confident that it would be safe for a =
Service Provider to bet on this with respect to Data =
Protection.</FONT></P>

<P><FONT SIZE=3D2>(ii) requires standardisation and IMO doesn't quite =
work anyway.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Thus, I am not sure</FONT>
<BR><FONT SIZE=3D2>&gt; what there is to do, but to emphasize to our =
friends in 3gpp to NOT</FONT>
<BR><FONT SIZE=3D2>&gt; always put anonymous values in the To/From as =
this clearly will break</FONT>
<BR><FONT SIZE=3D2>&gt; all reasonable interop with the outside world. =
(1) is BAD. Does anyone</FONT>
<BR><FONT SIZE=3D2>&gt; dispute that?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I haven't seen anyone dispute that here. My point is =
that to credibly assert that (1) is bad, we have to describe an =
alternative. Does anyone dispute that (3) is a valid alternative =
solution to this problem ?</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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From daemon@optimus.ietf.org  Fri Apr 12 08:48:02 2002
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From: William Marshall <wtm@research.att.com>
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To: jdrosen@dynamicsoft.com, mwatson@nortelnetworks.com
Cc: sip@ietf.org
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Jonathan wrote, in part:
> building such an anonyzmizer in (3) is well
> within the scope of the SIP protocol, and requires no standardization
> activity or consensus from IETF.. full B2BUA.. <part snipped>  I am not sure
> what there is to do, but to emphasize to our friends in 3gpp to NOT
> always put anonymous values in the To/From as this clearly will break
> all reasonable interop with the outside world. (1) is BAD. Does anyone
> dispute that?

It has been stated in this group often that B2BUAs are BAD, and that B2BUAs
break services.  There has never yet been an example of a service that
is broken because of a B2BUA, but this belief seems generally accepted.

This is the first claim I've seen that putting anonymous strings in the
From and To headers also will break services.  I must admit I have more
trouble accepting this belief than the "B2BUA breaks services" belief.
An example of a service that this breaks would be even more useful.

But if both are true, then the question is: which is less bad?
I think this is a judgement call based on perceived gleanings
from crystal balls, and mine is cloudy.  Does someone else have 
a clearer crystal ball on the subject?

Bill Marshall
wtm@research.att.com

-----original message-----
Date: Thu, 11 Apr 2002 22:01:10 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: Mark Watson <mwatson@nortelnetworks.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft

inline.

Mark Watson wrote:
> 
> Just on this one issue of obfuscation of the From/To field....
> 
> Let's be really clear about why this has come about:
> 
> Some Service Providers *believe* in the following proposition:
> 
> (*) IF the UA is allowed to put identity information into From/To, THEN
> the Service Provider will be forced (by Data Protection) to provide
> services which modify these fields, in order to protect the privacy of
> people other than the actual calling user - subscriber, forwarding etc.
> 
> First, set aside the question of whether to agree with (*) or not.
> 
> There are three options for people with this belief:
> 1) *Require* permanantly that the UA does not put identity information
> into From/To
> 2) Reject requests where the UA puts identity information in From/To if
> privacy is required
> 3) Build network services which provide the necessary modification
> (+whatever else is needed)
> 
> I do not think there are any other options.
> 
> We all agree that (1) is bad. (2) looks attractive but is not backwards
> compatible, requires a double request (not good for wireless) & has
> issues as to whether you trust the UA (in redirection scenarios).
> 
> I was looking for a consensus that (3) is a possible and valid thing to
> do for these Service Providers. If the IETF SIP group agree that (3) and
> not (1) is _a_ valid solution to this problem, then there is more chance
> of other fora adopting (3) instead of (1).

Honestly, I am flattered that you are hoping for the IETF blessing for
building a service. However, building such an anonyzmizer in (3) is well
within the scope of the SIP protocol, and requires no standardization
activity or consensus from IETF. I personally believe that you need a
full b2bua, since privacy, from a user perspecitve, WILL involve
stripping everything which might reveal identity, no matter whether the
header formally has identity information or not. Thus, I am not sure
what there is to do, but to emphasize to our friends in 3gpp to NOT
always put anonymous values in the To/From as this clearly will break
all reasonable interop with the outside world. (1) is BAD. Does anyone
dispute that?


-Jonathan R.

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From daemon@optimus.ietf.org  Fri Apr 12 09:34:03 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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But (3) is achievable now with a B2BUA, and there is no standardization 
work necessary.  Jonathan correctly pointed out that it is non-trivial
to achieve what appears to be desired, and a B2BUA that thoroughly
scrubbed all fields, including relaying of media, will be needed.
No additional headers can help you.

I think we should stop worrying about this case, and try to focus on
something we can't reasonably do now, which is to provide a network
asserted identity when the UA itself provides anonymity.  That is a
case worth working on, and it represents the way I think it SHOULD work.
The UA is by far in the best position to determine the level of
anonymity it wishes, and if it can't get enough (because, for example,
it wants it's IP addresses obscured), it will have to employ a
B2BUA as above.  

Brian


-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Thursday, April 11, 2002 11:29 AM
To: 'Jonathan Rosenberg'
Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon; 'William
Marshall'
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Just on this one issue of obfuscation of the From/To field.... 
Let's be really clear about why this has come about: 
Some Service Providers *believe* in the following proposition: 
(*) IF the UA is allowed to put identity information into From/To, THEN the
Service Provider will be forced (by Data Protection) to provide services
which modify these fields, in order to protect the privacy of people other
than the actual calling user - subscriber, forwarding etc.
First, set aside the question of whether to agree with (*) or not. 
There are three options for people with this belief: 
1) *Require* permanantly that the UA does not put identity information into
From/To 
2) Reject requests where the UA puts identity information in From/To if
privacy is required 
3) Build network services which provide the necessary modification
(+whatever else is needed) 
I do not think there are any other options. 
We all agree that (1) is bad. (2) looks attractive but is not backwards
compatible, requires a double request (not good for wireless) & has issues
as to whether you trust the UA (in redirection scenarios).
I was looking for a consensus that (3) is a possible and valid thing to do
for these Service Providers. If the IETF SIP group agree that (3) and not
(1) is _a_ valid solution to this problem, then there is more chance of
other fora adopting (3) instead of (1).
So far, people have been arguing against the belief (*). The question is not
whether (*) is demonstrably true but whether there is sufficient *chance* of
it being true to justify serious consideration. A significant group of
Service Providers obviously think it is - they are worried they will be in
this situation and want a solution defined - that should be enough for us to
discuss solutions.
So, can we agree to recommend at least that people with belief (*) adopt
solution (3) and not solution (1) ? 
...Mark 



> -----Original Message----- 
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
> Sent: 11 April 2002 08:28 
> To: Watson, Mark [MDN05:EP10:EXCH] 
> Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon; 
> 'William Marshall' 
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft 
> 
> 
> 
> 
> Mark Watson wrote: 
> > 
> > All, 
> > 
> > I feel profoundly lucky that yesterday and Friday were 
> public holidays 
> > in the UK :-) 
> > 
> > Perhaps I can take advantage of this vantage point to offer 
> a summary of 
> > the thread. Protagonists please speak up if I have 
> mis-represented you 
> > below, but please don't re-open old fronts. 
> 
> Thanks for the summary. It is well needed. I suspect much of 
> the lack of 
> comment derives from the difficulty in following this long thread. 
> 
> I would kindly request all participants to please keep the discussions 
> technical. A lot of the emails in this thread were personal attacks. 
> Many more were threats along the general line of "do X or 
> standards body 
> Y will do Z". I really hate the latter ones. Let us hear no more of 
> either type of nonsense. It is not the IETF way. 
> 
> > 
> > 1) Call-Info 
> > 
> > Long arguments about the alledged similarities/differences between 
> > Call-Info and RPID. Suffice to say that the point has been made that 
> > Call-Info (and perhaps other things) could be abused to do 
> the things 
> > that RPID does, and in particular to do the 'Bad Things' 
> that some are 
> > worried about with RPID. The point of disagreement was 
> whether in the 
> > case of Call-Info, this would be 'major abuse', with RPID 
> was set up to 
> > make these 'Bad Things' easy, or whether there was some equivalence. 
> > 
> > The key point is that this kind of abuse of Call-Info may 
> happen if we 
> > do not provide an alternative. 
> 
> It was never our intent for Call-Info to provide any form of network 
> asserted identity that could be used in any useful way. Generally, the 
> idea was that insertion of Call-Info was either done by the UA, or in 
> the case of proxy-inserted values, was for "opt-in" services were the 
> user had signed up for content to be added. That said, there are 
> certainly cases where the caller may opt in for this service 
> but require 
> privacy for a particular call, and we need a mechanism to 
> support that. 
> 
> However, in no way would I support or advocate the use of Call-Info as 
> an alternative for R-P-ID. 
> 
> > 
> > 2) Abuse of From/To headers 
> > 
> > Jon is concerned that 'untrusted RPID' provides a standardised 
> > alternative to From/To, allowing Bad Things, like never 
> putting the real 
> > From/To information in the From/To fields. 
> 
> I think that the proper way to handle this, as pursued in 
> other threads, 
> is to provide a sane alternative to 3gpp so that they don't do this. I 
> believe we all agreed it would be bad. 
> 
> > 
> > Others have pointed out that the draft explicitly states 
> that untrusted 
> > UAs MUST NOT add RPID. However the draft explicitly described proxy 
> > handling of RPID headers from untrusted sources (proxies OR 
> UAs), and 
> > allows these to be propogated, marked as untrusted. 
> > 
> > Therefore, a UA which DID add RPID, although non-compliant 
> to the draft, 
> > would have its RPID propogated through the network. 
> > 
> > 3) 'Untrustworthy RPID' 
> > 
> > Flemming, Bill et al argued that RPID information is still 
> useful when 
> > it is 'untrustworthy'. This can occur legitimately within 
> the draft when 
> > an RPID which is not 'private' (and so not encrypted) 
> crosses a trust 
> > boundary. This would represent an identity asserted by Network A and 
> > passed to Network B, when B has no explicit trust 
> relationship with A. 
> > 
> > The contention was that since this information still has 
> some value, it 
> > should be passed to the UAS for display, and might also have some 
> > application for call trace, although obviously not as useful as a 
> > 'trustworthy' identity. 
> > 
> > Jon argued that the From field provides well enough for 
> display, and the 
> > value of untrustworthy information for call trace is 
> debatable (and has 
> > been debated at length). 
> 
> IMHO, 'untrusted RPID' is identical to the From, except that it is 
> purportedly inserted by a network element, whereas From is purportedly 
> inserted by the end user. Neither provides any guarantees. The essence 
> of the debate, I think, is whether you believe there is value 
> to the end 
> user in a field which is the identity purportedly inserted by some 
> network element.  Since there is apparently value to the identity 
> purportedly inserted by the end user, I see no reason why we should 
> discriminate. 
> 
> -Jonathan R. 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue 
> Chief Scientist                         First Floor 
> dynamicsoft                             East Hanover, NJ 07936 
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050 
> http://www.jdrosen.net                  PH:  (973) 952-5000 
> http://www.dynamicsoft.com 
> 

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From daemon@optimus.ietf.org  Fri Apr 12 09:56:53 2002
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From: William Marshall <wtm@research.att.com>
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Date: Fri, 12 Apr 2002 09:02:58 -0400 (EDT)
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To: Brian.Rosen@marconi.com
Cc: sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Brian Rosen wrote:
> I think B2BUAs are great for some things.
> 
> I suspect that an anoymizer B2BUA is a very good thing.
> 
> To me, the service that breaks with anonymous From/To is
> the ability to call me.  If my phone rings, and the "From"
> which is displayed says "anonymous", I'm not going to answer it.
> I'll probably instruct my voicemail service to do the same.
> The Police Tip line might very well accept such calls however.

Agree on all points.

But if the best example of a "broken service" is equivalent to
the claim that "my deleting of unopened email is a
failure of the email delivery system"  ???  This is somehow a
different category than call transfers (for instance) failing.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'William Marshall'" <wtm@research.att.com>, jdrosen@dynamicsoft.com,
        mwatson@nortelnetworks.com
Cc: sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 08:44:32 -0400

I think B2BUAs are great for some things.

I suspect that an anoymizer B2BUA is a very good thing.

To me, the service that breaks with anonymous From/To is
the ability to call me.  If my phone rings, and the "From"
which is displayed says "anonymous", I'm not going to answer it.
I'll probably instruct my voicemail service to do the same.
The Police Tip line might very well accept such calls however.

I'm reminded of the practices of some Hollywood types that
block Caller-Id, but won't accept calls that don't provide
Caller-Id!

Brian

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: Friday, April 12, 2002 7:17 AM
> To: jdrosen@dynamicsoft.com; mwatson@nortelnetworks.com
> Cc: sip@ietf.org
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Jonathan wrote, in part:
> > building such an anonyzmizer in (3) is well
> > within the scope of the SIP protocol, and requires no 
> standardization
> > activity or consensus from IETF.. full B2BUA.. <part 
> snipped>  I am not sure
> > what there is to do, but to emphasize to our friends in 3gpp to NOT
> > always put anonymous values in the To/From as this clearly 
> will break
> > all reasonable interop with the outside world. (1) is BAD. 
> Does anyone
> > dispute that?
> 
> It has been stated in this group often that B2BUAs are BAD, 
> and that B2BUAs
> break services.  There has never yet been an example of a service that
> is broken because of a B2BUA, but this belief seems generally 
> accepted.
> 
> This is the first claim I've seen that putting anonymous 
> strings in the
> From and To headers also will break services.  I must admit I 
> have more
> trouble accepting this belief than the "B2BUA breaks services" belief.
> An example of a service that this breaks would be even more useful.
> 
> But if both are true, then the question is: which is less bad?
> I think this is a judgement call based on perceived gleanings
> from crystal balls, and mine is cloudy.  Does someone else have 
> a clearer crystal ball on the subject?
> 
> Bill Marshall
> wtm@research.att.com
> 
> -----original message-----
> Date: Thu, 11 Apr 2002 22:01:10 -0400
> From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> To: Mark Watson <mwatson@nortelnetworks.com>
> Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
>         Flemming Andreasen <fandreas@cisco.com>,
>         "Peterson, Jon" <jon.peterson@neustar.biz>,
>         "'William Marshall'" <wtm@research.att.com>
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> inline.
> 
> Mark Watson wrote:
> > 
> > Just on this one issue of obfuscation of the From/To field....
> > 
> > Let's be really clear about why this has come about:
> > 
> > Some Service Providers *believe* in the following proposition:
> > 
> > (*) IF the UA is allowed to put identity information into 
> From/To, THEN
> > the Service Provider will be forced (by Data Protection) to provide
> > services which modify these fields, in order to protect the 
> privacy of
> > people other than the actual calling user - subscriber, 
> forwarding etc.
> > 
> > First, set aside the question of whether to agree with (*) or not.
> > 
> > There are three options for people with this belief:
> > 1) *Require* permanantly that the UA does not put identity 
> information
> > into From/To
> > 2) Reject requests where the UA puts identity information 
> in From/To if
> > privacy is required
> > 3) Build network services which provide the necessary modification
> > (+whatever else is needed)
> > 
> > I do not think there are any other options.
> > 
> > We all agree that (1) is bad. (2) looks attractive but is 
> not backwards
> > compatible, requires a double request (not good for wireless) & has
> > issues as to whether you trust the UA (in redirection scenarios).
> > 
> > I was looking for a consensus that (3) is a possible and 
> valid thing to
> > do for these Service Providers. If the IETF SIP group agree 
> that (3) and
> > not (1) is _a_ valid solution to this problem, then there 
> is more chance
> > of other fora adopting (3) instead of (1).
> 
> Honestly, I am flattered that you are hoping for the IETF blessing for
> building a service. However, building such an anonyzmizer in 
> (3) is well
> within the scope of the SIP protocol, and requires no standardization
> activity or consensus from IETF. I personally believe that you need a
> full b2bua, since privacy, from a user perspecitve, WILL involve
> stripping everything which might reveal identity, no matter 
> whether the
> header formally has identity information or not. Thus, I am not sure
> what there is to do, but to emphasize to our friends in 3gpp to NOT
> always put anonymous values in the To/From as this clearly will break
> all reasonable interop with the outside world. (1) is BAD. Does anyone
> dispute that?
> 
> 
> -Jonathan R.
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> 

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "Rosen, Brian"
	 <Brian.Rosen@marconi.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
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        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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>> The UA is by far in the best position to determine the level of 
>> anonymity it wishes, and if it can't get enough (because, for example, 
>> it wants it's IP addresses obscured), it will have to employ a 
>> B2BUA as above.  
> 
>Agreed. But the UA is not in a good position to determine the 
>subscribers anonymity requirements, or other parties not represented 
>by a UA. In a Service Provider model, these people have some rights. 
>So, the network may have to employ a B2BUA or whatever on behalf of 
>these people. There is no problem with this and so no case for 
>obfuscating the From/To fields - agreed ?

Agreed.  Don't forget the obvious problem of any crypto (signatures
for example) on any of these fields!

Brian


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'William Marshall'" <wtm@research.att.com>, jdrosen@dynamicsoft.com,
        mwatson@nortelnetworks.com
Cc: sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 08:44:32 -0400
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I think B2BUAs are great for some things.

I suspect that an anoymizer B2BUA is a very good thing.

To me, the service that breaks with anonymous From/To is
the ability to call me.  If my phone rings, and the "From"
which is displayed says "anonymous", I'm not going to answer it.
I'll probably instruct my voicemail service to do the same.
The Police Tip line might very well accept such calls however.

I'm reminded of the practices of some Hollywood types that
block Caller-Id, but won't accept calls that don't provide
Caller-Id!

Brian

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: Friday, April 12, 2002 7:17 AM
> To: jdrosen@dynamicsoft.com; mwatson@nortelnetworks.com
> Cc: sip@ietf.org
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Jonathan wrote, in part:
> > building such an anonyzmizer in (3) is well
> > within the scope of the SIP protocol, and requires no 
> standardization
> > activity or consensus from IETF.. full B2BUA.. <part 
> snipped>  I am not sure
> > what there is to do, but to emphasize to our friends in 3gpp to NOT
> > always put anonymous values in the To/From as this clearly 
> will break
> > all reasonable interop with the outside world. (1) is BAD. 
> Does anyone
> > dispute that?
> 
> It has been stated in this group often that B2BUAs are BAD, 
> and that B2BUAs
> break services.  There has never yet been an example of a service that
> is broken because of a B2BUA, but this belief seems generally 
> accepted.
> 
> This is the first claim I've seen that putting anonymous 
> strings in the
> From and To headers also will break services.  I must admit I 
> have more
> trouble accepting this belief than the "B2BUA breaks services" belief.
> An example of a service that this breaks would be even more useful.
> 
> But if both are true, then the question is: which is less bad?
> I think this is a judgement call based on perceived gleanings
> from crystal balls, and mine is cloudy.  Does someone else have 
> a clearer crystal ball on the subject?
> 
> Bill Marshall
> wtm@research.att.com
> 
> -----original message-----
> Date: Thu, 11 Apr 2002 22:01:10 -0400
> From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> To: Mark Watson <mwatson@nortelnetworks.com>
> Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
>         Flemming Andreasen <fandreas@cisco.com>,
>         "Peterson, Jon" <jon.peterson@neustar.biz>,
>         "'William Marshall'" <wtm@research.att.com>
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> inline.
> 
> Mark Watson wrote:
> > 
> > Just on this one issue of obfuscation of the From/To field....
> > 
> > Let's be really clear about why this has come about:
> > 
> > Some Service Providers *believe* in the following proposition:
> > 
> > (*) IF the UA is allowed to put identity information into 
> From/To, THEN
> > the Service Provider will be forced (by Data Protection) to provide
> > services which modify these fields, in order to protect the 
> privacy of
> > people other than the actual calling user - subscriber, 
> forwarding etc.
> > 
> > First, set aside the question of whether to agree with (*) or not.
> > 
> > There are three options for people with this belief:
> > 1) *Require* permanantly that the UA does not put identity 
> information
> > into From/To
> > 2) Reject requests where the UA puts identity information 
> in From/To if
> > privacy is required
> > 3) Build network services which provide the necessary modification
> > (+whatever else is needed)
> > 
> > I do not think there are any other options.
> > 
> > We all agree that (1) is bad. (2) looks attractive but is 
> not backwards
> > compatible, requires a double request (not good for wireless) & has
> > issues as to whether you trust the UA (in redirection scenarios).
> > 
> > I was looking for a consensus that (3) is a possible and 
> valid thing to
> > do for these Service Providers. If the IETF SIP group agree 
> that (3) and
> > not (1) is _a_ valid solution to this problem, then there 
> is more chance
> > of other fora adopting (3) instead of (1).
> 
> Honestly, I am flattered that you are hoping for the IETF blessing for
> building a service. However, building such an anonyzmizer in 
> (3) is well
> within the scope of the SIP protocol, and requires no standardization
> activity or consensus from IETF. I personally believe that you need a
> full b2bua, since privacy, from a user perspecitve, WILL involve
> stripping everything which might reveal identity, no matter 
> whether the
> header formally has identity information or not. Thus, I am not sure
> what there is to do, but to emphasize to our friends in 3gpp to NOT
> always put anonymous values in the To/From as this clearly will break
> all reasonable interop with the outside world. (1) is BAD. Does anyone
> dispute that?
> 
> 
> -Jonathan R.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org, Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 13:44:05 +0100
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Brian wrote:

> But (3) is achievable now with a B2BUA, and there is no 
> standardization 
> work necessary.  Jonathan correctly pointed out that it is non-trivial
> to achieve what appears to be desired, and a B2BUA that thoroughly
> scrubbed all fields, including relaying of media, will be needed.
> No additional headers can help you.
> 

I agree completely. Others claimed that the problem did not exist.

Also, there can be a range of privacy services, requiring more, or less,
field modification etc. You will at least need to modify From/To & probably
Call-Info, Reply-To etc.

> I think we should stop worrying about this case, and try to focus on
> something we can't reasonably do now, which is to provide a network
> asserted identity when the UA itself provides anonymity.  That is a
> case worth working on, and it represents the way I think it 
> SHOULD work.

Absolutely. I hope that this thread has demonstrated, though, that it is not
*because* of RPID that people have done Bad Things with From/To, more the
other way round.

> The UA is by far in the best position to determine the level of
> anonymity it wishes, and if it can't get enough (because, for example,
> it wants it's IP addresses obscured), it will have to employ a
> B2BUA as above.  
> 

Agreed. But the UA is not in a good position to determine the subscribers
anonymity requirements, or other parties not represented by a UA. In a
Service Provider model, these people have some rights. So, the network may
have to employ a B2BUA or whatever on behalf of these people. There is no
problem with this and so no case for obfuscating the From/To fields - agreed
?

...Mark

> Brian
> 
> 
> -----Original Message-----
> From: Mark Watson [mailto:mwatson@nortelnetworks.com]
> Sent: Thursday, April 11, 2002 11:29 AM
> To: 'Jonathan Rosenberg'
> Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, 
> Jon; 'William
> Marshall'
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Just on this one issue of obfuscation of the From/To field.... 
> Let's be really clear about why this has come about: 
> Some Service Providers *believe* in the following proposition: 
> (*) IF the UA is allowed to put identity information into 
> From/To, THEN the
> Service Provider will be forced (by Data Protection) to 
> provide services
> which modify these fields, in order to protect the privacy of 
> people other
> than the actual calling user - subscriber, forwarding etc.
> First, set aside the question of whether to agree with (*) or not. 
> There are three options for people with this belief: 
> 1) *Require* permanantly that the UA does not put identity 
> information into
> From/To 
> 2) Reject requests where the UA puts identity information in 
> From/To if
> privacy is required 
> 3) Build network services which provide the necessary modification
> (+whatever else is needed) 
> I do not think there are any other options. 
> We all agree that (1) is bad. (2) looks attractive but is not 
> backwards
> compatible, requires a double request (not good for wireless) 
> & has issues
> as to whether you trust the UA (in redirection scenarios).
> I was looking for a consensus that (3) is a possible and 
> valid thing to do
> for these Service Providers. If the IETF SIP group agree that 
> (3) and not
> (1) is _a_ valid solution to this problem, then there is more 
> chance of
> other fora adopting (3) instead of (1).
> So far, people have been arguing against the belief (*). The 
> question is not
> whether (*) is demonstrably true but whether there is 
> sufficient *chance* of
> it being true to justify serious consideration. A significant group of
> Service Providers obviously think it is - they are worried 
> they will be in
> this situation and want a solution defined - that should be 
> enough for us to
> discuss solutions.
> So, can we agree to recommend at least that people with 
> belief (*) adopt
> solution (3) and not solution (1) ? 
> ...Mark 
> 
> 
> 
> > -----Original Message----- 
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
> > Sent: 11 April 2002 08:28 
> > To: Watson, Mark [MDN05:EP10:EXCH] 
> > Cc: sip@ietf.org; Ben Campbell; Flemming Andreasen; Peterson, Jon; 
> > 'William Marshall' 
> > Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft 
> > 
> > 
> > 
> > 
> > Mark Watson wrote: 
> > > 
> > > All, 
> > > 
> > > I feel profoundly lucky that yesterday and Friday were 
> > public holidays 
> > > in the UK :-) 
> > > 
> > > Perhaps I can take advantage of this vantage point to offer 
> > a summary of 
> > > the thread. Protagonists please speak up if I have 
> > mis-represented you 
> > > below, but please don't re-open old fronts. 
> > 
> > Thanks for the summary. It is well needed. I suspect much of 
> > the lack of 
> > comment derives from the difficulty in following this long thread. 
> > 
> > I would kindly request all participants to please keep the 
> discussions 
> > technical. A lot of the emails in this thread were personal 
> attacks. 
> > Many more were threats along the general line of "do X or 
> > standards body 
> > Y will do Z". I really hate the latter ones. Let us hear no more of 
> > either type of nonsense. It is not the IETF way. 
> > 
> > > 
> > > 1) Call-Info 
> > > 
> > > Long arguments about the alledged 
> similarities/differences between 
> > > Call-Info and RPID. Suffice to say that the point has 
> been made that 
> > > Call-Info (and perhaps other things) could be abused to do 
> > the things 
> > > that RPID does, and in particular to do the 'Bad Things' 
> > that some are 
> > > worried about with RPID. The point of disagreement was 
> > whether in the 
> > > case of Call-Info, this would be 'major abuse', with RPID 
> > was set up to 
> > > make these 'Bad Things' easy, or whether there was some 
> equivalence. 
> > > 
> > > The key point is that this kind of abuse of Call-Info may 
> > happen if we 
> > > do not provide an alternative. 
> > 
> > It was never our intent for Call-Info to provide any form 
> of network 
> > asserted identity that could be used in any useful way. 
> Generally, the 
> > idea was that insertion of Call-Info was either done by the 
> UA, or in 
> > the case of proxy-inserted values, was for "opt-in" 
> services were the 
> > user had signed up for content to be added. That said, there are 
> > certainly cases where the caller may opt in for this service 
> > but require 
> > privacy for a particular call, and we need a mechanism to 
> > support that. 
> > 
> > However, in no way would I support or advocate the use of 
> Call-Info as 
> > an alternative for R-P-ID. 
> > 
> > > 
> > > 2) Abuse of From/To headers 
> > > 
> > > Jon is concerned that 'untrusted RPID' provides a standardised 
> > > alternative to From/To, allowing Bad Things, like never 
> > putting the real 
> > > From/To information in the From/To fields. 
> > 
> > I think that the proper way to handle this, as pursued in 
> > other threads, 
> > is to provide a sane alternative to 3gpp so that they don't 
> do this. I 
> > believe we all agreed it would be bad. 
> > 
> > > 
> > > Others have pointed out that the draft explicitly states 
> > that untrusted 
> > > UAs MUST NOT add RPID. However the draft explicitly 
> described proxy 
> > > handling of RPID headers from untrusted sources (proxies OR 
> > UAs), and 
> > > allows these to be propogated, marked as untrusted. 
> > > 
> > > Therefore, a UA which DID add RPID, although non-compliant 
> > to the draft, 
> > > would have its RPID propogated through the network. 
> > > 
> > > 3) 'Untrustworthy RPID' 
> > > 
> > > Flemming, Bill et al argued that RPID information is still 
> > useful when 
> > > it is 'untrustworthy'. This can occur legitimately within 
> > the draft when 
> > > an RPID which is not 'private' (and so not encrypted) 
> > crosses a trust 
> > > boundary. This would represent an identity asserted by 
> Network A and 
> > > passed to Network B, when B has no explicit trust 
> > relationship with A. 
> > > 
> > > The contention was that since this information still has 
> > some value, it 
> > > should be passed to the UAS for display, and might also have some 
> > > application for call trace, although obviously not as useful as a 
> > > 'trustworthy' identity. 
> > > 
> > > Jon argued that the From field provides well enough for 
> > display, and the 
> > > value of untrustworthy information for call trace is 
> > debatable (and has 
> > > been debated at length). 
> > 
> > IMHO, 'untrusted RPID' is identical to the From, except that it is 
> > purportedly inserted by a network element, whereas From is 
> purportedly 
> > inserted by the end user. Neither provides any guarantees. 
> The essence 
> > of the debate, I think, is whether you believe there is value 
> > to the end 
> > user in a field which is the identity purportedly inserted by some 
> > network element.  Since there is apparently value to the identity 
> > purportedly inserted by the end user, I see no reason why we should 
> > discriminate. 
> > 
> > -Jonathan R. 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue 
> > Chief Scientist                         First Floor 
> > dynamicsoft                             East Hanover, NJ 07936 
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050 
> > http://www.jdrosen.net                  PH:  (973) 952-5000 
> > http://www.dynamicsoft.com 
> > 
> 

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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<P><FONT SIZE=3D2>Brian wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; But (3) is achievable now with a B2BUA, and =
there is no </FONT>
<BR><FONT SIZE=3D2>&gt; standardization </FONT>
<BR><FONT SIZE=3D2>&gt; work necessary.&nbsp; Jonathan correctly =
pointed out that it is non-trivial</FONT>
<BR><FONT SIZE=3D2>&gt; to achieve what appears to be desired, and a =
B2BUA that thoroughly</FONT>
<BR><FONT SIZE=3D2>&gt; scrubbed all fields, including relaying of =
media, will be needed.</FONT>
<BR><FONT SIZE=3D2>&gt; No additional headers can help you.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I agree completely. Others claimed that the problem =
did not exist.</FONT>
</P>

<P><FONT SIZE=3D2>Also, there can be a range of privacy services, =
requiring more, or less, field modification etc. You will at least need =
to modify From/To &amp; probably Call-Info, Reply-To etc.</FONT></P>

<P><FONT SIZE=3D2>&gt; I think we should stop worrying about this case, =
and try to focus on</FONT>
<BR><FONT SIZE=3D2>&gt; something we can't reasonably do now, which is =
to provide a network</FONT>
<BR><FONT SIZE=3D2>&gt; asserted identity when the UA itself provides =
anonymity.&nbsp; That is a</FONT>
<BR><FONT SIZE=3D2>&gt; case worth working on, and it represents the =
way I think it </FONT>
<BR><FONT SIZE=3D2>&gt; SHOULD work.</FONT>
</P>

<P><FONT SIZE=3D2>Absolutely. I hope that this thread has demonstrated, =
though, that it is not *because* of RPID that people have done Bad =
Things with From/To, more the other way round.</FONT></P>

<P><FONT SIZE=3D2>&gt; The UA is by far in the best position to =
determine the level of</FONT>
<BR><FONT SIZE=3D2>&gt; anonymity it wishes, and if it can't get enough =
(because, for example,</FONT>
<BR><FONT SIZE=3D2>&gt; it wants it's IP addresses obscured), it will =
have to employ a</FONT>
<BR><FONT SIZE=3D2>&gt; B2BUA as above.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>Agreed. But the UA is not in a good position to =
determine the subscribers anonymity requirements, or other parties not =
represented by a UA. In a Service Provider model, these people have =
some rights. So, the network may have to employ a B2BUA or whatever on =
behalf of these people. There is no problem with this and so no case =
for obfuscating the From/To fields - agreed ?</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Brian</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Mark Watson [<A =
HREF=3D"mailto:mwatson@nortelnetworks.com">mailto:mwatson@nortelnetworks=
.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, April 11, 2002 11:29 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Jonathan Rosenberg'</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org; Ben Campbell; Flemming =
Andreasen; Peterson, </FONT>
<BR><FONT SIZE=3D2>&gt; Jon; 'William</FONT>
<BR><FONT SIZE=3D2>&gt; Marshall'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Just on this one issue of obfuscation of the =
From/To field.... </FONT>
<BR><FONT SIZE=3D2>&gt; Let's be really clear about why this has come =
about: </FONT>
<BR><FONT SIZE=3D2>&gt; Some Service Providers *believe* in the =
following proposition: </FONT>
<BR><FONT SIZE=3D2>&gt; (*) IF the UA is allowed to put identity =
information into </FONT>
<BR><FONT SIZE=3D2>&gt; From/To, THEN the</FONT>
<BR><FONT SIZE=3D2>&gt; Service Provider will be forced (by Data =
Protection) to </FONT>
<BR><FONT SIZE=3D2>&gt; provide services</FONT>
<BR><FONT SIZE=3D2>&gt; which modify these fields, in order to protect =
the privacy of </FONT>
<BR><FONT SIZE=3D2>&gt; people other</FONT>
<BR><FONT SIZE=3D2>&gt; than the actual calling user - subscriber, =
forwarding etc.</FONT>
<BR><FONT SIZE=3D2>&gt; First, set aside the question of whether to =
agree with (*) or not. </FONT>
<BR><FONT SIZE=3D2>&gt; There are three options for people with this =
belief: </FONT>
<BR><FONT SIZE=3D2>&gt; 1) *Require* permanantly that the UA does not =
put identity </FONT>
<BR><FONT SIZE=3D2>&gt; information into</FONT>
<BR><FONT SIZE=3D2>&gt; From/To </FONT>
<BR><FONT SIZE=3D2>&gt; 2) Reject requests where the UA puts identity =
information in </FONT>
<BR><FONT SIZE=3D2>&gt; From/To if</FONT>
<BR><FONT SIZE=3D2>&gt; privacy is required </FONT>
<BR><FONT SIZE=3D2>&gt; 3) Build network services which provide the =
necessary modification</FONT>
<BR><FONT SIZE=3D2>&gt; (+whatever else is needed) </FONT>
<BR><FONT SIZE=3D2>&gt; I do not think there are any other options. =
</FONT>
<BR><FONT SIZE=3D2>&gt; We all agree that (1) is bad. (2) looks =
attractive but is not </FONT>
<BR><FONT SIZE=3D2>&gt; backwards</FONT>
<BR><FONT SIZE=3D2>&gt; compatible, requires a double request (not good =
for wireless) </FONT>
<BR><FONT SIZE=3D2>&gt; &amp; has issues</FONT>
<BR><FONT SIZE=3D2>&gt; as to whether you trust the UA (in redirection =
scenarios).</FONT>
<BR><FONT SIZE=3D2>&gt; I was looking for a consensus that (3) is a =
possible and </FONT>
<BR><FONT SIZE=3D2>&gt; valid thing to do</FONT>
<BR><FONT SIZE=3D2>&gt; for these Service Providers. If the IETF SIP =
group agree that </FONT>
<BR><FONT SIZE=3D2>&gt; (3) and not</FONT>
<BR><FONT SIZE=3D2>&gt; (1) is _a_ valid solution to this problem, then =
there is more </FONT>
<BR><FONT SIZE=3D2>&gt; chance of</FONT>
<BR><FONT SIZE=3D2>&gt; other fora adopting (3) instead of (1).</FONT>
<BR><FONT SIZE=3D2>&gt; So far, people have been arguing against the =
belief (*). The </FONT>
<BR><FONT SIZE=3D2>&gt; question is not</FONT>
<BR><FONT SIZE=3D2>&gt; whether (*) is demonstrably true but whether =
there is </FONT>
<BR><FONT SIZE=3D2>&gt; sufficient *chance* of</FONT>
<BR><FONT SIZE=3D2>&gt; it being true to justify serious consideration. =
A significant group of</FONT>
<BR><FONT SIZE=3D2>&gt; Service Providers obviously think it is - they =
are worried </FONT>
<BR><FONT SIZE=3D2>&gt; they will be in</FONT>
<BR><FONT SIZE=3D2>&gt; this situation and want a solution defined - =
that should be </FONT>
<BR><FONT SIZE=3D2>&gt; enough for us to</FONT>
<BR><FONT SIZE=3D2>&gt; discuss solutions.</FONT>
<BR><FONT SIZE=3D2>&gt; So, can we agree to recommend at least that =
people with </FONT>
<BR><FONT SIZE=3D2>&gt; belief (*) adopt</FONT>
<BR><FONT SIZE=3D2>&gt; solution (3) and not solution (1) ? </FONT>
<BR><FONT SIZE=3D2>&gt; ...Mark </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message----- </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>] </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: 11 April 2002 08:28 </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To: Watson, Mark [MDN05:EP10:EXCH] </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cc: sip@ietf.org; Ben Campbell; Flemming =
Andreasen; Peterson, Jon; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 'William Marshall' </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: Re: Summary of RE: [Sip] Comment, =
SIP Privacy draft </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Mark Watson wrote: </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; All, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I feel profoundly lucky that =
yesterday and Friday were </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; public holidays </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; in the UK :-) </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Perhaps I can take advantage of this =
vantage point to offer </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; a summary of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; the thread. Protagonists please speak =
up if I have </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; mis-represented you </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; below, but please don't re-open old =
fronts. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Thanks for the summary. It is well needed. =
I suspect much of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the lack of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; comment derives from the difficulty in =
following this long thread. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I would kindly request all participants to =
please keep the </FONT>
<BR><FONT SIZE=3D2>&gt; discussions </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; technical. A lot of the emails in this =
thread were personal </FONT>
<BR><FONT SIZE=3D2>&gt; attacks. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Many more were threats along the general =
line of &quot;do X or </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; standards body </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Y will do Z&quot;. I really hate the =
latter ones. Let us hear no more of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; either type of nonsense. It is not the =
IETF way. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; 1) Call-Info </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Long arguments about the alledged =
</FONT>
<BR><FONT SIZE=3D2>&gt; similarities/differences between </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Call-Info and RPID. Suffice to say =
that the point has </FONT>
<BR><FONT SIZE=3D2>&gt; been made that </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Call-Info (and perhaps other things) =
could be abused to do </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the things </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; that RPID does, and in particular to =
do the 'Bad Things' </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that some are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; worried about with RPID. The point of =
disagreement was </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; whether in the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; case of Call-Info, this would be =
'major abuse', with RPID </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; was set up to </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; make these 'Bad Things' easy, or =
whether there was some </FONT>
<BR><FONT SIZE=3D2>&gt; equivalence. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The key point is that this kind of =
abuse of Call-Info may </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; happen if we </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; do not provide an alternative. =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; It was never our intent for Call-Info to =
provide any form </FONT>
<BR><FONT SIZE=3D2>&gt; of network </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; asserted identity that could be used in =
any useful way. </FONT>
<BR><FONT SIZE=3D2>&gt; Generally, the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; idea was that insertion of Call-Info was =
either done by the </FONT>
<BR><FONT SIZE=3D2>&gt; UA, or in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the case of proxy-inserted values, was for =
&quot;opt-in&quot; </FONT>
<BR><FONT SIZE=3D2>&gt; services were the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; user had signed up for content to be =
added. That said, there are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; certainly cases where the caller may opt =
in for this service </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; but require </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; privacy for a particular call, and we need =
a mechanism to </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; support that. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; However, in no way would I support or =
advocate the use of </FONT>
<BR><FONT SIZE=3D2>&gt; Call-Info as </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; an alternative for R-P-ID. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; 2) Abuse of From/To headers </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Jon is concerned that 'untrusted =
RPID' provides a standardised </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; alternative to From/To, allowing Bad =
Things, like never </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; putting the real </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From/To information in the From/To =
fields. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I think that the proper way to handle =
this, as pursued in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; other threads, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; is to provide a sane alternative to 3gpp =
so that they don't </FONT>
<BR><FONT SIZE=3D2>&gt; do this. I </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; believe we all agreed it would be bad. =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Others have pointed out that the =
draft explicitly states </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that untrusted </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; UAs MUST NOT add RPID. However the =
draft explicitly </FONT>
<BR><FONT SIZE=3D2>&gt; described proxy </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; handling of RPID headers from =
untrusted sources (proxies OR </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; UAs), and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; allows these to be propogated, marked =
as untrusted. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Therefore, a UA which DID add RPID, =
although non-compliant </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to the draft, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; would have its RPID propogated =
through the network. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; 3) 'Untrustworthy RPID' </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Flemming, Bill et al argued that RPID =
information is still </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; useful when </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; it is 'untrustworthy'. This can occur =
legitimately within </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the draft when </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; an RPID which is not 'private' (and =
so not encrypted) </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; crosses a trust </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; boundary. This would represent an =
identity asserted by </FONT>
<BR><FONT SIZE=3D2>&gt; Network A and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; passed to Network B, when B has no =
explicit trust </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; relationship with A. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The contention was that since this =
information still has </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; some value, it </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; should be passed to the UAS for =
display, and might also have some </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; application for call trace, although =
obviously not as useful as a </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; 'trustworthy' identity. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Jon argued that the From field =
provides well enough for </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; display, and the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; value of untrustworthy information =
for call trace is </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; debatable (and has </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; been debated at length). </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; IMHO, 'untrusted RPID' is identical to the =
From, except that it is </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; purportedly inserted by a network element, =
whereas From is </FONT>
<BR><FONT SIZE=3D2>&gt; purportedly </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; inserted by the end user. Neither provides =
any guarantees. </FONT>
<BR><FONT SIZE=3D2>&gt; The essence </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; of the debate, I think, is whether you =
believe there is value </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to the end </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; user in a field which is the identity =
purportedly inserted by some </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; network element.&nbsp; Since there is =
apparently value to the identity </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; purportedly inserted by the end user, I =
see no reason why we should </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; discriminate. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -Jonathan R. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
72 Eagle Rock Avenue </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; First Floor </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; East Hanover, NJ 07936 </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX: (973) 952-5050 =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; PH:&nbsp; (973) 952-5000 </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A> </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@optimus.ietf.org  Fri Apr 12 10:02:21 2002
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Date: Fri, 12 Apr 2002 09:00:38 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
CC: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org,
        Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
References: <313680C9A886D511A06000204840E1CF57CE3C@whq-msgusr-02.pit.comms.marconi.com>
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> it wants it's IP addresses obscured), it will have to employ a
> B2BUA as above.
> 

I agree that, in general, the UA should be treated like a mature adult
and get to make decisions as to what information it wants to include in
its call setup requests or not. I don't agree that anonymity requires a
B2BUA in all cases. For example, one could imagine that a large ISP
could hand out random IP addresses from a large pool, making it
essentially similar to an anonymizer service. There is no principal
difference between knowing that "user came from anonymizer service
anon.com" and "user came from ISP large-isp.com". (Well, almost: it may
be easier to switch anonymity providers than ISPs.) With VPN services
and L2 tunneling, you can probably create IP-level anonymous ISPs that
don't require B2B UAs and cover a very large geographical area.

A privacy B2BUA would break services, since it would have to strip all
headers, thus disabling any service that it doesn't happen to know
about.

B2BUAs are also, it seems, much more loop-prone than proxies. For
examples, see mailing list hosts, which are essentially SMTP B2BUAs.
(They also illustrate the service transparency issues, with endless
debates as to what fields should have which values and whether header
fields should or should not be copied.)

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'William Marshall'"
	 <wtm@research.att.com>, jdrosen@dynamicsoft.com
Cc: sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Brian wrote:

> To me, the service that breaks with anonymous From/To is
> the ability to call me.  If my phone rings, and the "From"
> which is displayed says "anonymous", I'm not going to answer it.

Very sensible. It could be anyone!

You might answer it if the call came from a Service Provider that you
trusted, and that Service Provider had mechanisms to verify and record the
identity of the caller, and to pass that identifier to the authorities if
the call turns out to be malicious.

There is always a balance between the desire of the caller to remain
anonymous and of the called party to reject anonymous calls. Without an
intermediary then the balance gets tipped completely to one side or the
other.

> I'll probably instruct my voicemail service to do the same.
> The Police Tip line might very well accept such calls however.
> 
> I'm reminded of the practices of some Hollywood types that
> block Caller-Id, but won't accept calls that don't provide
> Caller-Id!
> 

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<P><FONT SIZE=3D2>Brian wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; To me, the service that breaks with anonymous =
From/To is</FONT>
<BR><FONT SIZE=3D2>&gt; the ability to call me.&nbsp; If my phone =
rings, and the &quot;From&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; which is displayed says &quot;anonymous&quot;, =
I'm not going to answer it.</FONT>
</P>

<P><FONT SIZE=3D2>Very sensible. It could be anyone!</FONT>
</P>

<P><FONT SIZE=3D2>You might answer it if the call came from a Service =
Provider that you trusted, and that Service Provider had mechanisms to =
verify and record the identity of the caller, and to pass that =
identifier to the authorities if the call turns out to be =
malicious.</FONT></P>

<P><FONT SIZE=3D2>There is always a balance between the desire of the =
caller to remain anonymous and of the called party to reject anonymous =
calls. Without an intermediary then the balance gets tipped completely =
to one side or the other.</FONT></P>

<P><FONT SIZE=3D2>&gt; I'll probably instruct my voicemail service to =
do the same.</FONT>
<BR><FONT SIZE=3D2>&gt; The Police Tip line might very well accept such =
calls however.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I'm reminded of the practices of some Hollywood =
types that</FONT>
<BR><FONT SIZE=3D2>&gt; block Caller-Id, but won't accept calls that =
don't provide</FONT>
<BR><FONT SIZE=3D2>&gt; Caller-Id!</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "Rosen, Brian"
	 <Brian.Rosen@marconi.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org,
        Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen
	 <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 14:16:40 +0100
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There are many privacy services between the extremes of 'entirely up to the
UAs' and 'full B2BUA with IP address anonymiser etc.'.

All of them require 'More Than Just A Proxy' because you need to modify the
From/To fields (unless you don't care about RFC2543 compatibility, in which
case you can modify these fields at something very like a proxy).

Other fields may or may not need to be modified/removed - certainly
Call-Info & Reply-To would have to go. I think you *can* run the end-to-end
data argument with things like Subject and the message body.

There is the possibility that new headers exist which carry identity
information or other Personal Data. One solution would be to remove all
unrecognised headers at the Privacy server. This will break all the new
things.

There is another way. This would be to define now a generic capability for
carrying Personal Data, along with an indication of its privacy status. All
new services introduced later which required to send Personal Data would use
this capability. So, privacy servers could be built now which would just
remove private data from this new place, leaving unrecognised headers alone.

The privacy server would still break new services based on Personal Data -
but this is the intention of the caller. The privacy server would not break
other new things.

...Mark

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: 12 April 2002 14:01
> To: Rosen, Brian
> Cc: Watson, Mark [MDN05:EP10:EXCH]; 'Jonathan Rosenberg'; 
> sip@ietf.org;
> Ben Campbell; Flemming Andreasen; Peterson, Jon; 'William Marshall'
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> > it wants it's IP addresses obscured), it will have to employ a
> > B2BUA as above.
> > 
> 
> I agree that, in general, the UA should be treated like a mature adult
> and get to make decisions as to what information it wants to 
> include in
> its call setup requests or not. I don't agree that anonymity 
> requires a
> B2BUA in all cases. For example, one could imagine that a large ISP
> could hand out random IP addresses from a large pool, making it
> essentially similar to an anonymizer service. There is no principal
> difference between knowing that "user came from anonymizer service
> anon.com" and "user came from ISP large-isp.com". (Well, 
> almost: it may
> be easier to switch anonymity providers than ISPs.) With VPN services
> and L2 tunneling, you can probably create IP-level anonymous ISPs that
> don't require B2B UAs and cover a very large geographical area.
> 
> A privacy B2BUA would break services, since it would have to strip all
> headers, thus disabling any service that it doesn't happen to know
> about.
> 
> B2BUAs are also, it seems, much more loop-prone than proxies. For
> examples, see mailing list hosts, which are essentially SMTP B2BUAs.
> (They also illustrate the service transparency issues, with endless
> debates as to what fields should have which values and whether header
> fields should or should not be copied.)
> 

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<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<BODY>

<P><FONT SIZE=3D2>There are many privacy services between the extremes =
of 'entirely up to the UAs' and 'full B2BUA with IP address anonymiser =
etc.'.</FONT></P>

<P><FONT SIZE=3D2>All of them require 'More Than Just A Proxy' because =
you need to modify the From/To fields (unless you don't care about =
RFC2543 compatibility, in which case you can modify these fields at =
something very like a proxy).</FONT></P>

<P><FONT SIZE=3D2>Other fields may or may not need to be =
modified/removed - certainly Call-Info &amp; Reply-To would have to go. =
I think you *can* run the end-to-end data argument with things like =
Subject and the message body.</FONT></P>

<P><FONT SIZE=3D2>There is the possibility that new headers exist which =
carry identity information or other Personal Data. One solution would =
be to remove all unrecognised headers at the Privacy server. This will =
break all the new things.</FONT></P>

<P><FONT SIZE=3D2>There is another way. This would be to define now a =
generic capability for carrying Personal Data, along with an indication =
of its privacy status. All new services introduced later which required =
to send Personal Data would use this capability. So, privacy servers =
could be built now which would just remove private data from this new =
place, leaving unrecognised headers alone.</FONT></P>

<P><FONT SIZE=3D2>The privacy server would still break new services =
based on Personal Data - but this is the intention of the caller. The =
privacy server would not break other new things.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Henning Schulzrinne [<A =
HREF=3D"mailto:hgs@cs.columbia.edu">mailto:hgs@cs.columbia.edu</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 12 April 2002 14:01</FONT>
<BR><FONT SIZE=3D2>&gt; To: Rosen, Brian</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Watson, Mark [MDN05:EP10:EXCH]; 'Jonathan =
Rosenberg'; </FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org;</FONT>
<BR><FONT SIZE=3D2>&gt; Ben Campbell; Flemming Andreasen; Peterson, =
Jon; 'William Marshall'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; it wants it's IP addresses obscured), it =
will have to employ a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; B2BUA as above.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I agree that, in general, the UA should be =
treated like a mature adult</FONT>
<BR><FONT SIZE=3D2>&gt; and get to make decisions as to what =
information it wants to </FONT>
<BR><FONT SIZE=3D2>&gt; include in</FONT>
<BR><FONT SIZE=3D2>&gt; its call setup requests or not. I don't agree =
that anonymity </FONT>
<BR><FONT SIZE=3D2>&gt; requires a</FONT>
<BR><FONT SIZE=3D2>&gt; B2BUA in all cases. For example, one could =
imagine that a large ISP</FONT>
<BR><FONT SIZE=3D2>&gt; could hand out random IP addresses from a large =
pool, making it</FONT>
<BR><FONT SIZE=3D2>&gt; essentially similar to an anonymizer service. =
There is no principal</FONT>
<BR><FONT SIZE=3D2>&gt; difference between knowing that &quot;user came =
from anonymizer service</FONT>
<BR><FONT SIZE=3D2>&gt; anon.com&quot; and &quot;user came from ISP =
large-isp.com&quot;. (Well, </FONT>
<BR><FONT SIZE=3D2>&gt; almost: it may</FONT>
<BR><FONT SIZE=3D2>&gt; be easier to switch anonymity providers than =
ISPs.) With VPN services</FONT>
<BR><FONT SIZE=3D2>&gt; and L2 tunneling, you can probably create =
IP-level anonymous ISPs that</FONT>
<BR><FONT SIZE=3D2>&gt; don't require B2B UAs and cover a very large =
geographical area.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A privacy B2BUA would break services, since it =
would have to strip all</FONT>
<BR><FONT SIZE=3D2>&gt; headers, thus disabling any service that it =
doesn't happen to know</FONT>
<BR><FONT SIZE=3D2>&gt; about.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; B2BUAs are also, it seems, much more loop-prone =
than proxies. For</FONT>
<BR><FONT SIZE=3D2>&gt; examples, see mailing list hosts, which are =
essentially SMTP B2BUAs.</FONT>
<BR><FONT SIZE=3D2>&gt; (They also illustrate the service transparency =
issues, with endless</FONT>
<BR><FONT SIZE=3D2>&gt; debates as to what fields should have which =
values and whether header</FONT>
<BR><FONT SIZE=3D2>&gt; fields should or should not be copied.)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

</BODY>
</HTML>
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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr 12 11:15:29 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "Mark Watson"<mwatson@nortelnetworks.com>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: sip@ietf.org, "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>,
        "'fandreas@cisco.com'"
	 <fandreas@cisco.com>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 09:50:17 -0400
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I have a feeling this thread should be moved to SIPPING: we're going round
and round with requirements.  Anyway, I've seen two sensible points today
that call for action:

Brian identified a need for a mechanism by which the network provides a
network-asserted identity.  Is RPID this mechanism, or do we look for
another, or do we explore the requirements further?

Mark has just come up with a very interesting idea for quarantining
privacy-related material.  I can see immediately that this requires a good
deal of careful work at the protocol level.  Do we need some more
exploration of the requirements here?

-----Original Message-----
From: Watson, Mark [MDN05:EP10:EXCH] 
Sent: Friday, April 12, 2002 9:17 AM
To: 'Henning Schulzrinne'; Rosen, Brian
Cc: 'Jonathan Rosenberg'; sip@ietf.org; Ben Campbell; Flemming Andreasen;
Peterson, Jon; 'William Marshall'
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


There are many privacy services between the extremes of 'entirely up to the
UAs' and 'full B2BUA with IP address anonymiser etc.'.
All of them require 'More Than Just A Proxy' because you need to modify the
From/To fields (unless you don't care about RFC2543 compatibility, in which
case you can modify these fields at something very like a proxy).
Other fields may or may not need to be modified/removed - certainly
Call-Info & Reply-To would have to go. I think you *can* run the end-to-end
data argument with things like Subject and the message body.
There is the possibility that new headers exist which carry identity
information or other Personal Data. One solution would be to remove all
unrecognised headers at the Privacy server. This will break all the new
things.
There is another way. This would be to define now a generic capability for
carrying Personal Data, along with an indication of its privacy status. All
new services introduced later which required to send Personal Data would use
this capability. So, privacy servers could be built now which would just
remove private data from this new place, leaving unrecognised headers alone.
The privacy server would still break new services based on Personal Data -
but this is the intention of the caller. The privacy server would not break
other new things.
...Mark 
> -----Original Message----- 
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu] 
> Sent: 12 April 2002 14:01 
> To: Rosen, Brian 
> Cc: Watson, Mark [MDN05:EP10:EXCH]; 'Jonathan Rosenberg'; 
> sip@ietf.org; 
> Ben Campbell; Flemming Andreasen; Peterson, Jon; 'William Marshall' 
> Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft 
> 
> 
> > it wants it's IP addresses obscured), it will have to employ a 
> > B2BUA as above. 
> > 
> 
> I agree that, in general, the UA should be treated like a mature adult 
> and get to make decisions as to what information it wants to 
> include in 
> its call setup requests or not. I don't agree that anonymity 
> requires a 
> B2BUA in all cases. For example, one could imagine that a large ISP 
> could hand out random IP addresses from a large pool, making it 
> essentially similar to an anonymizer service. There is no principal 
> difference between knowing that "user came from anonymizer service 
> anon.com" and "user came from ISP large-isp.com". (Well, 
> almost: it may 
> be easier to switch anonymity providers than ISPs.) With VPN services 
> and L2 tunneling, you can probably create IP-level anonymous ISPs that 
> don't require B2B UAs and cover a very large geographical area. 
> 
> A privacy B2BUA would break services, since it would have to strip all 
> headers, thus disabling any service that it doesn't happen to know 
> about. 
> 
> B2BUAs are also, it seems, much more loop-prone than proxies. For 
> examples, see mailing list hosts, which are essentially SMTP B2BUAs. 
> (They also illustrate the service transparency issues, with endless 
> debates as to what fields should have which values and whether header 
> fields should or should not be copied.) 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Fri, 12 Apr 2002 10:31:58 EDT
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
To: dean.willis@softarmor.com, mwatson@nortelnetworks.com,
        jdrosen@dynamicsoft.com
CC: sip@ietf.org, bcampbell@dynamicsoft.com, fandreas@cisco.com,
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In a message dated 4/11/02 2:30:03 PM Eastern Daylight Time, 
dean.willis@softarmor.com writes:


> If we earnestly believe that Data Protection is going to require that
> we, by policy, alter these human-supplied/human-consumed values, then
> we're logically going to have to do the same for every other
> human-supplied/human-consumed values. I would argue that this completely
> defeats the value of an Internet protocol in this context.
> 
> 

This discussion seems to have gotton off of the real requirement, which is to 
correctly handle calling user identities that are needed as part of the 
service offering. That is, the information needed to process a call setup but 
not necessarily needed by the called user (except out of curiousity). There 
have been a number of suggestions that the "privacy" service must 
remove/mask/encrypt any other information which the calling user entered, 
such as "Subject" that might give away their identity. This is absurd. No 
network or B2BUA of trusted proxy or whatever you want to call it could know 
when to do this. There can never be any "service" which changes or looks at 
the "Subject" header.

If I block my calling line ID today (telephone number) from being delivered 
to the other end and then, when someone answers, say "This is Mike Pierce", 
there is no requirement that the network remove that identity. Nor should 
there be any belief that the "network" should block the call if I voluntarily 
enter some identifying information.

As Mark Watson noted "A Service Provider may have some responsibility for 
apparently user-supplied information if it is implicit or required in the 
service that the user shall supply a certain piece of information in that 
field."

This discussion should only deal with the specific information that the 
identity/privacy service deals with. I guess that's the To and From headers 
and any new headers that might be created to get around limitations of these 
headers.

Mike



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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/11/02 2:30:03 PM Eastern Daylight Time, dean.willis@softarmor.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">If we earnestly believe that Data Protection is going to require that
<BR>we, by policy, alter these human-supplied/human-consumed values, then
<BR>we're logically going to have to do the same for every other
<BR>human-supplied/human-consumed values. I would argue that this completely
<BR>defeats the value of an Internet protocol in this context.
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>This discussion seems to have gotton off of the real requirement, which is to correctly handle calling user identities that are needed as part of the service offering. That is, the information needed to process a call setup but not necessarily needed by the called user (except out of curiousity). There have been a number of suggestions that the "privacy" service must remove/mask/encrypt any other information which the calling user entered, such as "Subject" that might give away their identity. This is absurd. No network or B2BUA of trusted proxy or whatever you want to call it could know when to do this. There can never be any "service" which changes or looks at the "Subject" header.
<BR>
<BR>If I block my calling line ID today (telephone number) from being delivered to the other end and then, when someone answers, say "This is Mike Pierce", there is no requirement that the network remove that identity. Nor should there be any belief that the "network" should block the call if I voluntarily enter some identifying information.
<BR>
<BR>As Mark Watson noted "A Service Provider may have some responsibility for apparently user-supplied information if it is implicit or required in the service that the user shall supply a certain piece of information in that field."
<BR>
<BR>This discussion should only deal with the specific information that the identity/privacy service deals with. I guess that's the To and From headers and any new headers that might be created to get around limitations of these headers.
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

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From daemon@ns.ietf.org  Fri Apr 12 11:43:26 2002
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
To: jdrosen@dynamicsoft.com, mwatson@nortelnetworks.com
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In a message dated 4/11/02 10:05:13 PM Eastern Daylight Time, 
jdrosen@dynamicsoft.com writes:


> Honestly, I am flattered that you are hoping for the IETF blessing for
> building a service. However, building such an anonyzmizer in (3) is well
> within the scope of the SIP protocol, and requires no standardization
> activity or consensus from IETF. I personally believe that you need a
> full b2bua, since privacy, from a user perspecitve, WILL involve
> stripping everything which might reveal identity, no matter whether the
> header formally has identity information or not. Thus, I am not sure
> what there is to do, but to emphasize to our friends in 3gpp to NOT
> always put anonymous values in the To/From as this clearly will break
> all reasonable interop with the outside world. (1) is BAD. Does anyone
> dispute that?
> 

I think the message from some on this list is that there is no way that the 
identity/privacy "service" is going to work correctly unless there is a full 
"standard" defining the required interoperation between end user UAs and 
"network" devices supplied by "Service Providers", whether you call them 
B2BUAs, proxies, or anonymizers. By "correctly", I mean in accordance with 
the law in effect at the location where the calling and called party is. 
Unless we admit to this model as the basis for requirements, no progress can 
be made. Of course, I am referring to the model # 1 I listed in my April 4 
e-mail which were:

1. With a trusted third party (Read: service provider) which provides 
services, one of those services being to guarantee privacy/protection/etc.
2. Without a trusted third party (Read: Wild-wild west) meaning that there 
are no services being provided. End users do anything they want and no one is 
guaranteed of anything (e.g., privacy, protection, or that anything they try 
to do will neccesarily function as they expect).

As Mark Watson wrote in his e-mail of April 12: "It's not about 'telco like' 
vs 'Internet like'. It's not about the technology at all. We have to look at 
the *service* as it is supplied to the user. If you have a Service Provider 
model at all, then the Service Provider has responsibilities about what they 
do with people's Personal Data, whether the service it implemented using 
IPv20 between nano-machines or over a wet piece of string."

Once again, it seems that this discussion needs to be split into addressing 
these two completely different environments with potentially completely 
different solutions. I know that sounds bad, and contradicts some views of 
what the Internet should be, but I see no other approach.

My 2 cents worth!

Mike



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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/11/02 10:05:13 PM Eastern Daylight Time, jdrosen@dynamicsoft.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Honestly, I am flattered that you are hoping for the IETF blessing for
<BR>building a service. However, building such an anonyzmizer in (3) is well
<BR>within the scope of the SIP protocol, and requires no standardization
<BR>activity or consensus from IETF. I personally believe that you need a
<BR>full b2bua, since privacy, from a user perspecitve, WILL involve
<BR>stripping everything which might reveal identity, no matter whether the
<BR>header formally has identity information or not. Thus, I am not sure
<BR>what there is to do, but to emphasize to our friends in 3gpp to NOT
<BR>always put anonymous values in the To/From as this clearly will break
<BR>all reasonable interop with the outside world. (1) is BAD. Does anyone
<BR>dispute that?
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I think the message from some on this list is that there is no way that the identity/privacy "service" is going to work correctly unless there is a full "standard" defining the required interoperation between end user UAs and "network" devices supplied by "Service Providers", whether you call them B2BUAs, proxies, or anonymizers. By "correctly", I mean in accordance with the law in effect at the location where the calling and called party is. Unless we admit to this model as the basis for requirements, no progress can be made. Of course, I am referring to the model # 1 I listed in my April 4 e-mail which were:
<BR>
<BR>1. With a trusted third party (Read: service provider) which provides services, one of those services being to guarantee privacy/protection/etc.
<BR>2. Without a trusted third party (Read: Wild-wild west) meaning that there are no services being provided. End users do anything they want and no one is guaranteed of anything (e.g., privacy, protection, or that anything they try to do will neccesarily function as they expect).
<BR>
<BR>As Mark Watson wrote in his e-mail of April 12: "It's not about 'telco like' vs 'Internet like'. It's not about the technology at all. We have to look at the *service* as it is supplied to the user. If you have a Service Provider model at all, then the Service Provider has responsibilities about what they do with people's Personal Data, whether the service it implemented using IPv20 between nano-machines or over a wet piece of string.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">"</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>Once again, it seems that this discussion needs to be split into addressing these two completely different environments with potentially completely different solutions. I know that sounds bad, and contradicts some views of what the Internet should be, but I see no other approach.
<BR>
<BR>My 2 cents worth!
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/11/02 2:30:03 PM Eastern Daylight Time, 
dean.willis@softarmor.com writes:


> 5) Use a non-Internet protocol which completely constrains the service
> functionality and user experience to a least common denominator and stop
> pretending that your requirements have anything to do with the Internet.
> 
> Look, there are two reasons why we may want to use Internet protocols:
> 
> <snip>
> 
> 2) Because we're jealous of the success the Internet has had and want to
> contaminate it with "telco like" requirements so that Internet people
> can be miserable too.
> 
> <snip>
> 
> If we earnestly believe that Data Protection is going to require that
> we, by policy, alter these human-supplied/human-consumed values, then
> we're logically going to have to do the same for every other
> human-supplied/human-consumed values. I would argue that this completely
> defeats the value of an Internet protocol in this context.
> 
> I believe it is significantly more effective to provide guidelines on
> human-supplied/human consumed values, and implement hard requirements
> for data protection only on network-supplied values.
> 
> 

The issue here is not what any of us want to do to "contaminate" the purity 
of IP or to make it look like the PSTN with its "limited" capabilities, but 
rather, what will be required by various regulatory agencies once they 
realize that, if IP is going to be used to replace traditional telephone 
service, it has to comply with all the same rules and regulations. The area 
of calling line ID presentation and restriction is the most complicated, with 
different rules in different states and countries.

Voice over IP isn't going to escape this regulation for long. It will get 
"telco-like" requirements sooner or later. It's pretty obvious to me that 
there needs to be an agreement on the requirements before any more discussion 
on the protocol details that might be needed to solve an (undefined) problem.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/11/02 2:30:03 PM Eastern Daylight Time, dean.willis@softarmor.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">5) Use a non-Internet protocol which completely constrains the service
<BR>functionality and user experience to a least common denominator and stop
<BR>pretending that your requirements have anything to do with the Internet.
<BR>
<BR>Look, there are two reasons why we may want to use Internet protocols:
<BR>
<BR>&lt;snip&gt;
<BR>
<BR>2) Because we're jealous of the success the Internet has had and want to
<BR>contaminate it with "telco like" requirements so that Internet people
<BR>can be miserable too.
<BR>
<BR>&lt;snip&gt;
<BR>
<BR>If we earnestly believe that Data Protection is going to require that
<BR>we, by policy, alter these human-supplied/human-consumed values, then
<BR>we're logically going to have to do the same for every other
<BR>human-supplied/human-consumed values. I would argue that this completely
<BR>defeats the value of an Internet protocol in this context.
<BR>
<BR>I believe it is significantly more effective to provide guidelines on
<BR>human-supplied/human consumed values, and implement hard requirements
<BR>for data protection only on network-supplied values.
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>The issue here is not what any of us want to do to "contaminate" the purity of IP or to make it look like the PSTN with its "limited" capabilities, but rather, what will be required by various regulatory agencies once they realize that, if IP is going to be used to replace traditional telephone service, it has to comply with all the same rules and regulations. The area of calling line ID presentation and restriction is the most complicated, with different rules in different states and countries.
<BR>
<BR>Voice over IP isn't going to escape this regulation for long. It will get "telco-like" requirements sooner or later. It's pretty obvious to me that there needs to be an agreement on the requirements before any more discussion on the protocol details that might be needed to solve an (undefined) problem.
<BR>
<BR>Mike
<BR></FONT></HTML>

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 11:32:15 -0400
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The problem with this line of thinking is that in the PSTN, identity is
always asserted by
the network, and never by the UA.  In SIP, presently, identity is always
asserted by
the UA and never the network.  Privacy when you have network asserted
identity
means something different than when you have user asserted identity.
 
There are no regulations, and I don't see any reason to believe that
regulators
will slavishly follow the current practice.  Until you have evidence to the
contrary,
arguing that the same SOLUTIONS are required is not helpful. 
 
What you do need to cater for is services.  If there are UAs that have
anonymity services,
a customer might prefer that UA from another.  If a service provider
provides an
anonymity service, that may find some customers.  If both exist, then
regulators may
be content, or they may require that all service providers provide B2BUAs.  
 
Without a network asserted identity, either a UA can implement anonymity, or
a B2BUA can do that.  No standards required in the IETF for that.
 
The problem comes with network asserted identity.  There is a clear
requirement
to be able to do a call trace on an anonymous call.  Proponents of a network
asserted identity posit that NAI is how you do call trace.  Opponents
counter that
you can't actually achieve anonymity with network asserted identity. That, I
believe
is the crux of the problem we now face: how to trace a call while allowing
the
user to actually be anonymous.
 
If we could get consensus that trace-with-anonymity is the problem, then we
can
work on solutions.  Until we get to something as straightforward as that,
we are going to spin a lot of wheels.
 
Brian 

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Friday, April 12, 2002 10:32 AM
To: sip@ietf.org
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft


In a message dated 4/11/02 2:30:03 PM Eastern Daylight Time,
dean.willis@softarmor.com writes: 




5) Use a non-Internet protocol which completely constrains the service 
functionality and user experience to a least common denominator and stop 
pretending that your requirements have anything to do with the Internet. 

Look, there are two reasons why we may want to use Internet protocols: 

<snip> 

2) Because we're jealous of the success the Internet has had and want to 
contaminate it with "telco like" requirements so that Internet people 
can be miserable too. 

<snip> 

If we earnestly believe that Data Protection is going to require that 
we, by policy, alter these human-supplied/human-consumed values, then 
we're logically going to have to do the same for every other 
human-supplied/human-consumed values. I would argue that this completely 
defeats the value of an Internet protocol in this context. 

I believe it is significantly more effective to provide guidelines on 
human-supplied/human consumed values, and implement hard requirements 
for data protection only on network-supplied values. 





The issue here is not what any of us want to do to "contaminate" the purity
of IP or to make it look like the PSTN with its "limited" capabilities, but
rather, what will be required by various regulatory agencies once they
realize that, if IP is going to be used to replace traditional telephone
service, it has to comply with all the same rules and regulations. The area
of calling line ID presentation and restriction is the most complicated,
with different rules in different states and countries. 

Voice over IP isn't going to escape this regulation for long. It will get
"telco-like" requirements sooner or later. It's pretty obvious to me that
there needs to be an agreement on the requirements before any more
discussion on the protocol details that might be needed to solve an
(undefined) problem. 

Mike 



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<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=ISO-8859-1">


<META content="MSHTML 6.00.2712.300" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>The problem 
with this line of thinking is that in the PSTN, identity is always asserted 
by</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>the network, 
and never by the UA.&nbsp; In SIP, presently, identity is always asserted 
by</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>the UA and 
never the network.&nbsp; Privacy when you have network asserted 
identity</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>means 
something different than when you have user asserted 
identity.</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>There are no 
regulations, and&nbsp;I don't see any reason to believe that 
regulators</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>will 
slavishly follow the current practice.&nbsp; Until you have evidence to the 
contrary,</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>arguing that 
the same SOLUTIONS are required is not helpful.&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>What you do 
need to cater for is services.&nbsp; If there are UAs that have anonymity 
services,</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>a customer 
might prefer that UA from another.&nbsp; If a service provider provides 
an</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>anonymity 
service, that may find some customers.&nbsp; If both exist, then regulators 
may</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>be content, 
or they may require that all service providers provide B2BUAs.&nbsp; 
</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial 
color=#0000ff></FONT></SPAN><SPAN class=480061415-12042002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>Without a 
network asserted identity, either a UA can implement anonymity, 
or</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>a B2BUA can 
do that.&nbsp; No standards required in the IETF for that.</FONT></SPAN></DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=480061415-12042002><FONT face=Arial color=#0000ff>The problem 
comes with network asserted identity.&nbsp;&nbsp;There is a clear 
requirement</FONT></SPAN></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>to be 
able to do a call trace on an anonymous call.&nbsp; Proponents of a 
network</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN 
class=480061415-12042002>asserted identity posit that&nbsp;NAI is how you do 
call trace.&nbsp; Opponents counter that</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>you 
can't actually achieve anonymity with network asserted identity. That, I 
believe</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>is the 
crux of the problem we&nbsp;now face: how to trace a call while allowing 
the</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>user to 
actually be anonymous.</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN 
class=480061415-12042002></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>If we 
could get consensus that trace-with-anonymity is the problem, then we 
can</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>work on 
solutions.&nbsp; Until we get to something as straightforward as 
that,</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN class=480061415-12042002>we are 
going to spin a lot of wheels.</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN 
class=480061415-12042002></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN 
class=480061415-12042002>Brian&nbsp;</SPAN></FONT></FONT></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Friday, April 12, 2002 10:32 
  AM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> Re: Summary of RE: [Sip] 
  Comment, SIP Privacy draft<BR><BR></FONT></DIV><FONT 
  face=arial,helvetica><FONT size=2>In a message dated 4/11/02 2:30:03 PM 
  Eastern Daylight Time, dean.willis@softarmor.com writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">5) Use a non-Internet protocol which completely constrains the 
    service <BR>functionality and user experience to a least common denominator 
    and stop <BR>pretending that your requirements have anything to do with the 
    Internet. <BR><BR>Look, there are two reasons why we may want to use 
    Internet protocols: <BR><BR>&lt;snip&gt; <BR><BR>2) Because we're jealous of 
    the success the Internet has had and want to <BR>contaminate it with "telco 
    like" requirements so that Internet people <BR>can be miserable too. 
    <BR><BR>&lt;snip&gt; <BR><BR>If we earnestly believe that Data Protection is 
    going to require that <BR>we, by policy, alter these 
    human-supplied/human-consumed values, then <BR>we're logically going to have 
    to do the same for every other <BR>human-supplied/human-consumed values. I 
    would argue that this completely <BR>defeats the value of an Internet 
    protocol in this context. <BR><BR>I believe it is significantly more 
    effective to provide guidelines on <BR>human-supplied/human consumed values, 
    and implement hard requirements <BR>for data protection only on 
    network-supplied values. <BR><BR></FONT><FONT lang=0 face=Arial 
    color=#000000 size=3 FAMILY="SANSSERIF"></BLOCKQUOTE><BR></FONT><FONT lang=0 
  face=Arial color=#000000 size=2 FAMILY="SANSSERIF"><BR>The issue here is not 
  what any of us want to do to "contaminate" the purity of IP or to make it look 
  like the PSTN with its "limited" capabilities, but rather, what will be 
  required by various regulatory agencies once they realize that, if IP is going 
  to be used to replace traditional telephone service, it has to comply with all 
  the same rules and regulations. The area of calling line ID presentation and 
  restriction is the most complicated, with different rules in different states 
  and countries. <BR><BR>Voice over IP isn't going to escape this regulation for 
  long. It will get "telco-like" requirements sooner or later. It's pretty 
  obvious to me that there needs to be an agreement on the requirements before 
  any more discussion on the protocol details that might be needed to solve an 
  (undefined) problem. <BR><BR>Mike <BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
To: mwatson@nortelnetworks.com, hgs@cs.columbia.edu, Brian.Rosen@marconi.com
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In a message dated 4/12/02 9:30:26 AM Eastern Daylight Time, 
mwatson@nortelnetworks.com writes:


> There are many privacy services between the extremes of 'entirely up to the 
> UAs' and 'full B2BUA with IP address anonymiser etc.'.
> All of them require 'More Than Just A Proxy' because you need to modify the 
> From/To fields (unless you don't care about RFC2543 compatibility, in which 
> case you can modify these fields at something very like a proxy).
> Other fields may or may not need to be modified/removed - certainly 
> Call-Info & Reply-To would have to go. I think you *can* run the end-to-end 
> data argument with things like Subject and the message body.
> There is the possibility that new headers exist which carry identity 
> information or other Personal Data. One solution would be to remove all 
> unrecognised headers at the Privacy server. This will break all the new 
> things.
> There is another way. This would be to define now a generic capability for 
> carrying Personal Data, along with an indication of its privacy status. All 
> new services introduced later which required to send Personal Data would 
> use this capability. So, privacy servers could be built now which would 
> just remove private data from this new place, leaving unrecognised headers 
> alone.
> The privacy server would still break new services based on Personal Data - 
> but this is the intention of the caller. The privacy server would not break 
> other new things.
> ...Mark 
> 

I continue to be confused by this emphasis on the "privacy" half of the 
draft. It seems to be expanding to consider "Privacy" of everything including 
what color socks the user is wearing! I know that the overwhelming thoughts 
in the IETF are to protect privacy, and I don't dispute that, but the primary 
purpose of this draft (and the "service" it defines) is to provide an 
"Identity" service, that is, providing one user's identity to the other user. 
"Privacy" is the second half. We need to agree on what that identity is and 
how it is provided. Then the second part of the issue is how that identity is 
protected when the user doesn't want it delivered to the other user (assuming 
the "network" needs it or knows it). I know this sounds elementary and maybe 
some think it is obvious, but when the discussion of "privacy" begins to 
define ways to treat future headers carrying other personal data, I think it 
is going beyond its purpose.

As Mark noted, a Service Provider may have responsibiity for user-supplied 
information if it is implicit or required in the service that the user supply 
it (to set up the call). Any extra stuff that is defined just to be passed 
end-to-end (like Subject) or for a completely different service is not a 
concern. If the user voluntarily includes it, it is not private.

For example, Dean started a discussion of a 'What-I-Want-You-To-Call-Me:' 
field instead of "From". To me, the fundamental difference is that the "From" 
field is part of the call setup specification and the "network" should 
require it, but the "What-I-Want-You-To-Call-Me" field could be see as only 
being of end-to-end significance and not looked at by the "network". The 
basic question for this and any field is: "Is it part of 'Identity' or is it 
supplemental information?"

Mike



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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/12/02 9:30:26 AM Eastern Daylight Time, mwatson@nortelnetworks.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">There are many privacy services between the extremes of 'entirely up to the UAs' and 'full B2BUA with IP address anonymiser etc.'.
<BR>All of them require 'More Than Just A Proxy' because you need to modify the From/To fields (unless you don't care about RFC2543 compatibility, in which case you can modify these fields at something very like a proxy).
<BR>Other fields may or may not need to be modified/removed - certainly Call-Info &amp; Reply-To would have to go. I think you *can* run the end-to-end data argument with things like Subject and the message body.
<BR>There is the possibility that new headers exist which carry identity information or other Personal Data. One solution would be to remove all unrecognised headers at the Privacy server. This will break all the new things.
<BR>There is another way. This would be to define now a generic capability for carrying Personal Data, along with an indication of its privacy status. All new services introduced later which required to send Personal Data would use this capability. So, privacy servers could be built now which would just remove private data from this new place, leaving unrecognised headers alone.
<BR>The privacy server would still break new services based on Personal Data - but this is the intention of the caller. The privacy server would not break other new things.
<BR>...Mark 
<BR></BLOCKQUOTE>
<BR>
<BR>I continue to be confused by this emphasis on the "privacy" half of the draft. It seems to be expanding to consider "Privacy" of everything including what color socks the user is wearing! I know that the overwhelming thoughts in the IETF are to protect privacy, and I don't dispute that, but the primary purpose of this draft (and the "service" it defines) is to provide an "Identity" service, that is, providing one user's identity to the other user. "Privacy" is the second half. We need to agree on what that identity is and how it is provided. Then the second part of the issue is how that identity is protected when the user doesn't want it delivered to the other user (assuming the "network" needs it or knows it). I know this sounds elementary and maybe some think it is obvious, but when the discussion of "privacy" begins to define ways to treat future headers carrying other personal data, I think it is going beyond its purpose.
<BR>
<BR>As Mark noted, a Service Provider may have responsibiity for user-supplied information if it is implicit or required in the service that the user supply it (to set up the call). Any extra stuff that is defined just to be passed end-to-end (like Subject) or for a completely different service is not a concern. If the user voluntarily includes it, it is not private.
<BR>
<BR>For example, Dean started a discussion of a 'What-I-Want-You-To-Call-Me:' field instead of "From". To me, the fundamental difference is that the "From" field is part of the call setup specification and the "network" should require it, but the "What-I-Want-You-To-Call-Me" field could be see as only being of end-to-end significance and not looked at by the "network". The basic question for this and any field is: "Is it part of 'Identity' or is it supplemental information?"
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

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Date: Fri, 12 Apr 2002 11:19:01 EDT
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
To: taylor@nortelnetworks.com, mwatson@nortelnetworks.com,
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In a message dated 4/12/02 9:57:14 AM Eastern Daylight Time, 
taylor@nortelnetworks.com writes:


> I have a feeling this thread should be moved to SIPPING: we're going round
> and round with requirements.  Anyway, I've seen two sensible points today
> that call for action:
> 
> Brian identified a need for a mechanism by which the network provides a
> network-asserted identity.  Is RPID this mechanism, or do we look for
> another, or do we explore the requirements further?
> 
> Mark has just come up with a very interesting idea for quarantining
> privacy-related material.  I can see immediately that this requires a good
> deal of careful work at the protocol level.  Do we need some more
> exploration of the requirements here?
> 

I agree that this needs to be moved to SIPPING and focus on the requirements. 
We need to get away from the discussion of protocol mechanisms for awhile and 
agree on what the requirements are.

The issues from the 2nd and 3rd paragraphs above, levaing out the protocol 
stuff, are:

What are the requirements for the network to verify a user provided identity?
What are the requirements for the network to add a network-asserted identity?
Are there requirements to "Quaratine" or hide any other user provided 
identity related material?

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/12/02 9:57:14 AM Eastern Daylight Time, taylor@nortelnetworks.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I have a feeling this thread should be moved to SIPPING: we're going round
<BR>and round with requirements. &nbsp;Anyway, I've seen two sensible points today
<BR>that call for action:
<BR>
<BR>Brian identified a need for a mechanism by which the network provides a
<BR>network-asserted identity. &nbsp;Is RPID this mechanism, or do we look for
<BR>another, or do we explore the requirements further?
<BR>
<BR>Mark has just come up with a very interesting idea for quarantining
<BR>privacy-related material. &nbsp;I can see immediately that this requires a good
<BR>deal of careful work at the protocol level. &nbsp;Do we need some more
<BR>exploration of the requirements here?
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I agree that this needs to be moved to SIPPING and focus on the requirements. We need to get away from the discussion of protocol mechanisms for awhile and agree on what the requirements are.
<BR>
<BR>The issues from the 2nd and 3rd paragraphs above, levaing out the protocol stuff, are:
<BR>
<BR>What are the requirements for the network to verify a user provided identity?
<BR>What are the requirements for the network to add a network-asserted identity?
<BR>Are there requirements to "Quaratine" or hide any other user provided identity related material?
<BR>
<BR>Mike
<BR></FONT></HTML>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <Mpierce1@aol.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 10:52:24 -0500
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Mike wrote:
------------- 
 Voice over IP isn't going to escape this regulation for long. It will
get "telco-like" requirements sooner or later. It's pretty obvious to me
that there needs to be an agreement on the requirements before any more
discussion on the protocol details that might be needed to solve an
(undefined) problem.  
-------------- 
 
I don't completely disagree. In fact, I'm arguing the requirements.
 
The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or
have any responsibility for them. Insisting that the service provider
change them is just like insisting that the service provider change any
other user-supplied information, such as the words I'm typing now.
Anything I put in my To or from field is deliberately and explicitly
intended to be delivered to the terminating side of the dialog, and if
as a service provider you FAIL to do that, then you have DAMAGED my
data, and my lawyers will be in touch!
 
The calling-party-identity presentation infromation (CLIP or RPID) is
NOT inserted by the user. It is inserted by the network, WITHOUT THE
CONSENT of the user. As such, the operator is responsible for the
protection of that information in a manner consistent with regulation
and the expressed wishes of the user.
 
The calling-line-identity-restriction information (CLIR or Privacy) is a
means by which the user expresses wishes with respect to the handling of
NETWORK provided information.
 
This divides data into three classes:
 
1) Data inserted by a user's system, under the control of a user, and
intended to be delivered to the termination of the dialog
2) Data inserted by a user's system, under the control of a user,
intended to be delivered to an intermediate system and not beyond
3) Data inserted by the network, without the awareness of a user,
intended to be delievred to an intermediary or terminal point.
 
Service providers are by regulation in reasonable nations required to
prevent class 2 information from being leaked, and to respect the user's
preferences for delivery of class 3 data. 
 
The problem is, the "normal" telephone network doesn't obviously HAVE
user-provided information in the signaling path which is intended to be
delivered to the other end, so telco people start by assuming that all
signaling is the responsibility of the service provider. But there
really is something analogous. Consider Q.SIG, user signaling at the
ISDN level generally used for PBX-to-PBX work. What Mark has been
arguing is essentially like arguing that a service provider must PREVENT
the delivery of Q.SIG data because it MIGHT compromise Data Privacy. Of
course it does! It's SUPPOSED to! The operator's responsibility is to
limit the delivery of the data to the intended party (as opposed to
deliverying it to somebody else), not to keep the user from
communicating with the intended party.
 
That's right, kids -- SIP by definition is an enhanced service channel.
It does MORE than plain-old-telephone-service. If you use it carefully,
you can emulate POTS.
 
So, what are these "anoymizer" services we keep talking about? Are they
consistent with my data class division? Notice that class #1 says "to
the termination of the dialog". In using such a service, I'm requesting
that my dialog terminate at an intermediate point which implements the
service of generating a second related but anonymized dialog. In other
words, the behavior is explicitly requested by the user, and is
therefore "ok with me".
 
--
Dean
 

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<DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
face=3DArial><FONT=20
size=3D2><SPAN class=3D819102015-12042002><FONT color=3D#0000ff>Mike=20
wrote:</FONT></SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
face=3DArial><FONT=20
size=3D2><SPAN class=3D819102015-12042002><FONT=20
color=3D#0000ff>-------------</FONT>&nbsp;</SPAN></FONT></FONT></FONT></D=
IV>
<DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT><FONT =
face=3DArial><FONT=20
size=3D2><SPAN class=3D819102015-12042002>&nbsp;</SPAN>Voice over IP =
isn't going to=20
escape this regulation for long. It will get "telco-like" requirements =
sooner or=20
later. It's pretty obvious to me that there needs to be an agreement on =
the=20
requirements before any more discussion on the protocol details that =
might be=20
needed to solve an (undefined) problem.&nbsp;<SPAN=20
class=3D819102015-12042002><FONT=20
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT size=3D2><SPAN class=3D819102015-12042002><FONT=20
color=3D#0000ff>--------------</FONT>&nbsp;</SPAN><BR><SPAN=20
class=3D819102015-12042002><FONT=20
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></FONT></FONT></FONT></=
DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
color=3D#0000ff>I=20
don't completely disagree. In fact, I'm arguing the=20
requirements.</FONT></SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT size=3D2><SPAN class=3D819102015-12042002><FONT=20
color=3D#0000ff></FONT></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</=
DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
color=3D#0000ff>The=20
point I'm trying to make is that To and From are USER supplied =
information.=20
The&nbsp;Service Provider doesn't insert them, inspect them, or have any =

responsibility for them. Insisting that the service provider change them =
is just=20
like&nbsp;insisting that the service provider change any other =
user-supplied=20
information,&nbsp;such as the&nbsp;words I'm typing now. Anything I put =
in my To=20
or from field is deliberately and explicitly intended to be delivered to =
the=20
terminating side of the dialog, and if as a service provider you FAIL to =
do=20
that, then you have DAMAGED my data, and my lawyers will be in=20
touch!</FONT></SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>The=20
calling-party-identity presentation infromation (CLIP or RPID) is NOT =
inserted=20
by the user. It is inserted by the network, WITHOUT THE CONSENT of the =
user. As=20
such, the operator is responsible for the protection of that information =
in a=20
manner consistent with regulation and the expressed wishes of the=20
user.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>The=20
calling-line-identity-restriction information (CLIR or Privacy) is a =
means by=20
which the user expresses wishes with respect to the handling of NETWORK =
provided=20
information.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>This=20
divides data into three =
classes:</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002>1)&nbsp;Data inserted by&nbsp;a user's =
system, under=20
the control of&nbsp;a user, and intended to be delivered to the =
termination of=20
the dialog</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002>2)&nbsp;Data inserted by&nbsp;a user's =
system, under=20
the control of&nbsp;a user, intended to be delivered to an intermediate =
system=20
and not beyond</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>3) Data=20
inserted by the network, without the awareness of&nbsp;a user, intended =
to be=20
delievred to an intermediary or terminal=20
point.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>Service=20
providers are by regulation in reasonable nations required to prevent =
class 2=20
information from being leaked, and to respect the user's preferences for =

delivery of class 3 data. =
</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>The problem=20
is, the "normal" telephone network doesn't&nbsp;obviously HAVE =
user-provided=20
information in the signaling path which is intended to be delivered to =
the other=20
end, so telco people start by assuming that all signaling is the =
responsibility=20
of the service provider. But there really is something analogous. =
Consider=20
Q.SIG, user signaling at the ISDN level generally used for PBX-to-PBX =
work. What=20
Mark has been arguing is essentially like arguing that a service =
provider must=20
PREVENT the delivery of Q.SIG data because it MIGHT compromise Data =
Privacy. Of=20
course it does! It's SUPPOSED to! The operator's responsibility is to =
limit the=20
delivery of the data to the intended party (as opposed to deliverying it =
to=20
somebody else), not to keep the user from communicating with the =
intended=20
party.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>That's=20
right, kids -- SIP by definition is an enhanced service channel. It does =
MORE=20
than plain-old-telephone-service. If you use it carefully, you can =
emulate=20
POTS.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN =
class=3D819102015-12042002>So, what=20
are these "anoymizer" services we keep talking about? Are they =
consistent with=20
my data class division? Notice that class #1 says "to the termination of =
the=20
dialog".&nbsp;In using such a service, I'm requesting that my dialog =
terminate=20
at an intermediate point which implements the service of generating a =
second=20
related but anonymized dialog. In other words, the behavior is =
explicitly=20
requested by the user, and is therefore "ok with=20
me".</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbs=
p;</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002>--</SPAN></FONT></FONT></FONT></FONT></FONT></=
DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002>Dean</SPAN></FONT></FONT></FONT></FONT></FONT>=
</DIV>
<DIV><FONT lang=3D0 color=3D#000000 =
FAMILY=3D"SANSSERIF"><FONT><FONT><FONT=20
face=3DArial><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT>&nbsp;</DIV></FONT></FONT=
></FONT></BODY></HTML>

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Message-ID: <A3C2399B2FACD411A54200508BE39C74054F7121@zwcwd00r.europe.nortel.com>
From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, hgs@cs.columbia.edu,
        Brian.Rosen@marconi.com
Cc: jdrosen@dynamicsoft.com, sip@ietf.org, bcampbell@dynamicsoft.com,
        fandreas@cisco.com, jon.peterson@neustar.biz, wtm@research.att.com
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 16:42:27 +0100
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Mike,
 
My suggestion of a generic mechanism to carry Personal Data was targetting
at future work, not the present privacy draft.
 
Two points:
1) It's not just about privacy for the calling user - that's relatively
easy, you just need a mechanism to 'privately' pass information to trusted
network entities and apart from that it's up to the UA what it does/doesn't
reveal. The big problem is when other people are involved who do not have an
'agent' representing them. A subscriber to a public service is an example.
These people have rights which only the Service Provider knows about. The
Service Provider cannot be responsible for sending Personal Data of the
subscriber to the called party. Unfortunately the identity of the calling
user can often include personal data of the subscriber.
 
2) You said 'If the user voluntarily includes it, it's not private'. Sure,
it's not private *as far as the user is concerned*. But it might be private
with respect to other people. So the question is who was *responsible* for
sending it. If it's entirely the user then no problem - same as if it's the
first thing the user says after Answer. The problem is that To/From are a
grey area, since the SIP spec clearly intends that they contain the
calling/called identities (as would a spec for
'What-I-Want-You-To-Call-Me'). A network based subscriber privacy service
which did not modify the From field would be pretty rubbish, whereas one
which did not modify the Subject would probably be fine.
 
...Mark

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: 12 April 2002 16:19
To: Watson, Mark [MDN05:EP10:EXCH]; hgs@cs.columbia.edu;
Brian.Rosen@marconi.com
Cc: jdrosen@dynamicsoft.com; sip@ietf.org; bcampbell@dynamicsoft.com;
fandreas@cisco.com; jon.peterson@neustar.biz; wtm@research.att.com
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft


In a message dated 4/12/02 9:30:26 AM Eastern Daylight Time,
mwatson@nortelnetworks.com writes: 




There are many privacy services between the extremes of 'entirely up to the
UAs' and 'full B2BUA with IP address anonymiser etc.'. 
All of them require 'More Than Just A Proxy' because you need to modify the
From/To fields (unless you don't care about RFC2543 compatibility, in which
case you can modify these fields at something very like a proxy). 
Other fields may or may not need to be modified/removed - certainly
Call-Info & Reply-To would have to go. I think you *can* run the end-to-end
data argument with things like Subject and the message body. 
There is the possibility that new headers exist which carry identity
information or other Personal Data. One solution would be to remove all
unrecognised headers at the Privacy server. This will break all the new
things. 
There is another way. This would be to define now a generic capability for
carrying Personal Data, along with an indication of its privacy status. All
new services introduced later which required to send Personal Data would use
this capability. So, privacy servers could be built now which would just
remove private data from this new place, leaving unrecognised headers alone.

The privacy server would still break new services based on Personal Data -
but this is the intention of the caller. The privacy server would not break
other new things. 
...Mark 




I continue to be confused by this emphasis on the "privacy" half of the
draft. It seems to be expanding to consider "Privacy" of everything
including what color socks the user is wearing! I know that the overwhelming
thoughts in the IETF are to protect privacy, and I don't dispute that, but
the primary purpose of this draft (and the "service" it defines) is to
provide an "Identity" service, that is, providing one user's identity to the
other user. "Privacy" is the second half. We need to agree on what that
identity is and how it is provided. Then the second part of the issue is how
that identity is protected when the user doesn't want it delivered to the
other user (assuming the "network" needs it or knows it). I know this sounds
elementary and maybe some think it is obvious, but when the discussion of
"privacy" begins to define ways to treat future headers carrying other
personal data, I think it is going beyond its purpose. 

As Mark noted, a Service Provider may have responsibiity for user-supplied
information if it is implicit or required in the service that the user
supply it (to set up the call). Any extra stuff that is defined just to be
passed end-to-end (like Subject) or for a completely different service is
not a concern. If the user voluntarily includes it, it is not private. 

For example, Dean started a discussion of a 'What-I-Want-You-To-Call-Me:'
field instead of "From". To me, the fundamental difference is that the
"From" field is part of the call setup specification and the "network"
should require it, but the "What-I-Want-You-To-Call-Me" field could be see
as only being of end-to-end significance and not looked at by the "network".
The basic question for this and any field is: "Is it part of 'Identity' or
is it supplemental information?" 

Mike 




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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002>Mike,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=134303515-12042002>My 
suggestion of a generic mechanism to carry Personal Data was targetting at 
future work, not the present privacy draft.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=134303515-12042002>Two 
points:</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=134303515-12042002>1) 
It's not just about privacy for the calling user - that's relatively easy, you 
just need a mechanism to 'privately' pass information to trusted network 
entities and apart from that it's up to the UA what it does/doesn't reveal. The 
big problem is when other people are involved who do not have an 'agent' 
representing them. A subscriber to a public service is an example. These people 
have rights which only the Service Provider knows about. The Service Provider 
cannot be responsible for sending Personal Data of the subscriber to the called 
party. Unfortunately the identity of the calling user can often include personal 
data of the subscriber.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=134303515-12042002>2) 
You said 'If the user voluntarily includes it, it's not private'. Sure, it's not 
private *as far as the user is concerned*. But it might be private with respect 
to other people. So the question is who was *responsible* for sending it. If 
it's entirely the user then no problem - same as if it's the first thing the 
user says after Answer. The problem is that To/From are a grey area, since the 
SIP spec clearly intends that they contain the calling/called identities (as 
would a spec for 'What-I-Want-You-To-Call-Me'). A network based subscriber 
privacy service which did not modify the From field would be pretty rubbish, 
whereas one which did not modify the Subject would probably be 
fine.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=134303515-12042002>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> 12 April 2002 16:19<BR><B>To:</B> 
  Watson, Mark [MDN05:EP10:EXCH]; hgs@cs.columbia.edu; 
  Brian.Rosen@marconi.com<BR><B>Cc:</B> jdrosen@dynamicsoft.com; sip@ietf.org; 
  bcampbell@dynamicsoft.com; fandreas@cisco.com; jon.peterson@neustar.biz; 
  wtm@research.att.com<BR><B>Subject:</B> Re: Summary of RE: [Sip] Comment, SIP 
  Privacy draft<BR><BR></DIV></FONT><FONT face=arial,helvetica><FONT size=2>In a 
  message dated 4/12/02 9:30:26 AM Eastern Daylight Time, 
  mwatson@nortelnetworks.com writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px" 
  TYPE="CITE">There are many privacy services between the extremes of 
    'entirely up to the UAs' and 'full B2BUA with IP address anonymiser etc.'. 
    <BR>All of them require 'More Than Just A Proxy' because you need to modify 
    the From/To fields (unless you don't care about RFC2543 compatibility, in 
    which case you can modify these fields at something very like a proxy). 
    <BR>Other fields may or may not need to be modified/removed - certainly 
    Call-Info &amp; Reply-To would have to go. I think you *can* run the 
    end-to-end data argument with things like Subject and the message body. 
    <BR>There is the possibility that new headers exist which carry identity 
    information or other Personal Data. One solution would be to remove all 
    unrecognised headers at the Privacy server. This will break all the new 
    things. <BR>There is another way. This would be to define now a generic 
    capability for carrying Personal Data, along with an indication of its 
    privacy status. All new services introduced later which required to send 
    Personal Data would use this capability. So, privacy servers could be built 
    now which would just remove private data from this new place, leaving 
    unrecognised headers alone. <BR>The privacy server would still break new 
    services based on Personal Data - but this is the intention of the caller. 
    The privacy server would not break other new things. <BR>...Mark 
  <BR></BLOCKQUOTE><BR><BR>I continue to be confused by this emphasis on the 
  "privacy" half of the draft. It seems to be expanding to consider "Privacy" of 
  everything including what color socks the user is wearing! I know that the 
  overwhelming thoughts in the IETF are to protect privacy, and I don't dispute 
  that, but the primary purpose of this draft (and the "service" it defines) is 
  to provide an "Identity" service, that is, providing one user's identity to 
  the other user. "Privacy" is the second half. We need to agree on what that 
  identity is and how it is provided. Then the second part of the issue is how 
  that identity is protected when the user doesn't want it delivered to the 
  other user (assuming the "network" needs it or knows it). I know this sounds 
  elementary and maybe some think it is obvious, but when the discussion of 
  "privacy" begins to define ways to treat future headers carrying other 
  personal data, I think it is going beyond its purpose. <BR><BR>As Mark noted, 
  a Service Provider may have responsibiity for user-supplied information if it 
  is implicit or required in the service that the user supply it (to set up the 
  call). Any extra stuff that is defined just to be passed end-to-end (like 
  Subject) or for a completely different service is not a concern. If the user 
  voluntarily includes it, it is not private. <BR><BR>For example, Dean started 
  a discussion of a 'What-I-Want-You-To-Call-Me:' field instead of "From". To 
  me, the fundamental difference is that the "From" field is part of the call 
  setup specification and the "network" should require it, but the 
  "What-I-Want-You-To-Call-Me" field could be see as only being of end-to-end 
  significance and not looked at by the "network". The basic question for this 
  and any field is: "Is it part of 'Identity' or is it supplemental 
  information?" <BR><BR>Mike <BR><BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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To: "'Dean Willis'" <dean.willis@softarmor.com>, Mpierce1@aol.com,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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At the risk of repeating myself  -  well, it's never bothered me before :-)
- please see below. All comments assume a Public Service Provider model...
 

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: 12 April 2002 16:52
To: Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Mike wrote:
------------- 
 Voice over IP isn't going to escape this regulation for long. It will get
"telco-like" requirements sooner or later. It's pretty obvious to me that
there needs to be an agreement on the requirements before any more
discussion on the protocol details that might be needed to solve an
(undefined) problem.  
-------------- 
 
I don't completely disagree. In fact, I'm arguing the requirements.
 
The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch! 
 

For this discussion to make ANY sense we have to stop talking about user
device & network devices and which does what etc. etc. The only important
point is what is the *service* that the service provider is offering. If it
is a mandatory part of the service that a certain piece of personal data is
passed to the service provider, then the service provider has a
responsibility not to reveal that if the user *or subscriber* does not want
it.
 
It does not matter one iota whether the piece of data was inserted by the
user equipment of something else.
 
Basically, the functional distribution of the system is irrelevant, what is
important is what it does.
 
Now if the service involves carrying an opaque piece of data to the called
party, then obviously the only person with responsibility for that is the
person generating the data.
 
If I understand correctly, Dean, you are arguing that the To and From fields
work like this.
 
I have been arguing that this MAY not be a reasonable position for a Service
Provider when arguing their case with respect to Data Protection. This is on
the basis that the SIP spec pretty clearly implies that these fields will
contain certain information and a Service Provider could reasonably be
expected to know this.
 
Plus "a 'subscriber privacy' service which did not modify these fields would
not impress the subscriber much, as 99% of calls would go through with
personal data of the subscriber presented to the called party" - (I'm going
to put that one on a function key :-)

 
The calling-party-identity presentation infromation (CLIP or RPID) is NOT
inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of
the user. As such, the operator is responsible for the protection of that
information in a manner consistent with regulation and the expressed wishes
of the user. 
 

Absolutely. 

 
The calling-line-identity-restriction information (CLIR or Privacy) is a
means by which the user expresses wishes with respect to the handling of
NETWORK provided information.
 
This divides data into three classes:
 
1) Data inserted by a user's system, under the control of a user, and
intended to be delivered to the termination of the dialog
2) Data inserted by a user's system, under the control of a user, intended
to be delivered to an intermediate system and not beyond
3) Data inserted by the network, without the awareness of a user, intended
to be delievred to an intermediary or terminal point.
 
Service providers are by regulation in reasonable nations required to
prevent class 2 information from being leaked, and to respect the user's
preferences for delivery of class 3 data.  
 

Or rather:
1) Opaque data on which the service places no requirement or expectation
2) Data required or expected to be supplied by the user to the service
provider as part of the operation of the service
3) Data used within the service provider for the provision of the service

 
The problem is, the "normal" telephone network doesn't obviously HAVE
user-provided information in the signaling path which is intended to be
delivered to the other end, so telco people start by assuming that all
signaling is the responsibility of the service provider. But there really is
something analogous. Consider Q.SIG, user signaling at the ISDN level
generally used for PBX-to-PBX work. What Mark has been arguing is
essentially like arguing that a service provider must PREVENT the delivery
of Q.SIG data because it MIGHT compromise Data Privacy. Of course it does!
It's SUPPOSED to! The operator's responsibility is to limit the delivery of
the data to the intended party (as opposed to deliverying it to somebody
else), not to keep the user from communicating with the intended party.
 

Great choice of example ... I know just a little bit about this ...
 
Applying my argument to the Q.SIG data would imply that the subscriber has a
right to request that this data not be delivered. Well, that's exactly how
the service works, certainly according to the ITU-T standards, because the
Q.SIG data is only delivered within a closed user group. i.e. the subscriber
expresses their preference as to exactly whom the data can be delivered to
in the specification of the closed user group.
 
Indeed, the subscriber would have the right to request that the information
is never delivered, but that would rather defeat the object. As has been
pointed out a few times there is a natural trade-off between services and
privacy.
 
...Mark

 
That's right, kids -- SIP by definition is an enhanced service channel. It
does MORE than plain-old-telephone-service. If you use it carefully, you can
emulate POTS.
 
So, what are these "anoymizer" services we keep talking about? Are they
consistent with my data class division? Notice that class #1 says "to the
termination of the dialog". In using such a service, I'm requesting that my
dialog terminate at an intermediate point which implements the service of
generating a second related but anonymized dialog. In other words, the
behavior is explicitly requested by the user, and is therefore "ok with me".
 
--
Dean
 


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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>At 
the risk of repeating myself&nbsp; -&nbsp; well, it's never bothered me before 
:-) - please see below. All comments assume a Public Service Provider 
model...</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Dean Willis 
  [mailto:dean.willis@softarmor.com]<BR><B>Sent:</B> 12 April 2002 
  16:52<BR><B>To:</B> Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: 
  Summary of RE: [Sip] Comment, SIP Privacy draft<BR><BR></DIV></FONT>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT face=Arial><FONT 
  size=2><SPAN class=819102015-12042002><FONT color=#0000ff>Mike 
  wrote:</FONT></SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT face=Arial><FONT 
  size=2><SPAN class=819102015-12042002><FONT 
  color=#0000ff>-------------</FONT>&nbsp;</SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  face=Arial><FONT size=2><SPAN class=819102015-12042002>&nbsp;</SPAN>Voice over 
  IP isn't going to escape this regulation for long. It will get "telco-like" 
  requirements sooner or later. It's pretty obvious to me that there needs to be 
  an agreement on the requirements before any more discussion on the protocol 
  details that might be needed to solve an (undefined) problem.&nbsp;<SPAN 
  class=819102015-12042002><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  color=#0000ff>--------------</FONT>&nbsp;</SPAN><BR><SPAN 
  class=819102015-12042002><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  color=#0000ff>I don't completely disagree. In fact, I'm arguing the 
  requirements.</FONT></SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  color=#0000ff></FONT></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002>The point 
  I'm trying to make is that To and From are USER supplied information. 
  The&nbsp;Service Provider doesn't insert them, inspect them, or have any 
  responsibility for them. Insisting that the service provider change them is 
  just like&nbsp;insisting that the service provider change any other 
  user-supplied information,&nbsp;such as the&nbsp;words I'm typing now. 
  Anything I put in my To or from field is deliberately and explicitly intended 
  to be delivered to the terminating side of the dialog, and if as a service 
  provider you FAIL to do that, then you have DAMAGED my data, and my lawyers 
  will be in touch!<FONT face=Verdana size=1><SPAN 
  class=540223417-12042002>&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  face=Verdana size=1><SPAN 
  class=540223417-12042002></SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN class=540223417-12042002>For this discussion to make 
ANY sense we have to stop talking about user device &amp; network devices and 
which does what etc. etc. The only important point is what is the *service* that 
the service provider is offering. If it is a mandatory part of the service that 
a certain piece of personal data is passed to the service provider, then the 
service provider has a responsibility not to reveal that if the user *or 
subscriber* does not want 
it.</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>It 
does not matter one iota whether the piece of data was inserted by the user 
equipment of something else.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002>Basically, the functional distribution of the system is 
irrelevant, what is important is what it does.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>Now 
if the service involves carrying an opaque piece of data to the called party, 
then obviously the only person with responsibility for that is the person 
generating the data.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>If I 
understand correctly, Dean, you are arguing that the To and From fields work 
like this.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>I 
have been arguing that this MAY not be a reasonable position for a Service 
Provider when arguing their case with respect to Data Protection. This is on the 
basis that the SIP spec pretty clearly implies that these fields will contain 
certain information and a Service Provider could reasonably be expected to know 
this.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=540223417-12042002>Plus 
"a 'subscriber privacy' service which did not modify these fields would&nbsp;not 
impress the subscriber much, as 99% of calls would go through with personal data 
of the subscriber presented to the called party" - (I'm going to put that one on 
a function key :-)</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002>The 
  calling-party-identity presentation infromation (CLIP or RPID) is NOT inserted 
  by the user. It is inserted by the network, WITHOUT THE CONSENT of the user. 
  As such, the operator is responsible for the protection of that information in 
  a manner consistent with regulation and the expressed wishes of the user.<FONT 
  face=Verdana size=1><SPAN 
  class=540223417-12042002>&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  face=Verdana size=1><SPAN 
  class=540223417-12042002></SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN 
class=540223417-12042002>Absolutely.&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>The calling-line-identity-restriction information 
  (CLIR or Privacy) is a means by which the user expresses wishes with respect 
  to the handling of NETWORK provided 
  information.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>This divides data into three 
  classes:</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>1)&nbsp;Data inserted by&nbsp;a user's system, under 
  the control of&nbsp;a user, and intended to be delivered to the termination of 
  the dialog</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>2)&nbsp;Data inserted by&nbsp;a user's system, under 
  the control of&nbsp;a user, intended to be delivered to an intermediate system 
  and not beyond</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>3) Data inserted by the network, without the 
  awareness of&nbsp;a user, intended to be delievred to an intermediary or 
  terminal point.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002>Service 
  providers are by regulation in reasonable nations required to prevent class 2 
  information from being leaked, and to respect the user's preferences for 
  delivery of class 3 data.&nbsp;<FONT face=Verdana size=1><SPAN 
  class=540223417-12042002>&nbsp;</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
  face=Verdana size=1><SPAN 
  class=540223417-12042002></SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN class=540223417-12042002>Or 
rather:</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN class=540223417-12042002>1)&nbsp;Opaque data on which 
the service places no requirement 
or&nbsp;expectation</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN class=540223417-12042002>2) Data required or expected 
to be supplied by the user to the service provider as part of the operation of 
the service</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
size=+0><FONT face=Arial><FONT size=2><SPAN class=819102015-12042002><FONT 
face=Verdana size=1><SPAN class=540223417-12042002>3) Data used within the 
service provider for the provision of the 
service</SPAN></FONT></SPAN></FONT></FONT></FONT></FONT></FONT></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>The problem is, the "normal" telephone network 
  doesn't&nbsp;obviously HAVE user-provided information in the signaling path 
  which is intended to be delivered to the other end, so telco people start by 
  assuming that all signaling is the responsibility of the service provider. But 
  there really is something analogous. Consider Q.SIG, user signaling at the 
  ISDN level generally used for PBX-to-PBX work. What Mark has been arguing is 
  essentially like arguing that a service provider must PREVENT the delivery of 
  Q.SIG data because it MIGHT compromise Data Privacy. Of course it does! It's 
  SUPPOSED to! The operator's responsibility is to limit the delivery of the 
  data to the intended party (as opposed to deliverying it to somebody else), 
  not to keep the user from communicating with the intended 
  party.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002>Great choice of example ... I know just a little bit 
about this ...</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002>Applying my argument to the Q.SIG data would imply that 
the subscriber has a right to request that this data not be delivered. Well, 
that's exactly how the service works, certainly according to the ITU-T 
standards, because the Q.SIG data is only delivered within a closed user group. 
i.e. the subscriber expresses their preference as to exactly whom the data can 
be delivered to in the specification of the closed user 
group.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002>Indeed, the subscriber would have the right to request 
that the information is never delivered, but that would rather defeat the 
object. As has been pointed out a few times there is a natural trade-off between 
services and privacy.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=540223417-12042002>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>That's right, kids -- SIP by definition is an 
  enhanced service channel. It does MORE than plain-old-telephone-service. If 
  you use it carefully, you can emulate 
  POTS.</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>So, what are these "anoymizer" services we keep 
  talking about? Are they consistent with my data class division? Notice that 
  class #1 says "to the termination of the dialog".&nbsp;In using such a 
  service, I'm requesting that my dialog terminate at an intermediate point 
  which implements the service of generating a second related but anonymized 
  dialog. In other words, the behavior is explicitly requested by the user, and 
  is therefore "ok with me".</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>--</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002>Dean</SPAN></FONT></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
  size=+0><FONT face=Arial><FONT color=#0000ff size=2><SPAN 
  class=819102015-12042002></SPAN></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE></FONT></FONT></FONT></BODY></HTML>

------_=_NextPart_001_01C1E24A.B68AA4A2--

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Hi,

Please read and comment on sec-agree.  We hummed to make this a WG item,
so we should get a few eyeballs on it sooner rather than later.

http://search.ietf.org/internet-drafts/draft-arkko-sip-sec-agree-01.txt

thanks,
-rohan



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We need to quickly review the SIM Message Method Extension. This draft
has been widely discussed, iterated several times in SIMPLE, and is on
its 2nd revision in SIP. The draft does not appear to be controversial.

See:

http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt

We expect to move to IETF last call next week, so please raise any
issues immediately.

Ben Campbell will coordinate for the draft editors.


--
Dean


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In a message dated 4/12/02 11:42:49 AM Eastern Daylight Time, 
mwatson@nortelnetworks.com writes:


> Mike,
>  
> My suggestion of a generic mechanism to carry Personal Data was targetting 
> at future work, not the present privacy draft.

Yes, I suspected that, but then I noticed that Tom picked up on it as an 
"very interesting idea " and seemed to be suggesting that we run with it.

>  
> Two points:
> 1) It's not just about privacy for the calling user - that's relatively 
> easy, you just need a mechanism to 'privately' pass information to trusted 
> network entities and apart from that it's up to the UA what it does/doesn't 
> reveal. The big problem is when other people are involved who do not have 
> an 'agent' representing them. A subscriber to a public service is an 
> example. These people have rights which only the Service Provider knows 
> about. The Service Provider cannot be responsible for sending Personal Data 
> of the subscriber to the called party. Unfortunately the identity of the 
> calling user can often include personal data of the subscriber.
>  
> 2) You said 'If the user voluntarily includes it, it's not private'. Sure, 
> it's not private *as far as the user is concerned*. But it might be private 
> with respect to other people. So the question is who was *responsible* for 
> sending it. If it's entirely the user then no problem - same as if it's the 
> first thing the user says after Answer. The problem is that To/From are a 
> grey area, since the SIP spec clearly intends that they contain the 
> calling/called identities (as would a spec for 
> 'What-I-Want-You-To-Call-Me'). A network based subscriber privacy service 
> which did not modify the From field would be pretty rubbish, whereas one 
> which did not modify the Subject would probably be fine.
> 

I'm confused by what you are getting at by "subscribers" and "other people" 
(who don't have an "agent"), and "calling user". There is a SIP phone 
(containing a "User Agent") and it might be used by multiple people to place 
calls. If people don't have their own identity, then every call must use the 
identity of the physical device. If there is a capability for each person to 
provide a separate identity, then that is what is used by the "network" as 
the basis for a "subscription" to service and to apply privacy.

If you are referring to the complicated cases of how to apply "privacy" to 
identities involved in 3-way services such as call transfer and call 
forwarding, then I understand the difficulties, having been through 
excrutiating discussions in standards development for ISDN. It isn't easy or 
pretty. But these issues will only be worked out in a specific discusion of 
the interworking of the identity/privacy service with the call transfer/call 
forwarding services, something that the IETF is intent on not doing.

I still say if the user voluntarily includes something beyond the specific 
identity fields, then the network has no obligation to apply privacy to it. I 
don't see a case in which the network has to do anything else to provide 
privacy for "other people". Again, maybe there is a difference in what "user" 
refers to in your point 2) above. If the "standard" service operation (in the 
User Agent or network) puts in some identity, or passes along one it got 
during Call Transfer, then of course privacy rules must apply to it. That's 
not the user (person) voluntarily putting in the information.

Mike

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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">In a message dated 4/12/02 11:42:49 AM Eastern Daylight Time, mwatson@nortelnetworks.com writes:
<BR>
<BR>
<BR></FONT><FONT  COLOR="#0000ff" SIZE=1 FAMILY="SANSSERIF" FACE="Verdana" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Mike,</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0"> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=1 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">My suggestion of a generic mechanism to carry Personal Data was targetting at future work, not the present privacy draft.</FONT><FONT  COLOR="#0000ff" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">Yes, I suspected that, but then I noticed that Tom picked up on it as an "very interesting idea " and seemed to be suggesting that we run with it.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"> </FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">Two points:</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=1 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">1) It's not just about privacy for the calling user - that's relatively easy, you just need a mechanism to 'privately' pass information to trusted network entities and apart from that it's up to the UA what it does/doesn't reveal. The big problem is when other people are involved who do not have an 'agent' representing them. A subscriber to a public service is an example. These people have rights which only the Service Provider knows about. The Service Provider cannot be responsible for sending Personal Data of the subscriber to the called party. Unfortunately the identity of the calling user can often include personal data of the subscriber.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=1 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">2) You said 'If the user voluntarily includes it, it's not private'. Sure, it's not private *as far as the user is concerned*. But it might be private with respect to other people. So the question is who was *responsible* for sending it. If it's entirely the user then no problem - same as if it's the first thing the user says after Answer. The problem is that To/From are a grey area, since the SIP spec clearly intends that they contain the calling/called identities (as would a spec for 'What-I-Want-You-To-Call-Me'). A network based subscriber privacy service which did not modify the From field would be pretty rubbish, whereas one which did not modify the Subject would probably be fine.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Verdana" LANG="0">
<BR></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I'm confused by what you are getting at by "subscribers" and "other people" (who don't have an "agent"), and "calling user". There is a SIP phone (containing a "User Agent") and it might be used by multiple people to place calls. If people don't have their own identity, then every call must use the identity of the physical device. If there is a capability for each person to provide a separate identity, then that is what is used by the "network" as the basis for a "subscription" to service and to apply privacy.
<BR>
<BR>If you are referring to the complicated cases of how to apply "privacy" to identities involved in 3-way services such as call transfer and call forwarding, then I understand the difficulties, having been through excrutiating discussions in standards development for ISDN. It isn't easy or pretty. But these issues will only be worked out in a specific discusion of the interworking of the identity/privacy service with the call transfer/call forwarding services, something that the IETF is intent on not doing.
<BR>
<BR>I still say if the user voluntarily includes something beyond the specific identity fields, then the network has no obligation to apply privacy to it. I don't see a case in which the network has to do anything else to provide privacy for "other people". Again, maybe there is a difference in what "user" refers to in your point 2) above. If the "standard" service operation (in the User Agent or network) puts in some identity, or passes along one it got during Call Transfer, then of course privacy rules must apply to it. That's not the user (person) voluntarily putting in the information.
<BR>
<BR>Mike</FONT></HTML>

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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
To: dean.willis@softarmor.com, Brian.Rosen@marconi.com, sip@ietf.org
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In a message dated 4/12/02 11:53:16 AM Eastern Daylight Time, 
dean.willis@softarmor.com writes:


> The point I'm trying to make is that To and From are USER supplied 
> information. The Service Provider doesn't insert them, inspect them, or 
> have any responsibility for them. Insisting that the service provider 
> change them is just like insisting that the service provider change any 
> other user-supplied information, such as the words I'm typing now. Anything 
> I put in my To or from field is deliberately and explicitly intended to be 
> delivered to the terminating side of the dialog, and if as a service 
> provider you FAIL to do that, then you have DAMAGED my data, and my lawyers 
> will be in touch!
>  
> The calling-party-identity presentation infromation (CLIP or RPID) is NOT 
> inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of 
> the user. As such, the operator is responsible for the protection of that 
> information in a manner consistent with regulation and the expressed wishes 
> of the user.
>  
> The calling-line-identity-restriction information (CLIR or Privacy) is a 
> means by which the user expresses wishes with respect to the handling of 
> NETWORK provided information.
> 

In a message dated 4/12/02 11:33:03 AM Eastern Daylight Time, 
Brian.Rosen@marconi.com writes:


> The problem with this line of thinking is that in the PSTN, identity is 
> always asserted by
> the network, and never by the UA.  In SIP, presently, identity is always 
> asserted by
> the UA and never the network.  Privacy when you have network asserted 
> identity
> means something different than when you have user asserted identity.
> 

There seems to be some confusion about the current "PSTN" situation. True, in 
the POTS PSTN (analog lines) the user doesn't actually include their identity 
in the signaling to the network. But, since the identity is directly 
associated with the physical phone line, it's the same thing. The telephone 
company simply converts its knowledge of the identity of the wire into a 
telephone number and goes from there.

In ISDN, on the other hand, the user can include their identity (phone 
number) in the signaling, since each physical interface and each device can 
support multiple telephone numbers. The number is put in the Calling Party 
Number field. The network may screen this number to make sure it is a valid 
number for that interface. Depending on the result of the screening, the 
network may include multiple numbers in the call set up, for example, an 
unverified user provided number and the network provided valid number.

The user may specify by per-call signaling or by a permanent subscription 
option whether or not they want the network to apply the restriction 
("privacy") feature to this identity (whether the identity came from the user 
or was added by the network or both doesn't matter). There is no attempt to 
also apply "privacy" if the user chooses to put some other identifying 
information in the User-to-user field.

We have to decide whether the From/To fields are like the Calling Party 
Number field in ISDN (meant for user-network signaling and left to the 
network to change if required by the service) or are they like the 
User-to-user field in ISDN (never touched by the network). I've heard both 
sides of the argument being made, so there is apparently not a consistent 
view on how basic SIP works.

It seems that the added RPID field is similar to the network added (verified) 
number in ISDN.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/12/02 11:53:16 AM Eastern Daylight Time, dean.willis@softarmor.com writes:
<BR>
<BR>
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">The point I'm trying to make is that To and From are USER supplied information. The Service Provider doesn't insert them, inspect them, or have any responsibility for them. Insisting that the service provider change them is just like insisting that the service provider change any other user-supplied information, such as the words I'm typing now. Anything I put in my To or from field is deliberately and explicitly intended to be delivered to the terminating side of the dialog, and if as a service provider you FAIL to do that, then you have DAMAGED my data, and my lawyers will be in touch!</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">The calling-party-identity presentation infromation (CLIP or RPID) is NOT inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of the user. As such, the operator is responsible for the protection of that information in a manner consistent with regulation and the expressed wishes of the user.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">The calling-line-identity-restriction information (CLIR or Privacy) is a means by which the user expresses wishes with respect to the handling of NETWORK provided information.</FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>In a message dated 4/12/02 11:33:03 AM Eastern Daylight Time, Brian.Rosen@marconi.com writes:
<BR>
<BR>
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">The problem with this line of thinking is that in the PSTN, identity is always asserted by</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">the network, and never by the UA. &nbsp;In SIP, presently, identity is always asserted by</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">the UA and never the network. &nbsp;Privacy when you have network asserted identity</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">means something different than when you have user asserted identity.</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></BLOCKQUOTE>
<BR>
<BR>There seems to be some confusion about the current "PSTN" situation. True, in the POTS PSTN (analog lines) the user doesn't actually include their identity in the signaling to the network. But, since the identity is directly associated with the physical phone line, it's the same thing. The telephone company simply converts its knowledge of the identity of the wire into a telephone number and goes from there.
<BR>
<BR>In ISDN, on the other hand, the user can include their identity (phone number) in the signaling, since each physical interface and each device can support multiple telephone numbers. The number is put in the Calling Party Number field. The network may screen this number to make sure it is a valid number for that interface. Depending on the result of the screening, the network may include multiple numbers in the call set up, for example, an unverified user provided number and the network provided valid number.
<BR>
<BR>The user may specify by per-call signaling or by a permanent subscription option whether or not they want the network to apply the restriction ("privacy") feature to this identity (whether the identity came from the user or was added by the network or both doesn't matter). There is no attempt to also apply "privacy" if the user chooses to put some other identifying information in the User-to-user field.
<BR>
<BR>We have to decide whether the From/To fields are like the Calling Party Number field in ISDN (meant for user-network signaling and left to the network to change if required by the service) or are they like the User-to-user field in ISDN (never touched by the network). I've heard both sides of the argument being made, so there is apparently not a consistent view on how basic SIP works.
<BR>
<BR>It seems that the added RPID field is similar to the network added (verified) number in ISDN.
<BR>
<BR>Mike
<BR></FONT></HTML>

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From daemon@ns.ietf.org  Fri Apr 12 15:47:04 2002
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From: "Shankara, Udaya" <u_shankara@trillium.com>
To: "'sip-implementors@cs.columbia.edu,'"<sip-implementors@cs.columbia.edu>, >,
        "'sip@ietf.org'" <sip@ietf.org>
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Subject: [Sip] Call establishment over TCP
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Hi,
    Let us consider a call between 2 UAs over TCP:
       UA1   <--------->    UA2

UA1 sends an INVITE to UA2 over TCP (channel 1). The Contact header of
INVITE contains UA1's IP address and port 8000. UA2 responds to INVITE over
channel 1 and ACK is also sent over the same channel.

   Let us assume that UA2 determines to terminate the call. The question is:
 
Does UA2 send BYE over channel 1 or does it open a new TCP connection
(channel 2) to the server running at port 8000?

Regards,
Udaya


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Date: Fri, 12 Apr 2002 21:31:58 +0300
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: <Mpierce1@aol.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Dean Willis writes:

 > The point I'm trying to make is that To and From are USER supplied
 > information. The Service Provider doesn't insert them, inspect them, or
 > have any responsibility for them.

if someone subscribes to my sip service, i DO check for each message
received from the user that the from field corresponds to what i have
agreed with the user.  otherwise the user could fake to be somone else
that i consider a bad thing.  if the from field doesn't match, i don't
forward the message.  if that is not acceptable for the user, the use
can choose another service provider or roll its own service.  ii do not
modify anything.

 > Anything I put in my To or from field is deliberately and explicitly
 > intended to be delivered to the terminating side of the dialog, and if
 > as a service provider you FAIL to do that, then you have DAMAGED my
 > data, and my lawyers will be in touch!

i'm not damaging your data.  i simply don't serve your message and you
have no way to win the case in court.

-- juha


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Date: Fri, 12 Apr 2002 12:24:59 -0700 (PDT)
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org,
        Ben Campbell <bcampbell@dynamicsoft.com>,
        Flemming Andreasen <fandreas@cisco.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'William Marshall'" <wtm@research.att.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Henning Schulzrinne writes:
 > I agree that, in general, the UA should be treated like a mature adult
 > and get to make decisions as to what information it wants to include in
 > its call setup requests or not. I don't agree that anonymity requires a
 > B2BUA in all cases. For example, one could imagine that a large ISP
 > could hand out random IP addresses from a large pool, making it
 > essentially similar to an anonymizer service. 
  
   BTW, IPv6 already allows for private (read: formed
   dynamically and randomly by the host) IP addresses.
   The ISP doesn't even need to "allow" it or provide
   for it.

 > There is no principal
 > difference between knowing that "user came from anonymizer service
 > anon.com" and "user came from ISP large-isp.com". (Well, almost: it may
 > be easier to switch anonymity providers than ISPs.)

   This all gets down to your paranoia level. Prefixes
   do tell you something about IP topology which could
   lead to educated guesses -- which may be all that's
   necessary. Hopefully this level of paranoia will only
   be needed in the direst of cases like mob snitches, etc.

	     Mike

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
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        <bcampbell@dynamicsoft.com>
Date: Fri, 12 Apr 2002 14:16:41 -0500
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Oops.

Please review :

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-message-02.txt

Until the -02 makes it into the draft repository. They're not as fast as
I'm wishing for (ok, light speed is too slow, while we're at it!)

Thanks,

--
Dean


Original message:
-------------
We need to quickly review the SIM Message Method Extension. This draft
has been widely discussed, iterated several times in SIMPLE, and is on
its 2nd revision in SIP. The draft does not appear to be controversial.

See:

http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt

We expect to move to IETF last call next week, so please raise any
issues immediately.

Ben Campbell will coordinate for the draft editors.

-----------------


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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Dean Willis" <dwillis@dynamicsoft.com>, <sip@ietf.org>
Cc: <rohan@cisco.com>, <brian.rosen@marconi.com>, <jo@ipdialog.com>
Date: Fri, 12 Apr 2002 14:12:16 -0500
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Actually, I just submitted draft-ietf-sip-message-02.txt yesterday. I
_thought_ that was the one we were going to WGLC. Since it does not seem to
have made the repository, I will send a link to it in a second message (any
moment now)

> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Friday, April 12, 2002 1:51 PM
> To: sip@ietf.org
> Cc: rohan@cisco.com; bcampbell@dynamicsoft.com; brian.rosen@marconi.com;
> jo@ipdialog.com
> Subject: SIP Message Draft Fast WGLC
>
>
>
> We need to quickly review the SIM Message Method Extension. This draft
> has been widely discussed, iterated several times in SIMPLE, and is on
> its 2nd revision in SIP. The draft does not appear to be controversial.
>
> See:
>
> http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt
>
> We expect to move to IETF last call next week, so please raise any
> issues immediately.
>
> Ben Campbell will coordinate for the draft editors.
>
>
> --
> Dean


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From daemon@ns.ietf.org  Fri Apr 12 16:37:05 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <Mpierce1@aol.com>, <Brian.Rosen@marconi.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Fri, 12 Apr 2002 14:37:40 -0500
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Mike wrote:
---------------------- 
There seems to be some confusion about the current "PSTN" situation.
True, in the POTS PSTN (analog lines) the user doesn't actually include
their identity in the signaling to the network. But, since the identity
is directly associated with the physical phone line, it's the same
thing. The telephone company simply converts its knowledge of the
identity of the wire into a telephone number and goes from there. 

In ISDN, on the other hand, the user can include their identity (phone
number) in the signaling, since each physical interface and each device
can support multiple telephone numbers. The number is put in the Calling
Party Number field. The network may screen this number to make sure it
is a valid number for that interface. Depending on the result of the
screening, the network may include multiple numbers in the call set up,
for example, an unverified user provided number and the network provided
valid number. 

The user may specify by per-call signaling or by a permanent
subscription option whether or not they want the network to apply the
restriction ("privacy") feature to this identity (whether the identity
came from the user or was added by the network or both doesn't matter).
There is no attempt to also apply "privacy" if the user chooses to put
some other identifying information in the User-to-user field. 

We have to decide whether the From/To fields are like the Calling Party
Number field in ISDN (meant for user-network signaling and left to the
network to change if required by the service) or are they like the
User-to-user field in ISDN (never touched by the network). I've heard
both sides of the argument being made, so there is apparently not a
consistent view on how basic SIP works. 

It seems that the added RPID field is similar to the network added
(verified) number in ISDN. 
------------------------ 

Excellent summary, Mike. That's exactly what I'm saying.

In short:

To:/From: are equivalent to ISDN user-to-user data, as is most of
current SIP.

RPID: is analogous to CPN, and its "Screened" parameter analogous to
what happens in ISDN today.

Privacy: is analogous to the calling line identification restriction
flag.
 
I think an understanding of this relationship should clear up 99% of the
debate over the "privacy" draft, which is really much more of a SIP-T
"telephony interworking" document than it is a general privacy document.

--
dean


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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[Trimming privacy-threatening cc lines, revealing list membership :-)]

I think part of the problem is, as I think Dean and Brian have alluded
to, is that many things that used to be tied together in one mechanism
in the PSTN are now split across several different ones:

- From display name
- From URL
- RPID
- user authentication name (Authorization)
- Organization
- Call-Info

Also, to pick up the "disable CLID and disallow anonymous calls": I
think part of this has to do with the commingling of identity and the
ability to call back. I suspect that even the Hollywood mogul wouldn't
mind if his name appeared on the starlets display screen, but wants to
maintain the "don't call us, we'll call you" attitude, i.e., avoid
revealing a callable number.

I don't think anticipating regulation is particularly helpful here.
Anybody can say anything about what some legislature somewhere may
legislate at some point in the future.

I wouldn't be too surprised if, instead of PSTN legislation, some of the
email spam legislation finds its way into SIP behavior, e.g., having to
declare a truthful return address, an indication of the call intent
(like some of the laws and bills calling for adding ADV to the subject
line) or including an opt-out link (header?). If we could make progress
there, I think we'd do more for "privacy" than hiding a user name.

Henning

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From daemon@ns.ietf.org  Fri Apr 12 17:42:10 2002
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Date: Fri, 12 Apr 2002 16:44:47 -0500
From: Chris Martin <cmartin@dasecurenetworks.com>
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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"Rosen, Brian" wrote:
> 
> >> The UA is by far in the best position to determine the level of
> >> anonymity it wishes, and if it can't get enough (because, for example,
> >> it wants it's IP addresses obscured), it will have to employ a
> >> B2BUA as above.
> >
> >Agreed. But the UA is not in a good position to determine the
> >subscribers anonymity requirements, or other parties not represented
> >by a UA. In a Service Provider model, these people have some rights.
> >So, the network may have to employ a B2BUA or whatever on behalf of
> >these people. There is no problem with this and so no case for
> >obfuscating the From/To fields - agreed ?
> 
> Agreed.  Don't forget the obvious problem of any crypto (signatures
> for example) on any of these fields!

Yep, current B2BUA's would have problems if encryption or hashing is
done for many of the fields.

The development of SIPS is an example where current B2BUA's may run into
issues as SIPS is developed and standardized, unless they are also able
to verify hash or provide decryption/encrytpion as needed to provide the
service.

Chris
> 
> Brian
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From daemon@ns.ietf.org  Fri Apr 12 17:43:13 2002
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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> Mark Watson wrote:
> 
> Brian wrote:
> 
> > To me, the service that breaks with anonymous From/To is
> > the ability to call me.  If my phone rings, and the "From"
> > which is displayed says "anonymous", I'm not going to answer it.
> 
> Very sensible. It could be anyone!
> 
> You might answer it if the call came from a Service Provider that you
> trusted, and that Service Provider had mechanisms to verify and record
> the identity of the caller, and to pass that identifier to the
> authorities if the call turns out to be malicious.

Speaking in terms of service provider, maybe that is where additional
headers should be required (Between B2BUA and/or SP/enterprise sip
proxies, rather than endpoints, for the interdomain traffic, in order to
hand off the "true" identity. Since hosts of service providers would
need to go through the registration and authentication process prior to
initiating sessions (for that point the SP customer would have to go
through a sign up for services process prior to be allowed to register).

> 
> There is always a balance between the desire of the caller to remain
> anonymous and of the called party to reject anonymous calls. Without
> an intermediary then the balance gets tipped completely to one side or
> the other.
> 
> > I'll probably instruct my voicemail service to do the same.
> > The Police Tip line might very well accept such calls however.
> >
> > I'm reminded of the practices of some Hollywood types that
> > block Caller-Id, but won't accept calls that don't provide
> > Caller-Id!
> >

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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I incorporated a few dozen suggested corrections into the Path draft. It
has been submitted to the internet-drafts queue and should appear there
shortly. In the interim, it should be available in the following
locations:

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-03.txt

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-03.html

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-03.xml

If you have extensive edits to suggest, please make them relative to the
xml file. I'm using the xml2rfc package as a formatter, so this is the
"source code".

Note that the header is still called "Path". The suggested
"Register-Record-Route" was just too long and nobody really cheered for
my suggestion, "RRR". The topic is still open.

Main changes were ordering the Path elements in the same direction as
Record-Route elements (they were apparently backward) and cleanup of
typos. As requested by Miguel, I did a "Table 2" addition. The
references have also been divided into normative and non-normative
sections as currently required.

Enjoy.

--
Dean


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: "Dean Willis" <dwillis@dynamicsoft.com>
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Minutes edited by Dean Willis from notes taken by Vijay Gurbani, 
Joerg Ott, and Renee Cohen.

Session 1
	
19:30 CST meeting started 
      Agenda accepted 
      Chairs discuss "Note Well" 2026 notice
	

Work Plan update:

     We have revised the SIP spec; went to IESG; resulted in a "Yea"
     from IESG. SIP events spec is done

     Session timer is back to haunt us -- goes back to authors for bis
     update.

     Caller pref - pending from author for bis. 

     Precondition extensions -- in WGLC now; in IESG by April SIP 

     Privacy spec from DCS in WGLC now, IESG by April 

     REFER Method; needs bis, security updates. IESG by May? No
     complaints from authors. More important since there are many
     implementations already.

     MESSAGE method ready for WG LC -- have not done it yet. 

     PATH method needs rev from editor. Call for volunteers. 

     NAT awareness: rev pending from author. 

     SIP Privacy and Security reqs to IESG? There is a feeling that
     the SIP extensions for privacy wrt to 3GPP is not achievable --
     Jon P: some privacy (user provided privacy vs. network provided
     privacy) nuances have not been captured as of today. Henning:
     that would make a lot of sense if we know what we wanted --
     another req document is not in and of itself useful. Dean: SIP
     privacy and security reqs predate the SIPPING WG -- IESG felt
     that it is such a critical piece that it should stay in SIP WG.

     SIP over SCTP? Gonzalo: we are ready for WG LC for
     SIP/SCTP. Dean: Also in July timeframe -- have a draft standard
     version of SIP -- #1 goal in July of 2003. Original SIP spec was
     2543, we want to claim new number of 3543 :-)

     State: Charter item of pushing certain things off the SIP
     signaling state -- state or cookies specification translated to
     SIP. Did we ever come to a consensus on how to go forward? Rohan:
     Cookies I-D is good for doing what you want to do with the state
     I-D and is more general. I support it. Dean: Anyone know of
     implementations? [No implementations yet] Jonathan Rosenberg: The
     reason no one has implemented it is because everyone is using R-R
     and Contact. It is not entirely clear to me that this is even
     needed. Flemming: The difference is that R-R requires the proxy to
     be in all signaling. Andrew Zmolek: We looked at R-R issue and it
     seemed that we were going to have some trouble -- either the
     state or the cookies would do fine JDR: R-R was not sufficient
     since there was no reliable way to put something and get it
     back. With the new R-R update, this is no longer an
     issue. rjsparks: You cannot change the state in a dialog, you can
     only push and get the state. JDR: Okay, if that is the
     requirement, then fine -- if the req is to just push state for
     the purpose of a dialog, we have a mechanism already in R-R,
     Contact. Brian Rosen: we may work on a requirement document
     offline, assuming that it is useful.

     General Scheduling: Henning: meta scheduling aspect -- can you
     get people involved long enough to remember what the issue was?
     Brian: Our goal is to get a lot of stuff out that has been
     hanging around for a long time -- but we do not want to get out
     12 LC I-Ds at the same time. We will revive the LC schedule we
     had going. Henning: Do other I-Ds that are not on your list wait
     until re- chartering? Brian: No. There are some things that are
     hanging around for a long time; as long as the ADs do not breath
     down on us, we will work on them as we go along. Henning: Need a
     priority list. Jonathan: Learn from successes -- Bundle 1 was a
     success delivery to 3GPP. With the current mechanism we were
     randomly LC'ing. One of the thing the Bundle did is to focus
     people's energy on that. Brian: We can consider that; Bundle 1's
     advantage was that the drafts were related to one another. These
     do not.

     What do we do with: sipping-conferencing-models? sip-3pcc?
     app-components? sip-vxml? Jonathan: need to finish bis update;
     but done as far as I know.  Rohan: 3PCC is a very pure usage
     draft describing the usage of baseline bis offer answer
     model. Most of the stuff above is usage or framework, we should
     get it done in SIPPING. Brian Rosen: You are basically saying
     what Jonathan said: Put it in Bundles. Rohan: Sure.
 

SIP Change process, Allison Mankin:

     This is not a SIP draft; it is an individual transport area
     draft. People have said that there is a need to control SIP
     information. The Replaces header was done in a freeform manner --
     this is not good. WG discipline needed over extensions of
     SIP. RFC 3261 (new bis) has an IANA consideration which is very
     different then what you have seen before. You need a standards
     track RFC for headers, method, response codes, warning codes. One
     place you do not need to do this is for the Events RFC -- just
     need WG yes for these. serverfeatures did not go too far with
     IESG since it offered unbridled extensions. We now have P-header
     (not X-header, more constrained). Still need a RFC, but you do
     not have to have the buy in of SIPPING or SIP. The string with P-
     is reserved if they are to have a future life. Henning: while I
     agree with the notion, the naming has the same problems that X
     headers had. Attaching a meaning to "P-" is not good. Allison:
     They do not have option tags and have to have applicability
     statements. Henning: There are 2 issues: naming and
     process/applicability. If we have a header name which was
     registered (say, foo); that header has the property that as long
     as it is in the non-RFC track, it looks like a normal header. If
     it reaches standards track, it retains its name. Lets say P-bar
     header becomes popular and widely implemented. Now, if P-bar
     header goes to standards track, they will have to rename this
     header. Allison: can you make a P-header a standards track header
     -- add an option tag. Dean: The P- name will still be registered
     and be useful. Now you will basically have to track P- and non-P
     extension headers -- makes the symbol table little large --
     that's okay. Dave Oran: feel uncomfortable in mixing naming
     conventions and algorithmic behavior. I do not like the idea of
     having to parse inside header string. Jonathan Rosenberg: The
     name of the option tag is unrelated to the name of the
     header. Henning: HTTP extension model is different then SIP
     extension model. There is no correlation in header names and
     option tags. Allison: If the I-D has any implications of this
     sort, we can fix it. Gonzalo: We have option tags that have no
     headers associated with them. Keith Drage: We need to make sure
     that this I-D does not contain any requirements on SIP
     implementation -- a SIP implementer must not have to read this
     I-D.

 

The UPDATE method - Jonathan Rosenberg :

     Open issues 

     1) Glare with PRACK - UPDATE only specifies glare resolution with
        itself. You can have glare with PRACK. Rejecting PRACK is
        bad. Solution: can't send UPDATE if you have sent an answer in
        18x for which you have not gotten a PRACK. Will put some words
        with general caveats.
    
     2) Repairable response codes -- automata can fix these without
        human intervention. What about 493 Undecipherable? May require
        user intervention to fix it. Proposal: include it, add text
        saying it retries if it would otherwise retry with that
        response. [No one objected].

     3) Generate 155 instead of 4xx MAY or SHOULD -- for backward
        compatibility. SHOULD is better if the UAS supports this
        capability, the proxy may not. This makes it work at proxies
        transparently. Comment: This text is screaming for a reason
        header, otherwise the UAC will have to infer what the problem
        is. JDR: I did not talk about reason header for a reason -- it
        is on a slower track. For the basic cases we are worried about
        immediately, the UAC can infer from the headers. Going
        forward, the reason header is the way to go. Gonzalo: The last
        review of the reason header already has this, so we can use
        it. Jonathan: Does the group agree that the message sip (or
        sip frag) is the appropriate approach? Or wait for the reason
        header (which is on a slower track). Rohan: Can we pull out
        155 out of this, then? [No consensus on this]

 

Manyfolks open issues (Gonzalo Camarillo)
draft-ietf-sip-manyfolks...-05.txt 

     We are now defining a framework for preconditions of different
     types. We define the current status of the precond vs. desired
     status. We always know if current status is better or worse then
     desired status. Two status types: e2e -- always present in
     manyfolks (-04). Segmented status type introduced in -04.

     Open issues: Meaning of Require: precondition -- 2 approaches: I
     refuse everything I don't understand. Or be liberal and accept
     the offer if the preconditions can be met without your
     intervention? Which is better? 1 or 2? Comment: you can have a
     thing called "criticality" which gives a hint on what to
     do. Gonzalo: I will decide after speaking to Mark.

 

Reason code, Gonzalo Camarillo.:

     Requirements: same functionality needed in several WG items --
     why is this request (or response) being sent?

     Useful in many works: ISUP/SIP mapping, in manyfolks
     (precondition failure, unacceptable here), HERFP, 3pcc.

     Jonathan: Throw in another use: in the event that you fork the
     request to a bunch of phones and one of them picks up. The proxy
     generates CANCEL. The reason for CANCEL is not because the user
     hung up, but because 1 of N answered.

     Rohan: Lot of overlap in the reqs that generated this document
     and request history.

     Eric Burger: This is H.450 all over again.

     Jonathan R: Don't we have an enumerated list of response code in
     the bis already? This is exactly that, and then some.

     Comment: we have one address space for responses, and we have
     just added one more with the Reason header. Has a kitchen-sink
     feeling to it.

     Henning: The motivation was exactly to prevent reinventing the
     same thing every time. We are not adding new error classes that
     will have to be percolated to all existing implementation. This
     is a fine grained status code which is there if you need
     it. Example: Q.850 error code will not be pertinent to many
     implementations, but to the one that it is pertinent to, it can
     use it without too much perturbation.

     Brian Rosen: What do we do now? 2 possibilities: crisp set of
     reqs, which are clear and this is a reasonable solution. Or we do
     not have a crisp set of reqs. We need to determine this
     first. Lot of discussion on if this does or does not solve the
     job. But do we know what the job is? Should we push this back
     into sipping and generate a req document? Those who think we have
     a sensible set of reqs and we can move forward? Those who need
     more reqs? [The hum level was 50-50, no consensus by humming on
     if we have the reqs captured right.]

     Dave Oran: Need hum on slightly different -- do we get involved
     in reqs that requires identity (being able to communicate why you
     are sending it to this particular party).

     Brian Rosen: That is a reasonable suggestion -- so considered. We
     will get the reqs out before Yokohama and bring the solution out
     before then. Those of you who hummed against it should
     participate in the list when we discuss this on it.

 

 

Flemming Andreasen, SIP Extensions for Media Authorization.
draft-ietf-sip-call-auth-04.txt :

     Changes: Category is informational -- Header is now
     P-Media-Authorization. Applicability statement about appropriate
     use (SIP Proxy and Policy Server (PDP) must belong to the same
     domain) Updated rules about when to add a P-Media-Authorization
     header. Additional security considerations -- don't encrypt
     message bodies (proxies need to examine them).

     Open issues: None known (authors list need to be trimmed),
     currently in WGLC.

     Jonathan: I will send you some minor reviews. More look-see
     needed in the security section. This token is about media
     authorization -- authorization follows authentication. DOes the
     I-D point out this issue?

     Flemming: You do not necessarily need to authenticate before
     authorization. Some entity has been given an authorization token
     to access some resource.

     Jonathan: More discussion maybe needed on the security section --
     you are giving a token to a party that you may not have
     authenticated. If that is your model, fine; a couple of sentences
     would probably suffice in the I-D.

     Brian Rosen: The WGLC is going to get over, if anyone wants to
     raise more issues please do so. It fills the needs, even though
     it has a lot of limits. LC be it, we will move forward.

 

 

Ben Campbell, SIP Extensions for IM :

     -05 draft; recent changes -- remove CPIM mapping to separate
     draft. Would like to include in 3rd bundle to IESG.

     Highlights - sends IM; does not initiate a dialog, does not
     discuss message sessions; actual message in bodies.

     Open issues: No recent discussions. Needs minor editorial changes
     (forking, threading -- couple of sentences). Anything else? Is it
     ready for LC? One more revision -- no change in substance, more
     editorial.

     Brian Rosen: Ok, as soon as you have the revision, we will post
     it as LC.

     
Closing Remarks :

     Brian Rosen: Administering this list is no fun -- people forward
     their email to accounts that consistently run over quota. The
     list is setup so that only subscribers can post.

21:29 CST - WG adjourned. 



SIP Session 2, 53rd IETF 

Start 13:06 CST 

      Added AKA Digest and Path discussion to the Agenda. 
      Agenda accepted. 

Digest based authentication -- James Undery, Ubiquity 

      Quick run through of improvements to Digest
      authentication. UAC->UAS auth, UAC->Proxy authentication
      supported in the SIP spec. Our draft adds Proxy->UAS auth.,
      bid-down protection, mutual auth., integrity. Added 3 new
      headers and 1 new response (492) for Proxy->UAS authentication
      Bid-down protection: prefix added to nonces, protects scheme and
      quality of detection.

      Open issues: 1) No algorithm protection - if a hashing algo is
      broken, we need algo revocation. Proposal: make limitation
      explicit and rule that algo revocation is out of scope. 2) No
      negotiation of body integrity protection -- Proxies can't alter
      message bodies; Proposal : leave unchanged 3) No protection
      against weak passwords: Proposal - make limitation explicit, the
      solution is out of scope. 4) Client side can't initiate
      authentication 5) Forking and response collation issues - can't
      guarantee upstream entities see the challenges; response
      collation oriented towards success. Proposal: make limitations
      explicit.

      Jon: What are you trying to accomplish as a UAS by
      authenticating your upstream proxy? James: You may trust the
      proxy but not the link between the proxy and the UAS (for
      example: a radio interface). Jon: So this is the integrity
      process, not an authentication one. James: Authentication and
      integrity are closely linked. Rohan: We need to think about if
      this is needed? Looks like it is solved by TLS anyway. We need
      to understand under what circumstances we will use this
      approach. Jonathan: With this you do not know which proxy -- one
      up or 2 up -- you want to authenticate.

      Comment: You mention weak passwords; there are no strong
      passwords. You are unlikely to create a password that is not
      uncrackable, regardless of them appearing in the dictionary or
      not. Relying on a long random key is of no help. Christian: I
      would like to reinforce this point. Digest should be combined
      with strong authentication of the server, not on its
      own. Henning: when people talk about passwords, I wouldn't think
      that this is human generated; it is random string generated by
      some automata.

      Dean: Are we going to close any of these today? If not, let's
      take this to the mailing list. Brian: This has been hanging on
      for a long time; are we going to extend digest or not fortify it
      anymore. I do not know how to go ahed. People are saying that
      digest is terrible, but others are saying that it is still
      useful. I do not see other alternatives on the floor. Christian:
      2 forms with digest: 1 is if you are sending it as cleartext. 2
      is when you are doing digest with a 3rd party you have not
      authenticated. Simple thing for us is to use digest only for
      REGISTER not anything else. Brian: But there is no other
      solution on the table. Allison: The security review for bis came
      out as digest being a very lightweight way to do user
      authentication is okay. There is a good possibility of taking
      some time over this and making it better; maybe we should say
      that this document is not a WG document. We should discuss for
      the charter document something that exploits S/MIME. Rohan:
      There is some stuff in James' draft we can get consensus
      quickly; others may take some time. Allison: There is a huge
      deployment of digest. MD5 digest is known weak, but is not going
      to be thrown out. The topic we have now is not extending digest,
      but supporting some other password (a la AKA). For this
      document, consider what requirements it is meeting? Henning: One
      thing desperately needed in digest is registrar
      authentication. If we do something with digest at all, it must
      be this. Authenticating previous hops is nice, but not a known
      vulnerability we need solve now. Steven (3GPP): We do have a
      basic need to protect last hop integrity. We need to know what
      the intentions of the WG are towards digest. We need the
      direction very soon otherwise we are in a bind. Allison: Maybe,
      as Steven said, IPSec could be used for the short duration --
      could be the right way of meeting the requirement. Maybe we can
      have 5 minutes on some other agenda to talk about this. Brian:
      We are not getting anywhere -- let's move on and take it to
      list.


Digest AKA Authentication - Aki Nieme 

      AKA is a shared secret based auth that uses a smart card like
      device. Previous proposal (...-eap-01.txt) got good reception at
      SLC.

      Digest AKA reuses the digest scheme and uses the AKA parameters
      as input to the digest mechanism. AKA generates "one-time"
      passwords for Digest.

      Issues: 1) "Choke point" attack - similar to the weak password
      attack. 2) Should we adopt draf-niemi-sipping-digest-aka-00? It
      provides message integrity and is complementary to vanilla
      digest used today.

      Future: Will this become a work item for SIP WG? RFC category?
      There is some time pressure since 3GPP R5 is coming up. Can
      draft-niemi-digest-aka-00.txt be adopted as a solution?

      Allison: This does not involve any SIP extensions; just the
      extension of HTTP methods. It is good to get the SIP people's
      knowledge. There is no need for it to be a WG document; you can
      take it to RFC as an individual submission. Brian: Is there
      sufficient interest in the group to make it a WG item, or
      continue as an individual submission? [Took hum; the hum for
      people who want to make it a WG document prevailed]. Miguel
      Garcia: why use a 3G specific technology which has no broad
      impact on the Internet? Adam seconded. [The chairs agreed that
      we may have to revisit this issue again]


Security Negotiation open issues, Jari Akko 

     Presented issues, asked what to do next. Brian: Anyone object to
     NOT go forward with this? [No one objected; this is part of SIP
     WG]

SIP Extensions for Network Asserted Caller Identity and Privacy within
trusted networks - Flemming Andreasen

     Good list discussion 1 month ago; currently in WG LC. There have
     been some offline comment that have not been incorporated in the
     I-D yet.

     Overview of changes: applicability statement (only suitable in
     the same admin domain, draft is for network-asserted identity,
     not user-asserted). Anonymity header got removed. Grammar fixes
     to be consistent with -09 bis.

     Open issues: 1) Proxy handling of RP-ID received from untrusted
     entity (proxy or UA) 2 options: 1) if verifiable, set screen=yes
     2) always remove untrusted RP-ID Option 1 seems more general then
     2; Recommendation: Option 1.

     This generated a lot of discussion, mostly revising around
     policies vs. protocols, network asserted identities vs. user
     asserted identities, (in)security of this I-D and why it will not
     be acceptable to IESG, and this being more for the benefit of
     3GPP.

     Randy Bush discussed current unacceptability of draft to
     operations directorate, and agreed to Send Text.


REFER Open Issues -- Robert Sparks

      There was extended discussion about transitive security of the
      referral token (Referred-By) and its impact on the delivery
      schedule. Three-party problems are considered to be among the
      hardest of security problems to solve (Eric Rescorla). The
      authors recommend separating the problems, and doing REFER
      without Referred-By as a near term deliverable then tackle
      Referred-By as a separate task. The working group seems very
      interested in solving this problem and there was no clear
      consensus on separating it. However, it appears unlikely that
      this will be solved in the required May 30 timeframe, so we may
      need to administratively divide the problem space.

      Several proposals to resolve the transitive security requirement
      were discussed, with consensus seeming to form around an S/MIME
      approach.



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From daemon@optimus.ietf.org  Sat Apr 13 17:11:04 2002
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Date: Sat, 13 Apr 2002 13:39:07 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: ietf@ietf.org, <sip@ietf.org>, <sipping@ietf.org>
cc: rohan@cisco.com, <dwillis@dynamicsoft.com>, <brian.rosen@marconi.com>,
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Subject: [Sip] SIP/SIPPING Interim Meeting
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Hi,

I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in Las
Vegas, Nevada.  This date and location were changed (with AD approval)
as a fallback due to conflicts with the previously proposed time and place
(Boston, the following week).

The meeting will occur at the Las Vegas Convention Center or the adjacent
Las Vegas Hilton Hotel (not to be confused with the Flamingo Hilton).
Inexpensive rooms and flights are still available.

Proposed Agenda:

Monday May 6
08:00  Agenda bashing, logistics
08:30  Open issues for REFER, Replaces, and cc-transfer
11:00  Multiparty and conferencing requirements
12:30  lunch
14:00  call-info and conference-info packages
16:00  Content-indirection
16:30  Binding published data to SIP
17:30  Wrap

Tuesday May 7
08:00   The long-term privacy solution
12:00   lunch
13:30   outstanding security issues/requirements
15:00   AAA requirements
17:30   Wrap

As always, drafts must be submitted by two weeks before the meeting,
(April 22).  Detailed minutes and blue sheets will be taken and posted to
the list.  All decisions based on a proto-consensus at the interim will go
back to the list.

If you are planning to attend, please send mail to me (rohan@cisco.com)
privately so I can get a room of an appropriate size.

thanks,
-rohan

p.s. A couple folks have asked for my motivation for suggesting an interim.
Basically we've got a lot of stuff on our plate in both WGs, and much of
it is in that midway done phase were face time is most valuable.
An interim would give us time to resolve several sticky technical issues
and then get drafts out before the cutoff date for IETF54.  I think
we could have a much more productive meeting in Japan as a result.


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In a message dated 4/13/02 4:43:51 PM Eastern Daylight Time, rohan@cisco.com 
writes:


> I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in Las
> Vegas, Nevada.  This date and location were changed (with AD approval)
> as a fallback due to conflicts with the previously proposed time and place
> (Boston, the following week).
> 
> 
What happened to the 30 day advance notice (as your 4 April e-mail said was 
required)? I'm sure that other besides myself have already planned other 
things for May 6-7, since no interim meeting was announced on time.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/13/02 4:43:51 PM Eastern Daylight Time, rohan@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in Las
<BR>Vegas, Nevada. &nbsp;This date and location were changed (with AD approval)
<BR>as a fallback due to conflicts with the previously proposed time and place
<BR>(Boston, the following week).
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">What happened to the 30 day advance notice (as your 4 April e-mail said was required)? I'm sure that other besides myself have already planned other things for May 6-7, since no interim meeting was announced on time.
<BR>
<BR>Mike
<BR></FONT></HTML>

--part1_c.2669c2fb.29eb47a2_boundary--

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From daemon@optimus.ietf.org  Sun Apr 14 23:51:43 2002
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Thread-Topic: [Sipping] SIP/SIPPING Interim Meeting
Thread-Index: AcHj+AqSZns5TevyR+ufe3TBYtVP5AANCXfg
From: "Zmolek, Andrew (Andrew)" <zmolek@avaya.com>
To: <Mpierce1@aol.com>, <rohan@cisco.com>, <ietf@ietf.org>, <sip@ietf.org>,
        <sipping@ietf.org>
Cc: <dwillis@dynamicsoft.com>, <brian.rosen@marconi.com>, <jo@ipdialog.com>,
        <mankin@isi.edu>, <sob@harvard.edu>
Subject: [Sip] RE: [Sipping] SIP/SIPPING Interim Meeting
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This date also conflicts with upcoming W3C meetings. What was the =
conflict with the following week?
=20
--Andy

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Sunday, April 14, 2002 2:59 PM
To: rohan@cisco.com; ietf@ietf.org; sip@ietf.org; sipping@ietf.org
Cc: dwillis@dynamicsoft.com; brian.rosen@marconi.com; jo@ipdialog.com; =
mankin@isi.edu; sob@harvard.edu
Subject: Re: [Sipping] SIP/SIPPING Interim Meeting


In a message dated 4/13/02 4:43:51 PM Eastern Daylight Time, =
rohan@cisco.com writes:=20




I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in Las=20
Vegas, Nevada.  This date and location were changed (with AD approval)=20
as a fallback due to conflicts with the previously proposed time and =
place=20
(Boston, the following week).=20




What happened to the 30 day advance notice (as your 4 April e-mail said =
was required)? I'm sure that other besides myself have already planned =
other things for May 6-7, since no interim meeting was announced on =
time.=20

Mike=20



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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D555181803-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>This=20
date also conflicts with upcoming W3C meetings. What was the conflict =
with the=20
following week?</FONT></SPAN></DIV>
<DIV><SPAN class=3D555181803-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D555181803-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2>--Andy</FONT></SPAN></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com=20
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Sunday, April 14, 2002 2:59=20
  PM<BR><B>To:</B> rohan@cisco.com; ietf@ietf.org; sip@ietf.org;=20
  sipping@ietf.org<BR><B>Cc:</B> dwillis@dynamicsoft.com;=20
  brian.rosen@marconi.com; jo@ipdialog.com; mankin@isi.edu;=20
  sob@harvard.edu<BR><B>Subject:</B> Re: [Sipping] SIP/SIPPING Interim=20
  Meeting<BR><BR></FONT></DIV><FONT face=3Darial,helvetica><FONT =
size=3D2>In a=20
  message dated 4/13/02 4:43:51 PM Eastern Daylight Time, =
rohan@cisco.com=20
  writes: <BR><BR><BR>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px"=20
  TYPE=3D"CITE">I'd like to announce a joint SIP/SIPPING interim meeting =
May 6-7=20
    in Las <BR>Vegas, Nevada. &nbsp;This date and location were changed =
(with AD=20
    approval) <BR>as a fallback due to conflicts with the previously =
proposed=20
    time and place <BR>(Boston, the following week). =
<BR><BR></FONT><FONT lang=3D0=20
    face=3DArial color=3D#000000 size=3D3=20
  FAMILY=3D"SANSSERIF"></BLOCKQUOTE><BR></FONT><FONT lang=3D0 =
face=3DArial=20
  color=3D#000000 size=3D2 FAMILY=3D"SANSSERIF">What happened to the 30 =
day advance=20
  notice (as your 4 April e-mail said was required)? I'm sure that other =
besides=20
  myself have already planned other things for May 6-7, since no interim =
meeting=20
  was announced on time. <BR><BR>Mike=20
<BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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From daemon@ns.ietf.org  Mon Apr 15 06:05:36 2002
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From: "Salva Rey Calatayud" <salreyca@teleco.upv.es>
To: sip@ietf.org
Date: Sat, 15 Apr 2000 11:54:06 +0200
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Subject: Re: [Sip] O/A model
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Isn't there a mechanism whereby both UAs get back to an agreed 
state? How can the negotiation process continue properly 
otherwise?

regards,
Salva

On 10 Apr 2002, at 0:43, Jonathan Rosenberg wrote:

> THe spec does not detail error handling for the infinite different ways
> in which they might occur. The choice is at the discretion of the
> implementation.
> 
> -Jonathan R.
> 
> Salva Rey Calatayud wrote:
> > 
> > Hi,
> > 
> >         how should a UA act when let's say the other endpoint doesn't
> > behave compliantly to the O/A exchange model. For instance, we
> > send an offer, and none of the associated messages that could
> > contain an answer does (an UPDATE with a 200OK without
> > answer, INVITE with offer and none of the provisional response nor
> > final 200OK contain an answer )
> > 
> > thanks,
> > Salva
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com




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From daemon@ns.ietf.org  Mon Apr 15 06:43:20 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, Mpierce1@aol.com,
        Brian.Rosen@marconi.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 11:18:40 +0100
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Dean wrote:
 
> 
> Excellent summary, Mike. That's exactly what I'm saying.
> 
> In short:
> 
> To:/From: are equivalent to ISDN user-to-user data, as is most of
> current SIP.

But not in the case raised by Juha, where the Service Provider checks the
contents of the From field. Then these fields are more like the
user-provided Calling Party Number field which is checked by the local
exchange.

Even if the Service Provider does not do this checking, I would still argue
that there is an implicit expectation that these fields contain indentity
information, and indeed this is what most clients will put in there. I don't
think this is insignificant. It is completely different from User-to-user in
ISDN, or, say, Subject: in SIP where there is no expectation at all.

In ISDN, if you have a so-called 'Special Arrangement' then you can supply
an identity which is passed transparently through the network - no checking,
nothing. Obviously, the user only supplies an identity if they do not want
anonymity. But even if they do supply an identity, if the *subscriber* has
requested anonymity, then it is removed by the network.

Sounds pretty analogous to From: to me.

> 
> RPID: is analogous to CPN, and its "Screened" parameter analogous to
> what happens in ISDN today.
> 

Analogous to Calling Party Number within the network, and when delivered to
the Called party.

> Privacy: is analogous to the calling line identification restriction
> flag.
>  
> I think an understanding of this relationship should clear up 
> 99% of the
> debate over the "privacy" draft

Well, it would if we agreed :-)

...Mark

> , which is really much more of a SIP-T
> "telephony interworking" document than it is a general 
> privacy document.
> 

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<P><FONT SIZE=3D2>Dean wrote:</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Excellent summary, Mike. That's exactly what =
I'm saying.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In short:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; To:/From: are equivalent to ISDN user-to-user =
data, as is most of</FONT>
<BR><FONT SIZE=3D2>&gt; current SIP.</FONT>
</P>

<P><FONT SIZE=3D2>But not in the case raised by Juha, where the Service =
Provider checks the contents of the From field. Then these fields are =
more like the user-provided Calling Party Number field which is checked =
by the local exchange.</FONT></P>

<P><FONT SIZE=3D2>Even if the Service Provider does not do this =
checking, I would still argue that there is an implicit expectation =
that these fields contain indentity information, and indeed this is =
what most clients will put in there. I don't think this is =
insignificant. It is completely different from User-to-user in ISDN, =
or, say, Subject: in SIP where there is no expectation at =
all.</FONT></P>

<P><FONT SIZE=3D2>In ISDN, if you have a so-called 'Special =
Arrangement' then you can supply an identity which is passed =
transparently through the network - no checking, nothing. Obviously, =
the user only supplies an identity if they do not want anonymity. But =
even if they do supply an identity, if the *subscriber* has requested =
anonymity, then it is removed by the network.</FONT></P>

<P><FONT SIZE=3D2>Sounds pretty analogous to From: to me.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; RPID: is analogous to CPN, and its =
&quot;Screened&quot; parameter analogous to</FONT>
<BR><FONT SIZE=3D2>&gt; what happens in ISDN today.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>Analogous to Calling Party Number within the network, =
and when delivered to the Called party.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Privacy: is analogous to the calling line =
identification restriction</FONT>
<BR><FONT SIZE=3D2>&gt; flag.</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; I think an understanding of this relationship =
should clear up </FONT>
<BR><FONT SIZE=3D2>&gt; 99% of the</FONT>
<BR><FONT SIZE=3D2>&gt; debate over the &quot;privacy&quot; =
draft</FONT>
</P>

<P><FONT SIZE=3D2>Well, it would if we agreed :-)</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; , which is really much more of a SIP-T</FONT>
<BR><FONT SIZE=3D2>&gt; &quot;telephony interworking&quot; document =
than it is a general </FONT>
<BR><FONT SIZE=3D2>&gt; privacy document.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@ns.ietf.org  Mon Apr 15 06:48:01 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Mpierce1@aol.com'"
	 <Mpierce1@aol.com>, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 11:32:19 +0100
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Brain wrote:


The problem with this line of thinking is that in the PSTN, identity is
always asserted by
the network, and never by the UA.  In SIP, presently, identity is always
asserted by
the UA and never the network.  Privacy when you have network asserted
identity
means something different than when you have user asserted identity.

If you include in ' Network asserted identity' information which the Service
Provider *forces* the user to place in the messaging, then I agree. It is
about whether the service requires the information to be there, not which
device physically inserts it.  

 
There are no regulations, and I don't see any reason to believe that
regulators
will slavishly follow the current practice.  Until you have evidence to the
contrary,
arguing that the same SOLUTIONS are required is not helpful. 
 

There is Data Protection. We are not arguing about what regulations
will/won't apply, but how existing Data Protection will be interpreted in
the context of SIP services. We do have legislative guidance for services
which are 'Publically Available Telecommunications Services'. Perhaps there
is a case to argue that SIP services are so different from all other
Publically Available Telecommunications Services that Data Protection should
not apply in the same way - I doubt it though. The service as seen by the
user is not so different, but with multimedia & more flexibility. The
technology we use to provide the service is irrelevant to Data Protection.
 
My point is that it is at the *very least* 'likely' that Data Protection
will apply in the way I've described. A 'head in the sand' approach to this
is what is not helpful. We need to understand what the solutions are, or
would be, so that we can be confident we are not going to be floored by this
when it comes to deploying large-scale commercial services.
 
I'm quite happy with Jonathan's answer that you can do this with a B2BUA and
no standardisation is required. Others are arguing that the network should
NEVER modify the From/To fields, in which case I say that standardisation
*is* required for a client-based solution.
 
If we can agree that we don't need to do anything here - or that a client
based solution will be part of the more general privacy work - then we could
move on.
 
...Mark

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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002>Brain wrote:</SPAN></FONT><FONT face=Tahoma 
size=2><BR></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"></FONT>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>The problem 
  with this line of thinking is that in the PSTN, identity is always asserted 
  by</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>the 
  network, and never by the UA.&nbsp; In SIP, presently, identity is always 
  asserted by</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>the UA and 
  never the network.&nbsp; Privacy when you have network asserted 
  identity</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>means 
  something different than when you have user asserted 
  identity.</FONT></SPAN></DIV></BLOCKQUOTE>
<DIV><SPAN class=480061415-12042002><FONT face=Arial></FONT></SPAN><FONT 
size=1><FONT color=#0000ff><FONT face=Verdana><SPAN class=230572210-15042002>If 
you include in '</SPAN><SPAN class=230572210-15042002>&nbsp;Network asserted 
identity' information which the Service Provider *forces* the user to place in 
the messaging, then I agree. It is about whether the service requires the 
information to be there, not which device physically inserts 
it.&nbsp;</SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV>&nbsp;</DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>There are 
  no regulations, and&nbsp;I don't see any reason to believe that 
  regulators</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>will 
  slavishly follow the current practice.&nbsp; Until you have evidence to the 
  contrary,</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff face=Arial>arguing 
  that the same SOLUTIONS are required is not helpful.&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=480061415-12042002><FONT color=#0000ff 
  face=Arial></FONT></SPAN>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002>There is Data Protection. We are not arguing about what 
regulations will/won't apply, but how existing Data Protection will be 
interpreted in the context of SIP services. We do have legislative guidance for 
services which are 'Publically Available Telecommunications Services'. Perhaps 
there is a case to argue that SIP services are so different from all other 
Publically Available Telecommunications Services that&nbsp;Data Protection 
should not apply in the same way - I doubt it though. The service as seen by the 
user is not so different, but with multimedia &amp; more flexibility. The 
technology we use to provide the service is irrelevant to Data 
Protection.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=230572210-15042002>My 
point is that it is at the *very least* 'likely' that Data Protection will apply 
in the way I've described. A 'head in the sand' approach to this is what is not 
helpful. We need to understand what the solutions are, or would be, so that we 
can be confident we are not going to be floored by this when it comes to 
deploying large-scale commercial services.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=230572210-15042002>I'm 
quite happy with Jonathan's answer that you can do this with a B2BUA and no 
standardisation is required. Others are arguing that the network should NEVER 
modify the From/To fields, in which case I say that standardisation *is* 
required for a client-based solution.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=230572210-15042002>If 
we can agree that we don't need to do anything here - or that a client based 
solution will be part of the more general privacy work - then we could move 
on.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=230572210-15042002>...Mark</SPAN></FONT></DIV></BODY></HTML>

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From daemon@ns.ietf.org  Mon Apr 15 06:52:49 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>,
        Dean Willis
	 <dean.willis@softarmor.com>
Cc: Mpierce1@aol.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 11:11:05 +0100
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Juha wrote:

> if someone subscribes to my sip service, i DO check for each message
> received from the user that the from field corresponds to what i have
> agreed with the user.  otherwise the user could fake to be somone else
> that i consider a bad thing.  if the from field doesn't match, i don't
> forward the message.  if that is not acceptable for the user, the use
> can choose another service provider or roll its own service.  
> ii do not
> modify anything.
> 

I think many service providers will wish to do the same, and indeed this
checking is required in 3GPP. The situation is much clearer in this case,
because in order to access the service at all, the user MUST insert their
true identity (or 'anonymous') into the From: field.

So, although it is the user's equipment which *physically* puts in the
information, it is the service which is responsible for this information
being there. The Service Provider clearly has a responibility to remove that
information (or reject the session) if the subscriber has requested that it
not be present.

Rejecting the session and requesting the UA to re-try with 'anonymous' would
be great, but it doesn't work with RFC2543 clients and requires a double
attempt for session establishment, which is not good for wireless.

So, in this case I think the Service Provider *will* need to have the
capability to remove this information. They will need a B2BUA as described
in previous mails. No standardisation is required.

...Mark

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<P><FONT SIZE=3D2>Juha wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; if someone subscribes to my sip service, i DO =
check for each message</FONT>
<BR><FONT SIZE=3D2>&gt; received from the user that the from field =
corresponds to what i have</FONT>
<BR><FONT SIZE=3D2>&gt; agreed with the user.&nbsp; otherwise the user =
could fake to be somone else</FONT>
<BR><FONT SIZE=3D2>&gt; that i consider a bad thing.&nbsp; if the from =
field doesn't match, i don't</FONT>
<BR><FONT SIZE=3D2>&gt; forward the message.&nbsp; if that is not =
acceptable for the user, the use</FONT>
<BR><FONT SIZE=3D2>&gt; can choose another service provider or roll its =
own service.&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; ii do not</FONT>
<BR><FONT SIZE=3D2>&gt; modify anything.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I think many service providers will wish to do the =
same, and indeed this checking is required in 3GPP. The situation is =
much clearer in this case, because in order to access the service at =
all, the user MUST insert their true identity (or 'anonymous') into the =
From: field.</FONT></P>

<P><FONT SIZE=3D2>So, although it is the user's equipment which =
*physically* puts in the information, it is the service which is =
responsible for this information being there. The Service Provider =
clearly has a responibility to remove that information (or reject the =
session) if the subscriber has requested that it not be =
present.</FONT></P>

<P><FONT SIZE=3D2>Rejecting the session and requesting the UA to re-try =
with 'anonymous' would be great, but it doesn't work with RFC2543 =
clients and requires a double attempt for session establishment, which =
is not good for wireless.</FONT></P>

<P><FONT SIZE=3D2>So, in this case I think the Service Provider *will* =
need to have the capability to remove this information. They will need =
a B2BUA as described in previous mails. No standardisation is =
required.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, hgs@cs.columbia.edu,
        Brian.Rosen@marconi.com
Cc: jdrosen@dynamicsoft.com, sip@ietf.org, bcampbell@dynamicsoft.com,
        fandreas@cisco.com, jon.peterson@neustar.biz, wtm@research.att.com
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 11:37:14 +0100
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Mike,
 
When I said 'future work', the future could be very soon. I said it on
Friday, so it could even be today :-) We have a charter item for a more
general privacy mechanism and this could be included within that.
 
Regarding 'subscriber' this is a key point about the Service Provider model.
Whenever you have a Service Provider, then there is a 'subscriber' who is
the person (legal or natural) who has contracted with the Service Provider
for the provision of the service. (I am leaving out the case where the
Service Provider offers service to just anyone for free, so there is no
contract as such).
 
At least in Europe, if the service is a 'Publically available
telecommunications service' (note that's 'telecommunications' not
'telephony'), then the subscriber has a right under Data Protection for all
calls made via their subscription to be anonymous if they so choose.
 
The most common example is where the subscriber is a company and the users
are its employees.
 
Another example of a person who is not represented by a 'User Agent' in SIP
terms during a session invokation is someone who has set their Home Proxy to
forward calls (using CPL or whatever). When a call is made, the
forwarding-party is represented only by their Proxy - their UA may not even
be connected to the network. Again, this person has a right for their
Personal Data to not be revealed by the Service Provider, so if the Service
Provider is responsible for the insertion of any such data in the messaging,
they need to capability to remove it again.
 
Regards...Mark

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: 12 April 2002 19:27
To: Watson, Mark [MDN05:EP10:EXCH]; hgs@cs.columbia.edu;
Brian.Rosen@marconi.com
Cc: jdrosen@dynamicsoft.com; sip@ietf.org; bcampbell@dynamicsoft.com;
fandreas@cisco.com; jon.peterson@neustar.biz; wtm@research.att.com
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft


In a message dated 4/12/02 11:42:49 AM Eastern Daylight Time,
mwatson@nortelnetworks.com writes: 




Mike, 

My suggestion of a generic mechanism to carry Personal Data was targetting
at future work, not the present privacy draft.



Yes, I suspected that, but then I noticed that Tom picked up on it as an
"very interesting idea " and seemed to be suggesting that we run with it. 




Two points: 
1) It's not just about privacy for the calling user - that's relatively
easy, you just need a mechanism to 'privately' pass information to trusted
network entities and apart from that it's up to the UA what it does/doesn't
reveal. The big problem is when other people are involved who do not have an
'agent' representing them. A subscriber to a public service is an example.
These people have rights which only the Service Provider knows about. The
Service Provider cannot be responsible for sending Personal Data of the
subscriber to the called party. Unfortunately the identity of the calling
user can often include personal data of the subscriber. 

2) You said 'If the user voluntarily includes it, it's not private'. Sure,
it's not private *as far as the user is concerned*. But it might be private
with respect to other people. So the question is who was *responsible* for
sending it. If it's entirely the user then no problem - same as if it's the
first thing the user says after Answer. The problem is that To/From are a
grey area, since the SIP spec clearly intends that they contain the
calling/called identities (as would a spec for
'What-I-Want-You-To-Call-Me'). A network based subscriber privacy service
which did not modify the From field would be pretty rubbish, whereas one
which did not modify the Subject would probably be fine. 




I'm confused by what you are getting at by "subscribers" and "other people"
(who don't have an "agent"), and "calling user". There is a SIP phone
(containing a "User Agent") and it might be used by multiple people to place
calls. If people don't have their own identity, then every call must use the
identity of the physical device. If there is a capability for each person to
provide a separate identity, then that is what is used by the "network" as
the basis for a "subscription" to service and to apply privacy. 

If you are referring to the complicated cases of how to apply "privacy" to
identities involved in 3-way services such as call transfer and call
forwarding, then I understand the difficulties, having been through
excrutiating discussions in standards development for ISDN. It isn't easy or
pretty. But these issues will only be worked out in a specific discusion of
the interworking of the identity/privacy service with the call transfer/call
forwarding services, something that the IETF is intent on not doing. 

I still say if the user voluntarily includes something beyond the specific
identity fields, then the network has no obligation to apply privacy to it.
I don't see a case in which the network has to do anything else to provide
privacy for "other people". Again, maybe there is a difference in what
"user" refers to in your point 2) above. If the "standard" service operation
(in the User Agent or network) puts in some identity, or passes along one it
got during Call Transfer, then of course privacy rules must apply to it.
That's not the user (person) voluntarily putting in the information. 

Mike 


------_=_NextPart_001_01C1E469.81B1F20A
Content-Type: text/html

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">


<META content="MSHTML 5.00.3315.2870" name=GENERATOR></HEAD>
<BODY>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002>Mike,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=494145109-15042002>When 
I said 'future work', the future could be very soon. I said it on Friday, so it 
could even be today :-) We have a charter item for a more general privacy 
mechanism and this could be included within that.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002>Regarding 'subscriber' this is a key point about the 
Service Provider model. Whenever you have a Service Provider, then there is a 
'subscriber' who is the person (legal or natural) who has contracted with the 
Service Provider for the provision of the service. (I am leaving out the case 
where the Service Provider offers service to just anyone for free, so there is 
no contract as such).</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=494145109-15042002>At 
least in Europe, if the service is a 'Publically available telecommunications 
service' (note that's 'telecommunications' not 'telephony'), then the subscriber 
has a right under Data Protection for all calls made via their subscription to 
be anonymous if they so choose.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=494145109-15042002>The 
most common example is where the subscriber is a company and the users are its 
employees.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002>Another example of a person who is not represented by a 
'User Agent' in SIP terms during a session invokation is someone who has set 
their Home Proxy to forward calls (using CPL or whatever). When a call is made, 
the forwarding-party is represented only by their Proxy - their UA may not even 
be connected to the network. Again, this person has a right for their Personal 
Data to not be revealed by the Service Provider, so if the Service Provider is 
responsible for the insertion of any such data in the messaging, they need to 
capability to remove it again.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=494145109-15042002>Regards...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> 12 April 2002 19:27<BR><B>To:</B> 
  Watson, Mark [MDN05:EP10:EXCH]; hgs@cs.columbia.edu; 
  Brian.Rosen@marconi.com<BR><B>Cc:</B> jdrosen@dynamicsoft.com; sip@ietf.org; 
  bcampbell@dynamicsoft.com; fandreas@cisco.com; jon.peterson@neustar.biz; 
  wtm@research.att.com<BR><B>Subject:</B> Re: Summary of RE: [Sip] Comment, SIP 
  Privacy draft<BR><BR></DIV></FONT><FONT face=arial,helvetica><FONT face=Arial 
  lang=0 size=2 FAMILY="SANSSERIF">In a message dated 4/12/02 11:42:49 AM 
  Eastern Daylight Time, mwatson@nortelnetworks.com writes: 
  <BR><BR><BR></FONT><FONT color=#0000ff face=Verdana lang=0 size=1 
  FAMILY="SANSSERIF">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px" 
  TYPE="CITE">Mike,</FONT><FONT color=#000000 face=Verdana lang=0 size=2 
    FAMILY="SANSSERIF"> <BR></FONT><FONT color=#000000 face=Verdana lang=0 
    size=3 FAMILY="SANSSERIF"><BR></FONT><FONT color=#0000ff face=Verdana lang=0 
    size=1 FAMILY="SANSSERIF">My suggestion of a generic mechanism to carry 
    Personal Data was targetting at future work, not the present privacy 
    draft.</FONT><FONT color=#0000ff face=Verdana lang=0 size=3 
    FAMILY="SANSSERIF"></BLOCKQUOTE><BR></FONT><FONT color=#000000 face=Verdana 
  lang=0 size=3 FAMILY="SANSSERIF"><BR></FONT><FONT color=#000000 face=Arial 
  lang=0 size=2 FAMILY="SANSSERIF">Yes, I suspected that, but then I noticed 
  that Tom picked up on it as an "very interesting idea " and seemed to be 
  suggesting that we run with it.</FONT><FONT color=#000000 face=Verdana lang=0 
  size=3 FAMILY="SANSSERIF"> <BR></FONT><FONT color=#000000 face=Verdana lang=0 
  size=3 FAMILY="SANSSERIF"><BR></FONT><FONT color=#000000 face=Verdana lang=0 
  size=3 FAMILY="SANSSERIF">
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px" 
  TYPE="CITE"></FONT><FONT color=#000000 face=Arial lang=0 size=2 
    FAMILY="SANSSERIF"><BR></FONT><FONT color=#0000ff face=Arial lang=0 size=2 
    FAMILY="SANSSERIF">Two points:</FONT><FONT color=#000000 face=Verdana lang=0 
    size=2 FAMILY="SANSSERIF"> <BR></FONT><FONT color=#0000ff face=Verdana 
    lang=0 size=1 FAMILY="SANSSERIF">1) It's not just about privacy for the 
    calling user - that's relatively easy, you just need a mechanism to 
    'privately' pass information to trusted network entities and apart from that 
    it's up to the UA what it does/doesn't reveal. The big problem is when other 
    people are involved who do not have an 'agent' representing them. A 
    subscriber to a public service is an example. These people have rights which 
    only the Service Provider knows about. The Service Provider cannot be 
    responsible for sending Personal Data of the subscriber to the called party. 
    Unfortunately the identity of the calling user can often include personal 
    data of the subscriber.</FONT><FONT color=#000000 face=Verdana lang=0 size=3 
    FAMILY="SANSSERIF"> <BR><BR></FONT><FONT color=#0000ff face=Verdana lang=0 
    size=1 FAMILY="SANSSERIF">2) You said 'If the user voluntarily includes it, 
    it's not private'. Sure, it's not private *as far as the user is concerned*. 
    But it might be private with respect to other people. So the question is who 
    was *responsible* for sending it. If it's entirely the user then no problem 
    - same as if it's the first thing the user says after Answer. The problem is 
    that To/From are a grey area, since the SIP spec clearly intends that they 
    contain the calling/called identities (as would a spec for 
    'What-I-Want-You-To-Call-Me'). A network based subscriber privacy service 
    which did not modify the From field would be pretty rubbish, whereas one 
    which did not modify the Subject would probably be fine.</FONT><FONT 
    color=#000000 face=Verdana lang=0 size=3 FAMILY="SANSSERIF"> 
  <BR></BLOCKQUOTE><BR></FONT><FONT color=#000000 face=Arial lang=0 size=2 
  FAMILY="SANSSERIF"><BR>I'm confused by what you are getting at by 
  "subscribers" and "other people" (who don't have an "agent"), and "calling 
  user". There is a SIP phone (containing a "User Agent") and it might be used 
  by multiple people to place calls. If people don't have their own identity, 
  then every call must use the identity of the physical device. If there is a 
  capability for each person to provide a separate identity, then that is what 
  is used by the "network" as the basis for a "subscription" to service and to 
  apply privacy. <BR><BR>If you are referring to the complicated cases of how to 
  apply "privacy" to identities involved in 3-way services such as call transfer 
  and call forwarding, then I understand the difficulties, having been through 
  excrutiating discussions in standards development for ISDN. It isn't easy or 
  pretty. But these issues will only be worked out in a specific discusion of 
  the interworking of the identity/privacy service with the call transfer/call 
  forwarding services, something that the IETF is intent on not doing. <BR><BR>I 
  still say if the user voluntarily includes something beyond the specific 
  identity fields, then the network has no obligation to apply privacy to it. I 
  don't see a case in which the network has to do anything else to provide 
  privacy for "other people". Again, maybe there is a difference in what "user" 
  refers to in your point 2) above. If the "standard" service operation (in the 
  User Agent or network) puts in some identity, or passes along one it got 
  during Call Transfer, then of course privacy rules must apply to it. That's 
  not the user (person) voluntarily putting in the information. 
  <BR><BR>Mike</FONT> </FONT></BLOCKQUOTE></BODY></HTML>

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From: "Salva Rey Calatayud" <salreyca@teleco.upv.es>
To: sip@ietf.org
Date: Sat, 15 Apr 2000 15:00:13 +0200
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Isn't there a mechanism whereby both UAs get back to an agreed 
state? How can the negotiation process continue properly 
otherwise?

regards,
Salva

On 10 Apr 2002, at 0:43, Jonathan Rosenberg wrote:

> THe spec does not detail error handling for the infinite different 
ways
> in which they might occur. The choice is at the discretion of the
> implementation.
> 
> -Jonathan R.
> 
> Salva Rey Calatayud wrote:
> > 
> > Hi,
> > 
> >         how should a UA act when let's say the other endpoint 
doesn't
> > behave compliantly to the O/A exchange model. For instance, 
we
> > send an offer, and none of the associated messages that could
> > contain an answer does (an UPDATE with a 200OK without
> > answer, INVITE with offer and none of the provisional response 
nor
> > final 200OK contain an answer )
> > 
> > thanks,
> > Salva



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From: "Jesske, R" <R.Jesske@telekom.de>
To: mwatson@nortelnetworks.com, dean.willis@softarmor.com, Mpierce1@aol.com,
        Brian.Rosen@marconi.com, sip@ietf.org
Subject: AW: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 15:16:33 +0200
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Hi all,
regarding the discussion what SIP Header fits to what ISUP Parameter we should know what rules apply to the interworking from SIP to ISUP. From my understanding (of the last ITU discussion in Geneva) the following rules shall apply  to the mapping of the From and Remote Party ID to ISUP parameters:
....
Note: From: is mandatory and always checked for presence.
Case-a: If RPID is absent, then do the following:
1) Additional CgPN is derived from From: (Note Jesske:because it is user provided but not verified by the service provider)
2) CgPN is assigned with Network Generated Number. (Note Jesske: with a screening indication: network provided)
 
Case-b: If RPID is present and RPI-Screen="Yes", then do the following:
1) Additional CgPN is derived from From: (Note Jesske:because it is user provided but not verified by the service provider)
2) CgPN is derived from RPID: (Note Jesske: with a screening indication: user provided verified and passed )
 
Case-c: If RPID is present and RPI-Screen="No", then do the following:
1) Additional CgPN is derived from RPID: (Note Jesske:because it is user provided but not verified by the service provider)
2) CgPN is assigned with Network Generated Number. (Note Jesske: with a screening indication: network provided)
....
The interest of big Service Provider/Carriers is to have an proper inteworking of SIP and ISUP, because the will have in future hybrid networks with SIP and ISUP (BICC) signalling. And the above mentioned rules are the first step forward for a proper interworking (my opinion) between SIP and ISUP including the CLIP/CLIR Service.

I hope this information fits in your discussion. 

Best Regards 

Roland

 

	

-----Ursprüngliche Nachricht-----
Von: Mark Watson [mailto:mwatson@nortelnetworks.com]
Gesendet am: Montag, 15. April 2002 12:19
An: 'Dean Willis'; Mpierce1@aol.com; Brian.Rosen@marconi.com; sip@ietf.org
Betreff: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Dean wrote: 
  
> 
> Excellent summary, Mike. That's exactly what I'm saying. 
> 
> In short: 
> 
> To:/From: are equivalent to ISDN user-to-user data, as is most of 
> current SIP. 

But not in the case raised by Juha, where the Service Provider checks the contents of the From field. Then these fields are more like the user-provided Calling Party Number field which is checked by the local exchange.

Even if the Service Provider does not do this checking, I would still argue that there is an implicit expectation that these fields contain indentity information, and indeed this is what most clients will put in there. I don't think this is insignificant. It is completely different from User-to-user in ISDN, or, say, Subject: in SIP where there is no expectation at all.

In ISDN, if you have a so-called 'Special Arrangement' then you can supply an identity which is passed transparently through the network - no checking, nothing. Obviously, the user only supplies an identity if they do not want anonymity. But even if they do supply an identity, if the *subscriber* has requested anonymity, then it is removed by the network.

Sounds pretty analogous to From: to me. 

> 
> RPID: is analogous to CPN, and its "Screened" parameter analogous to 
> what happens in ISDN today. 
> 

Analogous to Calling Party Number within the network, and when delivered to the Called party. 

> Privacy: is analogous to the calling line identification restriction 
> flag. 
>  
> I think an understanding of this relationship should clear up 
> 99% of the 
> debate over the "privacy" draft 

Well, it would if we agreed :-) 

...Mark 

> , which is really much more of a SIP-T 
> "telephony interworking" document than it is a general 
> privacy document. 
> 


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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 09:27:39 -0500
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Thread-Topic: Summary of RE: [Sip] Comment, SIP Privacy draft
Thread-Index: AcHiQ4Sax+IzsyCjTM+M6z9mq960yACQbaQQ
From: "Chiou, Mark" <MChiou@Santera.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
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Dean wrote:
#####################################
The point I'm trying to make is that To and From are USER supplied =
information. The Service Provider doesn't insert them, inspect them, or =
have any responsibility for them. Insisting that the service provider =
change them is just like insisting that the service provider change any =
other user-supplied information, such as the words I'm typing now. =
Anything I put in my To or from field is deliberately and explicitly =
intended to be delivered to the terminating side of the dialog, and if =
as a service provider you FAIL to do that, then you have DAMAGED my =
data, and my lawyers will be in touch!
=20
The calling-party-identity presentation infromation (CLIP or RPID) is =
NOT inserted by the user. It is inserted by the network, WITHOUT THE =
CONSENT of the user. As such, the operator is responsible for the =
protection of that information in a manner consistent with regulation =
and the expressed wishes of the user.
<snip>
 ##########################################################
No, The CLIP is inserted by the network WITH THE CONSENT of the user. =
When you subscribe a line from a PSTN service provider, the form you =
filled indicating whether you want your telephone number to be PUBLIC or =
PRIVATE. (of course, the default is PUBLIC if you leave it blank.) When =
you make a call, the PRIVATE or PUBLIC status value is retrieved and set =
the Address Presentation Restricted Indicator (APRI) field in SS7 =
environment (Note that, CLIR toggles the values), then the terminating =
side of the network based on the APRI value to determine whether the =
Calling Party Number shall be delivered to the callee's CPE.
=20
Since we want to push the decision making to the UA, we should =
standardize the rule such that the UA shall display the From information =
based on Privacy indication. If we think that too many UAs won't follow =
this rule, than we should allow the Service Provider of terminating side =
(not the originating side) to alter the From information before =
terminating the call to the UA based on the Privacy indication.
=20
This is my penny thought,
=20
Mark Chiou=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D765415513-15042002>Dean=20
wrote:</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D765415513-15042002>#####################################</SPAN></=
FONT></DIV>
<DIV><FONT color=3D#0000ff size=3D2><SPAN class=3D765415513-15042002>
<DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
size=3D+0><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
color=3D#0000ff=20
face=3DArial>The point I'm trying to make is that To and From are USER =
supplied=20
information. The&nbsp;Service Provider doesn't insert them, inspect =
them, or=20
have any responsibility for them. Insisting that the service provider =
change=20
them is just like&nbsp;insisting that the service provider change any =
other=20
user-supplied information,&nbsp;such as the&nbsp;words I'm typing now. =
Anything=20
I put in my To or from field is deliberately and explicitly intended to =
be=20
delivered to the terminating side of the dialog, and if as a service =
provider=20
you FAIL to do that, then you have DAMAGED my data, and my lawyers will =
be in=20
touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
size=3D+0><FONT color=3D#0000ff size=3D2><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT=20
face=3DArial>&nbsp;</FONT></DIV>
<DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
size=3D+0><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D819102015-12042002>The=20
calling-party-identity presentation infromation (CLIP or RPID) is NOT =
inserted=20
by the user. It is inserted by the network, WITHOUT THE CONSENT of the =
user. As=20
such, the operator is responsible for the protection of that information =
in a=20
manner consistent with regulation and the expressed wishes of the=20
user.</SPAN></FONT></FONT></FONT></FONT></DIV>
<DIV><FONT lang=3D0 FAMILY=3D"SANSSERIF"><FONT face=3DArial><SPAN=20
class=3D819102015-12042002><SPAN=20
class=3D765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>=

<DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
color=3D#0000ff><SPAN=20
class=3D819102015-12042002></SPAN></FONT></FONT><FONT =
face=3DArial>&nbsp;<SPAN=20
class=3D765415513-15042002>##############################################=
############</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>No, The CLIP is =
inserted by=20
the network WITH THE CONSENT of the user. When you subscribe a line from =
a PSTN=20
service provider, the form you filled indicating whether you want your =
telephone=20
number to be PUBLIC or PRIVATE. (of course, the default is PUBLIC if you =
leave=20
it blank.) When you make a call, the PRIVATE or PUBLIC status value is =
retrieved=20
and set the Address Presentation Restricted Indicator (APRI) field in =
SS7=20
environment (Note that, CLIR toggles the values), then the terminating =
side of=20
the network based on the APRI value to determine whether the Calling =
Party=20
Number shall be delivered to the callee's CPE.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN =
class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>Since we want =
to push the=20
decision making to the UA, we should standardize the rule such that the =
UA shall=20
display the From information based on Privacy indication. If we think =
that too=20
many UAs won't follow this rule, than we should allow the Service =
Provider of=20
terminating side (not the originating side) to alter the From =
information before=20
terminating the call to the UA based on the Privacy=20
indication.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN =
class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
class=3D765415513-15042002>This is my penny=20
thought,</SPAN></FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
class=3D765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
class=3D765415513-15042002>Mark=20
Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BODY><=
/HTML>

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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 09:44:25 -0500
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MessageYou pretty much have to assume that the receiving user sees
_anything_ you send his UA, except in the rare case where the user has no
control over the UA behavior. Asking a UAS to hide information from its
owner is not useful in the generic case.

I agree with Dean on this in general, although it really doesn't matter if
any _network_ inserted identification is there with the consent of the
originating user or not. A privacy request need _only_ apply to
network-inserted identification data. The UAC has every opportunity _not_ to
insert identification if it does not want to.

Expecting the network to screen out ID that _I_ provide is a slippery slope.
What if I put "call from Ben" in a Subject header? Do you screen that out as
well? What if I _tell_ the destination user who I am in an IM associated
with the call? What about the media session,?

 -----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chiou, Mark
Sent: Monday, April 15, 2002 9:28 AM
To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Dean wrote:
  #####################################
  The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch!

  The calling-party-identity presentation infromation (CLIP or RPID) is NOT
inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of
the user. As such, the operator is responsible for the protection of that
information in a manner consistent with regulation and the expressed wishes
of the user.
  <snip>
   ##########################################################
  No, The CLIP is inserted by the network WITH THE CONSENT of the user. When
you subscribe a line from a PSTN service provider, the form you filled
indicating whether you want your telephone number to be PUBLIC or PRIVATE.
(of course, the default is PUBLIC if you leave it blank.) When you make a
call, the PRIVATE or PUBLIC status value is retrieved and set the Address
Presentation Restricted Indicator (APRI) field in SS7 environment (Note
that, CLIR toggles the values), then the terminating side of the network
based on the APRI value to determine whether the Calling Party Number shall
be delivered to the callee's CPE.

  Since we want to push the decision making to the UA, we should standardize
the rule such that the UA shall display the From information based on
Privacy indication. If we think that too many UAs won't follow this rule,
than we should allow the Service Provider of terminating side (not the
originating side) to alter the From information before terminating the call
to the UA based on the Privacy indication.

  This is my penny thought,

  Mark Chiou

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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D435513614-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>You=20
pretty much have to assume that the receiving user sees _anything_ you =
send his=20
UA, except in the rare case where the user has no control over the UA =
behavior.=20
Asking a UAS to hide information from its owner is not useful in the =
generic=20
case.</FONT></SPAN></DIV>
<DIV><SPAN class=3D435513614-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D435513614-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
agree with Dean on this in general, although it really doesn't matter if =
any=20
_network_ inserted identification is there with the consent of the =
originating=20
user or not. A privacy request need _only_ apply to network-inserted=20
identification data. The UAC has every opportunity _not_ to insert=20
identification if it does not want to. </FONT></SPAN></DIV>
<DIV><SPAN class=3D435513614-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D435513614-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2>Expecting the network to screen out ID that _I_ provide is a =
slippery=20
slope. What if I put "call from Ben" in a Subject header? Do you screen =
that out=20
as well? What if I _tell_ the destination user who I am in an IM =
associated with=20
the call? What about the media session,?</FONT></SPAN></DIV>
<DIV><SPAN class=3D435513614-15042002></SPAN><FONT face=3DTahoma><FONT =
size=3D2><SPAN=20
class=3D435513614-15042002><FONT face=3DArial=20
color=3D#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT face=3DTahoma><FONT size=3D2><SPAN=20
class=3D435513614-15042002>&nbsp;</SPAN>-----Original =
Message-----<BR><B>From:</B>=20
sip-admin@ietf.org [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou, =

Mark<BR><B>Sent:</B> Monday, April 15, 2002 9:28 AM<BR><B>To:</B> Dean =
Willis;=20
Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: =
[Sip]=20
Comment, SIP Privacy draft<BR><BR></DIV></FONT></FONT>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D765415513-15042002>Dean=20
  wrote:</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  =
class=3D765415513-15042002>#####################################</SPAN></=
FONT></DIV>
  <DIV><FONT color=3D#0000ff size=3D2><SPAN class=3D765415513-15042002>
  <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
  size=3D+0><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
face=3DArial=20
  color=3D#0000ff>The point I'm trying to make is that To and From are =
USER=20
  supplied information. The&nbsp;Service Provider doesn't insert them, =
inspect=20
  them, or have any responsibility for them. Insisting that the service =
provider=20
  change them is just like&nbsp;insisting that the service provider =
change any=20
  other user-supplied information,&nbsp;such as the&nbsp;words I'm =
typing now.=20
  Anything I put in my To or from field is deliberately and explicitly =
intended=20
  to be delivered to the terminating side of the dialog, and if as a =
service=20
  provider you FAIL to do that, then you have DAMAGED my data, and my =
lawyers=20
  will be in touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
  size=3D+0><FONT color=3D#0000ff size=3D2><SPAN=20
  class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT=20
  face=3DArial></FONT>&nbsp;</DIV>
  <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
  size=3D+0><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D819102015-12042002>The calling-party-identity presentation =
infromation=20
  (CLIP or RPID) is NOT inserted by the user. It is inserted by the =
network,=20
  WITHOUT THE CONSENT of the user. As such, the operator is responsible =
for the=20
  protection of that information in a manner consistent with regulation =
and the=20
  expressed wishes of the user.</SPAN></FONT></FONT></FONT></FONT></DIV>
  <DIV><FONT lang=3D0 FAMILY=3D"SANSSERIF"><FONT face=3DArial><SPAN=20
  class=3D819102015-12042002><SPAN=20
  =
class=3D765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>=

  <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
color=3D#0000ff><SPAN=20
  class=3D819102015-12042002></SPAN></FONT></FONT><FONT =
face=3DArial>&nbsp;<SPAN=20
  =
class=3D765415513-15042002>##############################################=
############</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>No, The CLIP =
is inserted=20
  by the network WITH THE CONSENT of the user. When you subscribe a line =
from a=20
  PSTN service provider, the form you filled indicating whether you want =
your=20
  telephone number to be PUBLIC or PRIVATE. (of course, the default is =
PUBLIC if=20
  you leave it blank.) When you make a call, the PRIVATE or PUBLIC =
status value=20
  is retrieved and set the Address Presentation Restricted Indicator =
(APRI)=20
  field in SS7 environment (Note that, CLIR toggles the values), then =
the=20
  terminating side of the network based on the APRI value to determine =
whether=20
  the Calling Party Number shall be delivered to the callee's=20
  CPE.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial><SPAN=20
class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>Since we want =
to push the=20
  decision making to the UA, we should standardize the rule such that =
the UA=20
  shall display the From information based on Privacy indication. If we =
think=20
  that too many UAs won't follow this rule, than we should allow the =
Service=20
  Provider of terminating side (not the originating side) to alter the =
From=20
  information before terminating the call to the UA based on the Privacy =

  indication.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial><SPAN=20
class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
  class=3D765415513-15042002>This is my penny=20
  thought,</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
  class=3D765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
  class=3D765415513-15042002>Mark=20
  =
Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQ=
UOTE></BODY></HTML>

------=_NextPart_000_003A_01C1E462.21297EA0--


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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 10:02:30 -0500
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Thread-Topic: Summary of RE: [Sip] Comment, SIP Privacy draft
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From: "Chiou, Mark" <MChiou@Santera.com>
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        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
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Sounds like you prefer separate the User provided information from =
Network provided information. I agree with you as long as we address a =
privacy header which is inserted by the originating Service Provider and =
extracted by the terminating Service provider. (The Privacy information =
should not be delivered to the UAS.)
The same Privacy header information is also used to interwork with PSTN =
world for APRI purpose.=20
=20
Mark Chiou

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: Monday, April 15, 2002 9:44 AM
To: Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


You pretty much have to assume that the receiving user sees _anything_ =
you send his UA, except in the rare case where the user has no control =
over the UA behavior. Asking a UAS to hide information from its owner is =
not useful in the generic case.
=20
I agree with Dean on this in general, although it really doesn't matter =
if any _network_ inserted identification is there with the consent of =
the originating user or not. A privacy request need _only_ apply to =
network-inserted identification data. The UAC has every opportunity =
_not_ to insert identification if it does not want to.=20
=20
Expecting the network to screen out ID that _I_ provide is a slippery =
slope. What if I put "call from Ben" in a Subject header? Do you screen =
that out as well? What if I _tell_ the destination user who I am in an =
IM associated with the call? What about the media session,?
=20
 -----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chiou, =
Mark
Sent: Monday, April 15, 2002 9:28 AM
To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft



Dean wrote:
#####################################

The point I'm trying to make is that To and From are USER supplied =
information. The Service Provider doesn't insert them, inspect them, or =
have any responsibility for them. Insisting that the service provider =
change them is just like insisting that the service provider change any =
other user-supplied information, such as the words I'm typing now. =
Anything I put in my To or from field is deliberately and explicitly =
intended to be delivered to the terminating side of the dialog, and if =
as a service provider you FAIL to do that, then you have DAMAGED my =
data, and my lawyers will be in touch!
=20
The calling-party-identity presentation infromation (CLIP or RPID) is =
NOT inserted by the user. It is inserted by the network, WITHOUT THE =
CONSENT of the user. As such, the operator is responsible for the =
protection of that information in a manner consistent with regulation =
and the expressed wishes of the user.
<snip>
 ##########################################################
No, The CLIP is inserted by the network WITH THE CONSENT of the user. =
When you subscribe a line from a PSTN service provider, the form you =
filled indicating whether you want your telephone number to be PUBLIC or =
PRIVATE. (of course, the default is PUBLIC if you leave it blank.) When =
you make a call, the PRIVATE or PUBLIC status value is retrieved and set =
the Address Presentation Restricted Indicator (APRI) field in SS7 =
environment (Note that, CLIR toggles the values), then the terminating =
side of the network based on the APRI value to determine whether the =
Calling Party Number shall be delivered to the callee's CPE.
=20
Since we want to push the decision making to the UA, we should =
standardize the rule such that the UA shall display the From information =
based on Privacy indication. If we think that too many UAs won't follow =
this rule, than we should allow the Service Provider of terminating side =
(not the originating side) to alter the From information before =
terminating the call to the UA based on the Privacy indication.
=20
This is my penny thought,
=20
Mark Chiou=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D297275314-15042002>Sounds=20
like you prefer separate the User provided information from Network =
provided=20
information. I agree with you as long as we address a privacy header =
which is=20
inserted by the originating Service Provider and extracted by the =
terminating=20
Service provider. (The Privacy information should not be delivered to =
the=20
UAS.)</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D297275314-15042002>The=20
same Privacy header information is also used to interwork with PSTN =
world for=20
APRI purpose. </SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D297275314-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D297275314-15042002>Mark=20
Chiou</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: =
0px; PADDING-LEFT: 5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Ben Campbell=20
  [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> Monday, April 15, =
2002 9:44=20
  AM<BR><B>To:</B> Chiou, Mark; Dean Willis; Mpierce1@aol.com;=20
  sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP =
Privacy=20
  draft<BR><BR></DIV></FONT>
  <DIV><SPAN class=3D435513614-15042002><FONT color=3D#0000ff =
face=3DArial size=3D2>You=20
  pretty much have to assume that the receiving user sees _anything_ you =
send=20
  his UA, except in the rare case where the user has no control over the =
UA=20
  behavior. Asking a UAS to hide information from its owner is not =
useful in the=20
  generic case.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D435513614-15042002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D435513614-15042002><FONT color=3D#0000ff =
face=3DArial size=3D2>I=20
  agree with Dean on this in general, although it really doesn't matter =
if any=20
  _network_ inserted identification is there with the consent of the =
originating=20
  user or not. A privacy request need _only_ apply to network-inserted=20
  identification data. The UAC has every opportunity _not_ to insert=20
  identification if it does not want to. </FONT></SPAN></DIV>
  <DIV><SPAN class=3D435513614-15042002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D435513614-15042002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2>Expecting the network to screen out ID that _I_ provide is a =
slippery=20
  slope. What if I put "call from Ben" in a Subject header? Do you =
screen that=20
  out as well? What if I _tell_ the destination user who I am in an IM=20
  associated with the call? What about the media =
session,?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D435513614-15042002></SPAN><FONT face=3DTahoma><FONT =

  size=3D2><SPAN class=3D435513614-15042002><FONT color=3D#0000ff=20
  face=3DArial>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=3DTahoma><FONT size=3D2><SPAN=20
  class=3D435513614-15042002>&nbsp;</SPAN>-----Original=20
  Message-----<BR><B>From:</B> sip-admin@ietf.org=20
  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou, =
Mark<BR><B>Sent:</B>=20
  Monday, April 15, 2002 9:28 AM<BR><B>To:</B> Dean Willis; =
Mpierce1@aol.com;=20
  sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP =
Privacy=20
  draft<BR><BR></DIV></FONT></FONT>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; =
MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
    class=3D765415513-15042002>Dean wrote:</SPAN></FONT></DIV>
    <DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
    =
class=3D765415513-15042002>#####################################</SPAN></=
FONT></DIV>
    <DIV><FONT color=3D#0000ff size=3D2><SPAN =
class=3D765415513-15042002>
    <DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
    size=3D+0><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
color=3D#0000ff=20
    face=3DArial>The point I'm trying to make is that To and From are =
USER=20
    supplied information. The&nbsp;Service Provider doesn't insert them, =
inspect=20
    them, or have any responsibility for them. Insisting that the =
service=20
    provider change them is just like&nbsp;insisting that the service =
provider=20
    change any other user-supplied information,&nbsp;such as =
the&nbsp;words I'm=20
    typing now. Anything I put in my To or from field is deliberately =
and=20
    explicitly intended to be delivered to the terminating side of the =
dialog,=20
    and if as a service provider you FAIL to do that, then you have =
DAMAGED my=20
    data, and my lawyers will be in=20
    touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
    <DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
    size=3D+0><FONT color=3D#0000ff size=3D2><SPAN=20
    class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT=20
    face=3DArial></FONT>&nbsp;</DIV>
    <DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
    size=3D+0><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
    class=3D819102015-12042002>The calling-party-identity presentation =
infromation=20
    (CLIP or RPID) is NOT inserted by the user. It is inserted by the =
network,=20
    WITHOUT THE CONSENT of the user. As such, the operator is =
responsible for=20
    the protection of that information in a manner consistent with =
regulation=20
    and the expressed wishes of the=20
    user.</SPAN></FONT></FONT></FONT></FONT></DIV>
    <DIV><FONT lang=3D0 FAMILY=3D"SANSSERIF"><FONT face=3DArial><SPAN=20
    class=3D819102015-12042002><SPAN=20
    =
class=3D765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>=

    <DIV><FONT color=3D#000000 lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
color=3D#0000ff><SPAN=20
    class=3D819102015-12042002></SPAN></FONT></FONT><FONT =
face=3DArial>&nbsp;<SPAN=20
    =
class=3D765415513-15042002>##############################################=
############</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>No, The =
CLIP is=20
    inserted by the network WITH THE CONSENT of the user. When you =
subscribe a=20
    line from a PSTN service provider, the form you filled indicating =
whether=20
    you want your telephone number to be PUBLIC or PRIVATE. (of course, =
the=20
    default is PUBLIC if you leave it blank.) When you make a call, the =
PRIVATE=20
    or PUBLIC status value is retrieved and set the Address Presentation =

    Restricted Indicator (APRI) field in SS7 environment (Note that, =
CLIR=20
    toggles the values), then the terminating side of the network based =
on the=20
    APRI value to determine whether the Calling Party Number shall be =
delivered=20
    to the callee's CPE.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial><SPAN=20
    class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>Since we =
want to push=20
    the decision making to the UA, we should standardize the rule such =
that the=20
    UA shall display the From information based on Privacy indication. =
If we=20
    think that too many UAs won't follow this rule, than we should allow =
the=20
    Service Provider of terminating side (not the originating side) to =
alter the=20
    From information before terminating the call to the UA based on the =
Privacy=20
    indication.</SPAN></FONT></DIV>
    <DIV><FONT face=3DArial><SPAN=20
    class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
    class=3D765415513-15042002>This is my penny=20
    thought,</SPAN></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
    class=3D765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =
face=3DArial><SPAN=20
    class=3D765415513-15042002>Mark=20
    =
Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQ=
UOTE></BLOCKQUOTE></BODY></HTML>

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From daemon@ns.ietf.org  Mon Apr 15 11:40:51 2002
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Date: Mon, 15 Apr 2002 18:06:16 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: Dean Willis
	 <dean.willis@softarmor.com>, Mpierce1@aol.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
In-Reply-To: <A3C2399B2FACD411A54200508BE39C74054F7128@zwcwd00r.europe.nortel.com>
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Mark Watson writes:

 > I think many service providers will wish to do the same, and indeed this
 > checking is required in 3GPP. The situation is much clearer in this case,
 > because in order to access the service at all, the user MUST insert their
 > true identity (or 'anonymous') into the From: field.

if the call goes to pstn, the user gets calling number anonymity by
using Anonymous as the display string.  if the call goes to another sip
user, the same thing happens, but i must implement b2ua and rewrite the
from header.  no standardization is needed in order to accomplish this.

-- juha

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From daemon@optimus.ietf.org  Mon Apr 15 12:43:30 2002
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Date: Mon, 15 Apr 2002 11:10:06 -0500
To: Rohan Mahy <rohan@cisco.com>, <sip@ietf.org>, <sipping@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
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Subject: [Sip] Re: [Sipping] SIP/SIPPING Interim Meeting
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At 01:39 PM 4/13/2002 -0700, Rohan Mahy wrote:
>Hi,
>
>I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in Las
>Vegas, Nevada.  

You chose the week of InterOp Las Vegas, plan on having the meeting *at* the
same location as that convention, and have really confirmed that there are
rooms available at the hotel attached to the convention center?!

Seems even if you've picked a location that does have the room, that you
couldn't have picked a more hectic week or place to have it in (except maybe
CES week in Januarys)...

Or was this a plan to have during InterOp?

>this date and location were changed (with AD approval)
>as a fallback due to conflicts with the previously proposed time and place
>(Boston, the following week).
*************************************
"People generally demand more respect for their own rights than they are
willing to allow for others"

James M. Polk
Consulting Engineer
Office of the CTO

Cisco Systems
2200 East President George Bush Turnpike 
Richardson, TX  75082 USA
w) 972.813.5208
f)  972.813.5280
www.cisco.com
--=====================_5317165==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
At 01:39 PM 4/13/2002 -0700, Rohan Mahy wrote:<br>
&gt;Hi,<br>
&gt;<br>
&gt;I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in
Las<br>
&gt;Vegas, Nevada.&nbsp; <br>
<br>
You chose the week of<u> InterOp Las Vegas</u>, plan on having the
meeting *at* the same location as that convention, and have really
confirmed that there are rooms available at the hotel attached to the
convention center?!<br>
<br>
Seems even if you've picked a location that does have the room, that you
couldn't have picked a more hectic week or place to have it in (except
maybe CES week in Januarys)...<br>
<br>
Or was this a plan to have during InterOp?<br>
<br>
&gt;this date and location were changed (with AD approval)<br>
&gt;as a fallback due to conflicts with the previously proposed time and
place<br>
&gt;(Boston, the following week).<br>

<div align="center">
*************************************<br>
&quot;People generally demand more respect for their own rights than they
are willing to allow for others&quot;<br>
<br>
</div>
James M. Polk<br>
Consulting Engineer<br>
Office of the CTO<br>
<br>
Cisco Systems<br>
2200 East President George Bush Turnpike <br>
Richardson, TX&nbsp; 75082 USA<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5280<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_5317165==_.ALT--


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From daemon@optimus.ietf.org  Mon Apr 15 12:47:25 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Ben,
 
I know this is a long thread, but I have addressed all these points before.
Please see below...

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: 15 April 2002 15:44
To: Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


You pretty much have to assume that the receiving user sees _anything_ you
send his UA, except in the rare case where the user has no control over the
UA behavior. Asking a UAS to hide information from its owner is not useful
in the generic case.
 
I agree with Dean on this in general, although it really doesn't matter if
any _network_ inserted identification is there with the consent of the
originating user or not. A privacy request need _only_ apply to
network-inserted identification data. The UAC has every opportunity _not_ to
insert identification if it does not want to. 

It doesn't matter what device inserts the information. It matters whether it
is part of the service for the information to be inserted. If it is a part
of the service, then the responsibility lies with the Service Provider -
they defined the service and they are responsible for things the users do
effectively at their behest.
 
If I check the From: field I receive from my users is valid, then I am
forcing them to put either their identity or 'Anonymous' in there.
 
If there is then a privacy requirement arising from the rights of ANOTHER
PERSON, NOT THE USER, then I have to do something about this - I as the
Service Provider have the responsibility, not the user. An example is when
the *subscriber* has requested anonymity.
 
If I let my users use any old From value, then the situation is not so
clear, but I still argue that for fields whose intention is defined for
transport of a particular identity, then the Service Provider has some
responsibility. (cf Special Arrangements in the ISDN).

Expecting the network to screen out ID that _I_ provide is a slippery slope.
What if I put "call from Ben" in a Subject header? Do you screen that out as
well? What if I _tell_ the destination user who I am in an IM associated
with the call? What about the media session,? 
 

Personally, I don't think screening out the Subject would be necessary,
although there might be some market demand for services which did this.
Subject is a genuinely transparent field. There is no intention that it
contain your identity, and UAs are unlikely to insert your identity here
automatically.
 
...Mark

 
 -----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chiou, Mark
Sent: Monday, April 15, 2002 9:28 AM
To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft



Dean wrote:
#####################################

The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch!
 
The calling-party-identity presentation infromation (CLIP or RPID) is NOT
inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of
the user. As such, the operator is responsible for the protection of that
information in a manner consistent with regulation and the expressed wishes
of the user.
<snip>
 ##########################################################
No, The CLIP is inserted by the network WITH THE CONSENT of the user. When
you subscribe a line from a PSTN service provider, the form you filled
indicating whether you want your telephone number to be PUBLIC or PRIVATE.
(of course, the default is PUBLIC if you leave it blank.) When you make a
call, the PRIVATE or PUBLIC status value is retrieved and set the Address
Presentation Restricted Indicator (APRI) field in SS7 environment (Note
that, CLIR toggles the values), then the terminating side of the network
based on the APRI value to determine whether the Calling Party Number shall
be delivered to the callee's CPE.
 
Since we want to push the decision making to the UA, we should standardize
the rule such that the UA shall display the From information based on
Privacy indication. If we think that too many UAs won't follow this rule,
than we should allow the Service Provider of terminating side (not the
originating side) to alter the From information before terminating the call
to the UA based on the Privacy indication.
 
This is my penny thought,
 
Mark Chiou 


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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=404191416-15042002>Ben,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=404191416-15042002>I 
know this is a long thread, but I have addressed all these points before. Please 
see below...</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ben Campbell 
  [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002 
  15:44<BR><B>To:</B> Chiou, Mark; Dean Willis; Mpierce1@aol.com; 
  sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP Privacy 
  draft<BR><BR></DIV></FONT>
  <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial size=2>You 
  pretty much have to assume that the receiving user sees _anything_ you send 
  his UA, except in the rare case where the user has no control over the UA 
  behavior. Asking a UAS to hide information from its owner is not useful in the 
  generic case.</FONT></SPAN></DIV>
  <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial size=2>I 
  agree with Dean on this in general, although it really doesn't matter if any 
  _network_ inserted identification is there with the consent of the originating 
  user or not. A privacy request need _only_ apply to network-inserted 
  identification data. The UAC has every opportunity _not_ to insert 
  identification if it does not want to. </FONT></SPAN></DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=404191416-15042002>It 
doesn't matter what device inserts the information. It matters whether it is 
part of the service for the information to be inserted. If it is a part of the 
service, then the responsibility lies with the Service Provider - they defined 
the service and they are responsible for things the users do effectively at 
their behest.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=404191416-15042002>If I 
check the From: field I receive from my users is valid, then I am forcing them 
to put either their identity or 'Anonymous' in there.</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=404191416-15042002>If 
there is then a privacy requirement arising from the rights of ANOTHER PERSON, 
NOT THE USER, then I have to do something about this - I as the Service Provider 
have the responsibility, not the user. An example is when the *subscriber* has 
requested anonymity.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=404191416-15042002>If I 
let my users use any old From value, then the situation is not so clear, but I 
still argue that for fields whose intention is defined for transport of a 
particular identity, then the Service Provider has some responsibility. (cf 
Special Arrangements in the ISDN).</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
  size=2>Expecting the network to screen out ID that _I_ provide is a slippery 
  slope. What if I put "call from Ben" in a Subject header? Do you screen that 
  out as well? What if I _tell_ the destination user who I am in an IM 
  associated with the call? What about the media session,?<FONT face=Verdana 
  size=1><SPAN 
  class=404191416-15042002>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
  size=2><FONT face=Verdana size=1><SPAN 
  class=404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV></BLOCKQUOTE>
<DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN class=404191416-15042002>Personally, I 
don't think screening out the Subject would be necessary, although there might 
be some market demand for services which did this.&nbsp;Subject is a genuinely 
transparent field. There is no intention that it contain your identity, and UAs 
are unlikely to insert your identity here 
automatically.</SPAN></FONT></FONT></SPAN></FONT></DIV>
<DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN 
class=404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN 
class=404191416-15042002>...Mark</SPAN></FONT></FONT></SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV><SPAN class=435513614-15042002></SPAN><FONT face=Tahoma><FONT 
  size=2><SPAN class=435513614-15042002><FONT color=#0000ff 
  face=Arial>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Tahoma><FONT size=2><SPAN 
  class=435513614-15042002>&nbsp;</SPAN>-----Original 
  Message-----<BR><B>From:</B> sip-admin@ietf.org 
  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou, Mark<BR><B>Sent:</B> 
  Monday, April 15, 2002 9:28 AM<BR><B>To:</B> Dean Willis; Mpierce1@aol.com; 
  sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP Privacy 
  draft<BR><BR></DIV></FONT></FONT>
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
    class=765415513-15042002>Dean wrote:</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
    class=765415513-15042002>#####################################</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff size=2><SPAN class=765415513-15042002>
    <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
    size=+0><FONT size=2><SPAN class=819102015-12042002><FONT color=#0000ff 
    face=Arial>The point I'm trying to make is that To and From are USER 
    supplied information. The&nbsp;Service Provider doesn't insert them, inspect 
    them, or have any responsibility for them. Insisting that the service 
    provider change them is just like&nbsp;insisting that the service provider 
    change any other user-supplied information,&nbsp;such as the&nbsp;words I'm 
    typing now. Anything I put in my To or from field is deliberately and 
    explicitly intended to be delivered to the terminating side of the dialog, 
    and if as a service provider you FAIL to do that, then you have DAMAGED my 
    data, and my lawyers will be in 
    touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
    <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
    size=+0><FONT color=#0000ff size=2><SPAN 
    class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT 
    face=Arial></FONT>&nbsp;</DIV>
    <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
    size=+0><FONT color=#0000ff face=Arial size=2><SPAN 
    class=819102015-12042002>The calling-party-identity presentation infromation 
    (CLIP or RPID) is NOT inserted by the user. It is inserted by the network, 
    WITHOUT THE CONSENT of the user. As such, the operator is responsible for 
    the protection of that information in a manner consistent with regulation 
    and the expressed wishes of the 
    user.</SPAN></FONT></FONT></FONT></FONT></DIV>
    <DIV><FONT lang=0 FAMILY="SANSSERIF"><FONT face=Arial><SPAN 
    class=819102015-12042002><SPAN 
    class=765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>
    <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT color=#0000ff><SPAN 
    class=819102015-12042002></SPAN></FONT></FONT><FONT face=Arial>&nbsp;<SPAN 
    class=765415513-15042002>##########################################################</SPAN></FONT></DIV>
    <DIV><FONT face=Arial><SPAN class=765415513-15042002>No, The CLIP is 
    inserted by the network WITH THE CONSENT of the user. When you subscribe a 
    line from a PSTN service provider, the form you filled indicating whether 
    you want your telephone number to be PUBLIC or PRIVATE. (of course, the 
    default is PUBLIC if you leave it blank.) When you make a call, the PRIVATE 
    or PUBLIC status value is retrieved and set the Address Presentation 
    Restricted Indicator (APRI) field in SS7 environment (Note that, CLIR 
    toggles the values), then the terminating side of the network based on the 
    APRI value to determine whether the Calling Party Number shall be delivered 
    to the callee's CPE.</SPAN></FONT></DIV>
    <DIV><FONT face=Arial><SPAN 
    class=765415513-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=Arial><SPAN class=765415513-15042002>Since we want to push 
    the decision making to the UA, we should standardize the rule such that the 
    UA shall display the From information based on Privacy indication. If we 
    think that too many UAs won't follow this rule, than we should allow the 
    Service Provider of terminating side (not the originating side) to alter the 
    From information before terminating the call to the UA based on the Privacy 
    indication.</SPAN></FONT></DIV>
    <DIV><FONT face=Arial><SPAN 
    class=765415513-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT face=Arial><SPAN 
    class=765415513-15042002>This is my penny 
    thought,</SPAN></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT face=Arial><SPAN 
    class=765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT face=Arial><SPAN 
    class=765415513-15042002>Mark 
    Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Mon Apr 15 12:50:59 2002
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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From: "Christian Huitema" <huitema@windows.microsoft.com>
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I have heard many contributors assert that to get real anonymity, one
will have to use a relay, e.g. an "anonymizing B2BUA". I would like to
point out a failure in this line of reasoning: anonymizing relays have
been tried before, for e-mail, and the experience was absolutely not
conclusive. In fact, two different kind of anonymizers have been tried:
free ones, that kept absolutely no record, and subscriber-based, which
kept some records. Both have failed, for different reason.

The failure mode of a completely free relay is basically spam, and
similar sort of abuse. There is documentation available on the fate for
example of the cypherpunk experiment: this set of relays was set up to
provide true anonymity on the Internet, using a sophisticated mix of
random redirection and cryptography. In theory, the design was
excellent. In practice, the system was quickly overwhelmed by porn
traffic, and the relays dropped one by one out of the system because
they could not bear the resulting congestion.

Subscriber based relays are relays that keep some state: they actually
identify the sender, before washing out the identity in the outgoing
message. They are more robust to congestion, since they can get some
funding in the process, but they have a different failure mode: lawsuit.
Since they keep some records, the records can be subpoenaed, and the
provider can be forced to disclose the identity of the sending party. In
fact, in many codes, if the providers refuse to do so, they may be
deemed an accomplice of whatever action was carried out in the anonymous
exchange. Many providers have chosen to close their doors rather than
have to defend more lawsuit -- often after a brush with the Church of
Scientology.

If we really want practical anonymity, it has to be based on the UA, and
it should not rely on the help of a third party. We should use a
combination of encryption, pre-paid subscriptions, and anonymous IP
addresses (e.g. IPv6 privacy addresses.)

-- Christian Huitema

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From daemon@optimus.ietf.org  Mon Apr 15 12:55:00 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Juha Heinanen'" <jh@lohi.eng.song.fi>
Cc: Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 17:24:17 +0100
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Juha,

I am not arguing that standardisation is required. I believe the necessary
capabilities can be provided with a B2BUA as described by Jonathan and
yourself (or something between a proxy and a full B2BUA).

I am arguing against the belief that there is NEVER a requirement for the
network to modify fields like From/To. It is this belief that has led other
people to distust these fields and specify that no information should be
included.

The worry is that if you allow potentially private information into those
fields, then this will cause trouble because there is an assumption/belief
at large in the SIP community that these fields should never be modified.

If we could agree that it is at least _likely_ that privacy services which
modify From/To will be required by Public Service providers, there would be
no more reason to distruct the fields and we could all use them for their
intended purpose.

...Mark

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@lohi.eng.song.fi]
> Sent: 15 April 2002 16:06
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: Dean Willis; Mpierce1@aol.com; sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Mark Watson writes:
> 
>  > I think many service providers will wish to do the same, 
> and indeed this
>  > checking is required in 3GPP. The situation is much 
> clearer in this case,
>  > because in order to access the service at all, the user 
> MUST insert their
>  > true identity (or 'anonymous') into the From: field.
> 
> if the call goes to pstn, the user gets calling number anonymity by
> using Anonymous as the display string.  if the call goes to 
> another sip
> user, the same thing happens, but i must implement b2ua and 
> rewrite the
> from header.  no standardization is needed in order to 
> accomplish this.
> 
> -- juha
> 

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<P><FONT SIZE=3D2>Juha,</FONT>
</P>

<P><FONT SIZE=3D2>I am not arguing that standardisation is required. I =
believe the necessary capabilities can be provided with a B2BUA as =
described by Jonathan and yourself (or something between a proxy and a =
full B2BUA).</FONT></P>

<P><FONT SIZE=3D2>I am arguing against the belief that there is NEVER a =
requirement for the network to modify fields like From/To. It is this =
belief that has led other people to distust these fields and specify =
that no information should be included.</FONT></P>

<P><FONT SIZE=3D2>The worry is that if you allow potentially private =
information into those fields, then this will cause trouble because =
there is an assumption/belief at large in the SIP community that these =
fields should never be modified.</FONT></P>

<P><FONT SIZE=3D2>If we could agree that it is at least _likely_ that =
privacy services which modify From/To will be required by Public =
Service providers, there would be no more reason to distruct the fields =
and we could all use them for their intended purpose.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Juha Heinanen [<A =
HREF=3D"mailto:jh@lohi.eng.song.fi">mailto:jh@lohi.eng.song.fi</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 15 April 2002 16:06</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Dean Willis; Mpierce1@aol.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark Watson writes:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; I think many service providers will =
wish to do the same, </FONT>
<BR><FONT SIZE=3D2>&gt; and indeed this</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; checking is required in 3GPP. The =
situation is much </FONT>
<BR><FONT SIZE=3D2>&gt; clearer in this case,</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; because in order to access the =
service at all, the user </FONT>
<BR><FONT SIZE=3D2>&gt; MUST insert their</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; true identity (or 'anonymous') into =
the From: field.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; if the call goes to pstn, the user gets calling =
number anonymity by</FONT>
<BR><FONT SIZE=3D2>&gt; using Anonymous as the display string.&nbsp; if =
the call goes to </FONT>
<BR><FONT SIZE=3D2>&gt; another sip</FONT>
<BR><FONT SIZE=3D2>&gt; user, the same thing happens, but i must =
implement b2ua and </FONT>
<BR><FONT SIZE=3D2>&gt; rewrite the</FONT>
<BR><FONT SIZE=3D2>&gt; from header.&nbsp; no standardization is needed =
in order to </FONT>
<BR><FONT SIZE=3D2>&gt; accomplish this.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- juha</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@optimus.ietf.org  Mon Apr 15 12:55:24 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Christian Huitema'" <huitema@windows.microsoft.com>, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 17:33:08 +0100
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Christian,

If you want to provide complete unassailable anonymity, I agree completely.
I have been arguing about providing a similar level of anonymity as the
PSTN, where the called party is not informed who the call is from by the
service itself. It's more about Data Protection than strict anonymity.

In the PSTN there is no problem for certain authorised organisations
(including the courts) to obtain the details of the call from the Service
Providers if they have good reason. Data Protection does not protect you
against these organisations.

...Mark


> -----Original Message-----
> From: Christian Huitema [mailto:huitema@windows.microsoft.com]
> Sent: 15 April 2002 17:10
> To: sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> I have heard many contributors assert that to get real anonymity, one
> will have to use a relay, e.g. an "anonymizing B2BUA". I would like to
> point out a failure in this line of reasoning: anonymizing relays have
> been tried before, for e-mail, and the experience was absolutely not
> conclusive. In fact, two different kind of anonymizers have 
> been tried:
> free ones, that kept absolutely no record, and subscriber-based, which
> kept some records. Both have failed, for different reason.
> 
> The failure mode of a completely free relay is basically spam, and
> similar sort of abuse. There is documentation available on 
> the fate for
> example of the cypherpunk experiment: this set of relays was set up to
> provide true anonymity on the Internet, using a sophisticated mix of
> random redirection and cryptography. In theory, the design was
> excellent. In practice, the system was quickly overwhelmed by porn
> traffic, and the relays dropped one by one out of the system because
> they could not bear the resulting congestion.
> 
> Subscriber based relays are relays that keep some state: they actually
> identify the sender, before washing out the identity in the outgoing
> message. They are more robust to congestion, since they can get some
> funding in the process, but they have a different failure 
> mode: lawsuit.
> Since they keep some records, the records can be subpoenaed, and the
> provider can be forced to disclose the identity of the 
> sending party. In
> fact, in many codes, if the providers refuse to do so, they may be
> deemed an accomplice of whatever action was carried out in 
> the anonymous
> exchange. Many providers have chosen to close their doors rather than
> have to defend more lawsuit -- often after a brush with the Church of
> Scientology.
> 
> If we really want practical anonymity, it has to be based on 
> the UA, and
> it should not rely on the help of a third party. We should use a
> combination of encryption, pre-paid subscriptions, and anonymous IP
> addresses (e.g. IPv6 privacy addresses.)
> 
> -- Christian Huitema
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<P><FONT SIZE=3D2>Christian,</FONT>
</P>

<P><FONT SIZE=3D2>If you want to provide complete unassailable =
anonymity, I agree completely. I have been arguing about providing a =
similar level of anonymity as the PSTN, where the called party is not =
informed who the call is from by the service itself. It's more about =
Data Protection than strict anonymity.</FONT></P>

<P><FONT SIZE=3D2>In the PSTN there is no problem for certain =
authorised organisations (including the courts) to obtain the details =
of the call from the Service Providers if they have good reason. Data =
Protection does not protect you against these organisations.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Christian Huitema [<A =
HREF=3D"mailto:huitema@windows.microsoft.com">mailto:huitema@windows.mic=
rosoft.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 15 April 2002 17:10</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I have heard many contributors assert that to =
get real anonymity, one</FONT>
<BR><FONT SIZE=3D2>&gt; will have to use a relay, e.g. an =
&quot;anonymizing B2BUA&quot;. I would like to</FONT>
<BR><FONT SIZE=3D2>&gt; point out a failure in this line of reasoning: =
anonymizing relays have</FONT>
<BR><FONT SIZE=3D2>&gt; been tried before, for e-mail, and the =
experience was absolutely not</FONT>
<BR><FONT SIZE=3D2>&gt; conclusive. In fact, two different kind of =
anonymizers have </FONT>
<BR><FONT SIZE=3D2>&gt; been tried:</FONT>
<BR><FONT SIZE=3D2>&gt; free ones, that kept absolutely no record, and =
subscriber-based, which</FONT>
<BR><FONT SIZE=3D2>&gt; kept some records. Both have failed, for =
different reason.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The failure mode of a completely free relay is =
basically spam, and</FONT>
<BR><FONT SIZE=3D2>&gt; similar sort of abuse. There is documentation =
available on </FONT>
<BR><FONT SIZE=3D2>&gt; the fate for</FONT>
<BR><FONT SIZE=3D2>&gt; example of the cypherpunk experiment: this set =
of relays was set up to</FONT>
<BR><FONT SIZE=3D2>&gt; provide true anonymity on the Internet, using a =
sophisticated mix of</FONT>
<BR><FONT SIZE=3D2>&gt; random redirection and cryptography. In theory, =
the design was</FONT>
<BR><FONT SIZE=3D2>&gt; excellent. In practice, the system was quickly =
overwhelmed by porn</FONT>
<BR><FONT SIZE=3D2>&gt; traffic, and the relays dropped one by one out =
of the system because</FONT>
<BR><FONT SIZE=3D2>&gt; they could not bear the resulting =
congestion.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Subscriber based relays are relays that keep =
some state: they actually</FONT>
<BR><FONT SIZE=3D2>&gt; identify the sender, before washing out the =
identity in the outgoing</FONT>
<BR><FONT SIZE=3D2>&gt; message. They are more robust to congestion, =
since they can get some</FONT>
<BR><FONT SIZE=3D2>&gt; funding in the process, but they have a =
different failure </FONT>
<BR><FONT SIZE=3D2>&gt; mode: lawsuit.</FONT>
<BR><FONT SIZE=3D2>&gt; Since they keep some records, the records can =
be subpoenaed, and the</FONT>
<BR><FONT SIZE=3D2>&gt; provider can be forced to disclose the identity =
of the </FONT>
<BR><FONT SIZE=3D2>&gt; sending party. In</FONT>
<BR><FONT SIZE=3D2>&gt; fact, in many codes, if the providers refuse to =
do so, they may be</FONT>
<BR><FONT SIZE=3D2>&gt; deemed an accomplice of whatever action was =
carried out in </FONT>
<BR><FONT SIZE=3D2>&gt; the anonymous</FONT>
<BR><FONT SIZE=3D2>&gt; exchange. Many providers have chosen to close =
their doors rather than</FONT>
<BR><FONT SIZE=3D2>&gt; have to defend more lawsuit -- often after a =
brush with the Church of</FONT>
<BR><FONT SIZE=3D2>&gt; Scientology.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; If we really want practical anonymity, it has =
to be based on </FONT>
<BR><FONT SIZE=3D2>&gt; the UA, and</FONT>
<BR><FONT SIZE=3D2>&gt; it should not rely on the help of a third =
party. We should use a</FONT>
<BR><FONT SIZE=3D2>&gt; combination of encryption, pre-paid =
subscriptions, and anonymous IP</FONT>
<BR><FONT SIZE=3D2>&gt; addresses (e.g. IPv6 privacy addresses.)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- Christian Huitema</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@optimus.ietf.org  Mon Apr 15 13:06:42 2002
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Date: Mon, 15 Apr 2002 19:16:55 +0300
To: "Ben Campbell" <bcampbell@dynamicsoft.com>
Cc: "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
In-Reply-To: <HNEOJECGFHIABDLENMMCOEMJCFAA.bcampbell@dynamicsoft.com>
References: <CD110021698980419241042CF576B8F2012BAB7B@EXCALIBUR.santera.com>
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Ben Campbell writes:

 > Expecting the network to screen out ID that _I_ provide is a slippery slope.
 > What if I put "call from Ben" in a Subject header? Do you screen that out as
 > well? What if I _tell_ the destination user who I am in an IM associated
 > with the call? What about the media session,?

i don't see any slippery slope here.  if you tell in the data part of
the message or call, who you are, then it is up to you.  the only thing
i care is that if the user in the from header claims to be someone that
someone has to be someone we have agreed for the subscription.

-- juha

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From daemon@optimus.ietf.org  Mon Apr 15 13:08:09 2002
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Date: Mon, 15 Apr 2002 09:35:34 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: sip@ietf.org
Message-ID: <Pine.WNT.4.44.0204150932200.-430659@chorizo.rapidconvergence.com>
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Subject: [Sip] rooms for interim
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Hi,

Rooms during the Interim can be reserved here:

http://www.pkghlrss.com/liveres/res.asp?EventCode=nilvgen0501&action=new

thanks,
-rohan



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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 11:39:33 -0500
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MessageYes, it is a long thread :-)

I pretty much agree with your original assessment for when the network
authenticates From, although I would propose a better behavior would be to
just disallow a request with a non-anonymous From header _and_ with privacy
requested. This approach is more backwards compatible (in case something
involved still freaks out about modified From headers), and a lot simpler to
implement in the network (no B2BUA required, at least for _this_ reason)

But in general, the requirement that a network must prevent a badly behaved
client from revealing its identity accidentally seems a bit excessive. It
should be sufficient to say that a client _can_ prevent its identity from
being revealed.
  -----Original Message-----
  From: Mark Watson [mailto:mwatson@nortelnetworks.com]
  Sent: Monday, April 15, 2002 11:17 AM
  To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
  Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Ben,

  I know this is a long thread, but I have addressed all these points
before. Please see below...
    -----Original Message-----
    From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
    Sent: 15 April 2002 15:44
    To: Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
    Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


    You pretty much have to assume that the receiving user sees _anything_
you send his UA, except in the rare case where the user has no control over
the UA behavior. Asking a UAS to hide information from its owner is not
useful in the generic case.

    I agree with Dean on this in general, although it really doesn't matter
if any _network_ inserted identification is there with the consent of the
originating user or not. A privacy request need _only_ apply to
network-inserted identification data. The UAC has every opportunity _not_ to
insert identification if it does not want to.
  It doesn't matter what device inserts the information. It matters whether
it is part of the service for the information to be inserted. If it is a
part of the service, then the responsibility lies with the Service
Provider - they defined the service and they are responsible for things the
users do effectively at their behest.

  If I check the From: field I receive from my users is valid, then I am
forcing them to put either their identity or 'Anonymous' in there.

  If there is then a privacy requirement arising from the rights of ANOTHER
PERSON, NOT THE USER, then I have to do something about this - I as the
Service Provider have the responsibility, not the user. An example is when
the *subscriber* has requested anonymity.

  If I let my users use any old From value, then the situation is not so
clear, but I still argue that for fields whose intention is defined for
transport of a particular identity, then the Service Provider has some
responsibility. (cf Special Arrangements in the ISDN).
    Expecting the network to screen out ID that _I_ provide is a slippery
slope. What if I put "call from Ben" in a Subject header? Do you screen that
out as well? What if I _tell_ the destination user who I am in an IM
associated with the call? What about the media session,?

  Personally, I don't think screening out the Subject would be necessary,
although there might be some market demand for services which did this.
Subject is a genuinely transparent field. There is no intention that it
contain your identity, and UAs are unlikely to insert your identity here
automatically.

  ...Mark

     -----Original Message-----
    From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chiou,
Mark
    Sent: Monday, April 15, 2002 9:28 AM
    To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
    Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


      Dean wrote:
      #####################################
      The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch!

      The calling-party-identity presentation infromation (CLIP or RPID) is
NOT inserted by the user. It is inserted by the network, WITHOUT THE CONSENT
of the user. As such, the operator is responsible for the protection of that
information in a manner consistent with regulation and the expressed wishes
of the user.
      <snip>
       ##########################################################
      No, The CLIP is inserted by the network WITH THE CONSENT of the user.
When you subscribe a line from a PSTN service provider, the form you filled
indicating whether you want your telephone number to be PUBLIC or PRIVATE.
(of course, the default is PUBLIC if you leave it blank.) When you make a
call, the PRIVATE or PUBLIC status value is retrieved and set the Address
Presentation Restricted Indicator (APRI) field in SS7 environment (Note
that, CLIR toggles the values), then the terminating side of the network
based on the APRI value to determine whether the Calling Party Number shall
be delivered to the callee's CPE.

      Since we want to push the decision making to the UA, we should
standardize the rule such that the UA shall display the From information
based on Privacy indication. If we think that too many UAs won't follow this
rule, than we should allow the Service Provider of terminating side (not the
originating side) to alter the From information before terminating the call
to the UA based on the Privacy indication.

      This is my penny thought,

      Mark Chiou

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<BODY>
<DIV><SPAN class=3D489291916-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>Yes,=20
it is a long thread :-)</FONT></SPAN></DIV>
<DIV><SPAN class=3D489291916-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D489291916-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
pretty much agree with your original assessment for when the network=20
authenticates From, although I would propose a better behavior would be =
to just=20
disallow a request with a non-anonymous From header _and_ with privacy=20
requested. This approach is more backwards compatible (in case something =

involved still freaks out about modified From headers), and a lot =
simpler to=20
implement in the network (no B2BUA required, at least for _this_=20
reason)</FONT></SPAN></DIV>
<DIV><SPAN class=3D489291916-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D489291916-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>But in=20
general, the requirement that a network must prevent a badly behaved =
client from=20
revealing its identity accidentally seems a bit excessive. It should be=20
sufficient to say that a client _can_ prevent its identity from being=20
revealed.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April 15, =
2002=20
  11:17 AM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis;=20
  Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: =
[Sip]=20
  Comment, SIP Privacy draft<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D404191416-15042002>Ben,</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D404191416-15042002>I=20
  know this is a long thread, but I have addressed all these points =
before.=20
  Please see below...</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Ben Campbell=20
    [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002=20
    15:44<BR><B>To:</B> Chiou, Mark; Dean Willis; Mpierce1@aol.com;=20
    sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, =
SIP=20
    Privacy draft<BR><BR></DIV></FONT>
    <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>You pretty much have to assume that the receiving user sees =

    _anything_ you send his UA, except in the rare case where the user =
has no=20
    control over the UA behavior. Asking a UAS to hide information from =
its=20
    owner is not useful in the generic case.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
    agree with Dean on this in general, although it really doesn't =
matter if any=20
    _network_ inserted identification is there with the consent of the=20
    originating user or not. A privacy request need _only_ apply to=20
    network-inserted identification data. The UAC has every opportunity =
_not_ to=20
    insert identification if it does not want to.=20
</FONT></SPAN></DIV></BLOCKQUOTE>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D404191416-15042002>It=20
  doesn't matter what device inserts the information. It matters whether =
it is=20
  part of the service for the information to be inserted. If it is a =
part of the=20
  service, then the responsibility lies with the Service Provider - they =
defined=20
  the service and they are responsible for things the users do =
effectively at=20
  their behest.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D404191416-15042002>If=20
  I check the From: field I receive from my users is valid, then I am =
forcing=20
  them to put either their identity or 'Anonymous' in =
there.</SPAN></FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D404191416-15042002>If=20
  there is then a privacy requirement arising from the rights of ANOTHER =
PERSON,=20
  NOT THE USER, then I have to do something about this - I as the =
Service=20
  Provider have the responsibility, not the user. An example is when the =

  *subscriber* has requested anonymity.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D404191416-15042002>If=20
  I let my users use any old From value, then the situation is not so =
clear, but=20
  I still argue that for fields whose intention is defined for transport =
of a=20
  particular identity, then the Service Provider has some =
responsibility. (cf=20
  Special Arrangements in the ISDN).</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
    size=3D2>Expecting the network to screen out ID that _I_ provide is =
a slippery=20
    slope. What if I put "call from Ben" in a Subject header? Do you =
screen that=20
    out as well? What if I _tell_ the destination user who I am in an IM =

    associated with the call? What about the media session,?<FONT =
face=3DVerdana=20
    size=3D1><SPAN=20
    =
class=3D404191416-15042002>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></DIV=
>
    <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
    size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
    =
class=3D404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV=
></BLOCKQUOTE>
  <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN =
class=3D404191416-15042002>Personally, I=20
  don't think screening out the Subject would be necessary, although =
there might=20
  be some market demand for services which did this.&nbsp;Subject is a =
genuinely=20
  transparent field. There is no intention that it contain your =
identity, and=20
  UAs are unlikely to insert your identity here=20
  automatically.</SPAN></FONT></FONT></SPAN></FONT></DIV>
  <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  =
class=3D404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV=
>
  <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  =
class=3D404191416-15042002>...Mark</SPAN></FONT></FONT></SPAN></FONT></DI=
V>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV><SPAN class=3D435513614-15042002></SPAN><FONT =
face=3DTahoma><FONT=20
    size=3D2><SPAN class=3D435513614-15042002><FONT face=3DArial=20
    color=3D#0000ff></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DTahoma><FONT size=3D2><SPAN=20
    class=3D435513614-15042002>&nbsp;</SPAN>-----Original=20
    Message-----<BR><B>From:</B> sip-admin@ietf.org=20
    [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou, =
Mark<BR><B>Sent:</B>=20
    Monday, April 15, 2002 9:28 AM<BR><B>To:</B> Dean Willis; =
Mpierce1@aol.com;=20
    sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, =
SIP=20
    Privacy draft<BR><BR></DIV></FONT></FONT>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
      class=3D765415513-15042002>Dean wrote:</SPAN></FONT></DIV>
      <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
      =
class=3D765415513-15042002>#####################################</SPAN></=
FONT></DIV>
      <DIV><FONT color=3D#0000ff size=3D2><SPAN =
class=3D765415513-15042002>
      <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
      size=3D+0><FONT size=3D2><SPAN class=3D819102015-12042002><FONT =
face=3DArial=20
      color=3D#0000ff>The point I'm trying to make is that To and From =
are USER=20
      supplied information. The&nbsp;Service Provider doesn't insert =
them,=20
      inspect them, or have any responsibility for them. Insisting that =
the=20
      service provider change them is just like&nbsp;insisting that the =
service=20
      provider change any other user-supplied information,&nbsp;such as=20
      the&nbsp;words I'm typing now. Anything I put in my To or from =
field is=20
      deliberately and explicitly intended to be delivered to the =
terminating=20
      side of the dialog, and if as a service provider you FAIL to do =
that, then=20
      you have DAMAGED my data, and my lawyers will be in=20
      touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
      <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
      size=3D+0><FONT color=3D#0000ff size=3D2><SPAN=20
      =
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT=20
      face=3DArial></FONT>&nbsp;</DIV>
      <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
      size=3D+0><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
      class=3D819102015-12042002>The calling-party-identity presentation =

      infromation (CLIP or RPID) is NOT inserted by the user. It is =
inserted by=20
      the network, WITHOUT THE CONSENT of the user. As such, the =
operator is=20
      responsible for the protection of that information in a manner =
consistent=20
      with regulation and the expressed wishes of the=20
      user.</SPAN></FONT></FONT></FONT></FONT></DIV>
      <DIV><FONT lang=3D0 FAMILY=3D"SANSSERIF"><FONT face=3DArial><SPAN=20
      class=3D819102015-12042002><SPAN=20
      =
class=3D765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>=

      <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT=20
      color=3D#0000ff><SPAN =
class=3D819102015-12042002></SPAN></FONT></FONT><FONT=20
      face=3DArial>&nbsp;<SPAN=20
      =
class=3D765415513-15042002>##############################################=
############</SPAN></FONT></DIV>
      <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>No, The =
CLIP is=20
      inserted by the network WITH THE CONSENT of the user. When you =
subscribe a=20
      line from a PSTN service provider, the form you filled indicating =
whether=20
      you want your telephone number to be PUBLIC or PRIVATE. (of =
course, the=20
      default is PUBLIC if you leave it blank.) When you make a call, =
the=20
      PRIVATE or PUBLIC status value is retrieved and set the Address=20
      Presentation Restricted Indicator (APRI) field in SS7 environment =
(Note=20
      that, CLIR toggles the values), then the terminating side of the =
network=20
      based on the APRI value to determine whether the Calling Party =
Number=20
      shall be delivered to the callee's CPE.</SPAN></FONT></DIV>
      <DIV><FONT face=3DArial><SPAN=20
      class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>Since we =
want to push=20
      the decision making to the UA, we should standardize the rule such =
that=20
      the UA shall display the From information based on Privacy =
indication. If=20
      we think that too many UAs won't follow this rule, than we should =
allow=20
      the Service Provider of terminating side (not the originating =
side) to=20
      alter the From information before terminating the call to the UA =
based on=20
      the Privacy indication.</SPAN></FONT></DIV>
      <DIV><FONT face=3DArial><SPAN=20
      class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT=20
      face=3DArial><SPAN class=3D765415513-15042002>This is my penny=20
      thought,</SPAN></FONT></SPAN></FONT></DIV>
      <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT=20
      face=3DArial><SPAN=20
      =
class=3D765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT=20
      face=3DArial><SPAN class=3D765415513-15042002>Mark=20
      =
Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQ=
UOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Mon Apr 15 13:15:50 2002
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Message-ID: <A3C2399B2FACD411A54200508BE39C74054F7136@zwcwd00r.europe.nortel.com>
From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 17:50:13 +0100
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Rejecting non-anonymous requests when anonymity is required is all very
well, but a 2543 client will not know what to do in order to get its call
through. It will require user intervention. For every call. How will the
user know when anonymity is required ? The user does not want to be forced
into anonymity unless absolutely necessary, so they will always try
non-anonymous first.
 
It's not a question of 'badly behaved clients'. The client is well-behaved
and following the users instructions to be 'non-anonymous' - sure, the
client can prevent its identity being revealed, but the user does not want
this. The SUBSCRIBER wants anonymity, not the user.
 
You could standardise a way of communicating the subscriber's wishes to the
UA, and then enforce them. But how does this work for a forwarding user (or
their subscriber) who does not wish to be identified ?
 
...Mark

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: 15 April 2002 17:40
To: Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis;
Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Yes, it is a long thread :-)
 
I pretty much agree with your original assessment for when the network
authenticates From, although I would propose a better behavior would be to
just disallow a request with a non-anonymous From header _and_ with privacy
requested. This approach is more backwards compatible (in case something
involved still freaks out about modified From headers), and a lot simpler to
implement in the network (no B2BUA required, at least for _this_ reason)
 
But in general, the requirement that a network must prevent a badly behaved
client from revealing its identity accidentally seems a bit excessive. It
should be sufficient to say that a client _can_ prevent its identity from
being revealed.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Monday, April 15, 2002 11:17 AM
To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Ben,
 
I know this is a long thread, but I have addressed all these points before.
Please see below...

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: 15 April 2002 15:44
To: Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


You pretty much have to assume that the receiving user sees _anything_ you
send his UA, except in the rare case where the user has no control over the
UA behavior. Asking a UAS to hide information from its owner is not useful
in the generic case.
 
I agree with Dean on this in general, although it really doesn't matter if
any _network_ inserted identification is there with the consent of the
originating user or not. A privacy request need _only_ apply to
network-inserted identification data. The UAC has every opportunity _not_ to
insert identification if it does not want to. 

It doesn't matter what device inserts the information. It matters whether it
is part of the service for the information to be inserted. If it is a part
of the service, then the responsibility lies with the Service Provider -
they defined the service and they are responsible for things the users do
effectively at their behest.
 
If I check the From: field I receive from my users is valid, then I am
forcing them to put either their identity or 'Anonymous' in there.
 
If there is then a privacy requirement arising from the rights of ANOTHER
PERSON, NOT THE USER, then I have to do something about this - I as the
Service Provider have the responsibility, not the user. An example is when
the *subscriber* has requested anonymity.
 
If I let my users use any old From value, then the situation is not so
clear, but I still argue that for fields whose intention is defined for
transport of a particular identity, then the Service Provider has some
responsibility. (cf Special Arrangements in the ISDN).

Expecting the network to screen out ID that _I_ provide is a slippery slope.
What if I put "call from Ben" in a Subject header? Do you screen that out as
well? What if I _tell_ the destination user who I am in an IM associated
with the call? What about the media session,? 
 

Personally, I don't think screening out the Subject would be necessary,
although there might be some market demand for services which did this.
Subject is a genuinely transparent field. There is no intention that it
contain your identity, and UAs are unlikely to insert your identity here
automatically.
 
...Mark

 
 -----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Chiou, Mark
Sent: Monday, April 15, 2002 9:28 AM
To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft



Dean wrote:
#####################################

The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch!
 
The calling-party-identity presentation infromation (CLIP or RPID) is NOT
inserted by the user. It is inserted by the network, WITHOUT THE CONSENT of
the user. As such, the operator is responsible for the protection of that
information in a manner consistent with regulation and the expressed wishes
of the user.
<snip>
 ##########################################################
No, The CLIP is inserted by the network WITH THE CONSENT of the user. When
you subscribe a line from a PSTN service provider, the form you filled
indicating whether you want your telephone number to be PUBLIC or PRIVATE.
(of course, the default is PUBLIC if you leave it blank.) When you make a
call, the PRIVATE or PUBLIC status value is retrieved and set the Address
Presentation Restricted Indicator (APRI) field in SS7 environment (Note
that, CLIR toggles the values), then the terminating side of the network
based on the APRI value to determine whether the Calling Party Number shall
be delivered to the callee's CPE.
 
Since we want to push the decision making to the UA, we should standardize
the rule such that the UA shall display the From information based on
Privacy indication. If we think that too many UAs won't follow this rule,
than we should allow the Service Provider of terminating side (not the
originating side) to alter the From information before terminating the call
to the UA based on the Privacy indication.
 
This is my penny thought,
 
Mark Chiou 


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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=755424716-15042002>Rejecting non-anonymous requests when anonymity is 
required is all very well, but a 2543 client will not know what to do in order 
to get its call through. It will require user intervention. For every call. How 
will the user know when anonymity is required ? The user does not want to be 
forced into anonymity unless absolutely necessary, so they will always try 
non-anonymous first.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=755424716-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=755424716-15042002>It's 
not a question of 'badly behaved clients'. The client is well-behaved and 
following the users instructions to be 'non-anonymous' - sure, the client can 
prevent its identity being revealed, but the user does not want this. The 
SUBSCRIBER wants anonymity, not the user.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=755424716-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=755424716-15042002>You 
could standardise a way of communicating the subscriber's wishes to the UA, and 
then enforce them. But how does this work for a forwarding user (or their 
subscriber) who does not wish to be identified ?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=755424716-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=755424716-15042002>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ben Campbell 
  [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002 
  17:40<BR><B>To:</B> Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis; 
  Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] 
  Comment, SIP Privacy draft<BR><BR></DIV></FONT>
  <DIV><SPAN class=489291916-15042002><FONT color=#0000ff face=Arial size=2>Yes, 
  it is a long thread :-)</FONT></SPAN></DIV>
  <DIV><SPAN class=489291916-15042002><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=489291916-15042002><FONT color=#0000ff face=Arial size=2>I 
  pretty much agree with your original assessment for when the network 
  authenticates From, although I would propose a better behavior would be to 
  just disallow a request with a non-anonymous From header _and_ with privacy 
  requested. This approach is more backwards compatible (in case something 
  involved still freaks out about modified From headers), and a lot simpler to 
  implement in the network (no B2BUA required, at least for _this_ 
  reason)</FONT></SPAN></DIV>
  <DIV><SPAN class=489291916-15042002><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=489291916-15042002><FONT color=#0000ff face=Arial size=2>But 
  in general, the requirement that a network must prevent a badly behaved client 
  from revealing its identity accidentally seems a bit excessive. It should be 
  sufficient to say that a client _can_ prevent its identity from being 
  revealed.</FONT></SPAN></DIV>
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
    size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
    [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April 15, 2002 
    11:17 AM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis; 
    Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] 
    Comment, SIP Privacy draft<BR><BR></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>Ben,</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>I know this is a long thread, but I have addressed 
    all these points before. Please see below...</SPAN></FONT></DIV>
    <BLOCKQUOTE dir=ltr 
    style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
      <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
      size=2>-----Original Message-----<BR><B>From:</B> Ben Campbell 
      [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002 
      15:44<BR><B>To:</B> Chiou, Mark; Dean Willis; Mpierce1@aol.com; 
      sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP 
      Privacy draft<BR><BR></DIV></FONT>
      <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial 
      size=2>You pretty much have to assume that the receiving user sees 
      _anything_ you send his UA, except in the rare case where the user has no 
      control over the UA behavior. Asking a UAS to hide information from its 
      owner is not useful in the generic case.</FONT></SPAN></DIV>
      <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial 
      size=2></FONT></SPAN>&nbsp;</DIV>
      <DIV><SPAN class=435513614-15042002><FONT color=#0000ff face=Arial 
      size=2>I agree with Dean on this in general, although it really doesn't 
      matter if any _network_ inserted identification is there with the consent 
      of the originating user or not. A privacy request need _only_ apply to 
      network-inserted identification data. The UAC has every opportunity _not_ 
      to insert identification if it does not want to. 
    </FONT></SPAN></DIV></BLOCKQUOTE>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>It doesn't matter what device inserts the 
    information. It matters whether it is part of the service for the 
    information to be inserted. If it is a part of the service, then the 
    responsibility lies with the Service Provider - they defined the service and 
    they are responsible for things the users do effectively at their 
    behest.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>If I check the From: field I receive from my users 
    is valid, then I am forcing them to put either their identity or 'Anonymous' 
    in there.</SPAN></FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>If there is then a privacy requirement arising from 
    the rights of ANOTHER PERSON, NOT THE USER, then I have to do something 
    about this - I as the Service Provider have the responsibility, not the 
    user. An example is when the *subscriber* has requested 
    anonymity.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=404191416-15042002>If I let my users use any old From value, then the 
    situation is not so clear, but I still argue that for fields whose intention 
    is defined for transport of a particular identity, then the Service Provider 
    has some responsibility. (cf Special Arrangements in the 
    ISDN).</SPAN></FONT></DIV>
    <BLOCKQUOTE dir=ltr 
    style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
      <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
      size=2>Expecting the network to screen out ID that _I_ provide is a 
      slippery slope. What if I put "call from Ben" in a Subject header? Do you 
      screen that out as well? What if I _tell_ the destination user who I am in 
      an IM associated with the call? What about the media session,?<FONT 
      face=Verdana size=1><SPAN 
      class=404191416-15042002>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
      size=2><FONT face=Verdana size=1><SPAN 
      class=404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV></BLOCKQUOTE>
    <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
    size=2><FONT face=Verdana size=1><SPAN class=404191416-15042002>Personally, 
    I don't think screening out the Subject would be necessary, although there 
    might be some market demand for services which did this.&nbsp;Subject is a 
    genuinely transparent field. There is no intention that it contain your 
    identity, and UAs are unlikely to insert your identity here 
    automatically.</SPAN></FONT></FONT></SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
    size=2><FONT face=Verdana size=1><SPAN 
    class=404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff><SPAN class=435513614-15042002><FONT face=Arial 
    size=2><FONT face=Verdana size=1><SPAN 
    class=404191416-15042002>...Mark</SPAN></FONT></FONT></SPAN></FONT></DIV>
    <BLOCKQUOTE dir=ltr 
    style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
      <DIV><SPAN class=435513614-15042002></SPAN><FONT face=Tahoma><FONT 
      size=2><SPAN class=435513614-15042002><FONT color=#0000ff 
      face=Arial></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
      <DIV><FONT face=Tahoma><FONT size=2><SPAN 
      class=435513614-15042002>&nbsp;</SPAN>-----Original 
      Message-----<BR><B>From:</B> sip-admin@ietf.org 
      [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou, Mark<BR><B>Sent:</B> 
      Monday, April 15, 2002 9:28 AM<BR><B>To:</B> Dean Willis; 
      Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] 
      Comment, SIP Privacy draft<BR><BR></DIV></FONT></FONT>
      <BLOCKQUOTE dir=ltr 
      style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
        <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
        class=765415513-15042002>Dean wrote:</SPAN></FONT></DIV>
        <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
        class=765415513-15042002>#####################################</SPAN></FONT></DIV>
        <DIV><FONT color=#0000ff size=2><SPAN class=765415513-15042002>
        <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
        size=+0><FONT size=2><SPAN class=819102015-12042002><FONT color=#0000ff 
        face=Arial>The point I'm trying to make is that To and From are USER 
        supplied information. The&nbsp;Service Provider doesn't insert them, 
        inspect them, or have any responsibility for them. Insisting that the 
        service provider change them is just like&nbsp;insisting that the 
        service provider change any other user-supplied information,&nbsp;such 
        as the&nbsp;words I'm typing now. Anything I put in my To or from field 
        is deliberately and explicitly intended to be delivered to the 
        terminating side of the dialog, and if as a service provider you FAIL to 
        do that, then you have DAMAGED my data, and my lawyers will be in 
        touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
        <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
        size=+0><FONT color=#0000ff size=2><SPAN 
        class=819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT 
        face=Arial></FONT>&nbsp;</DIV>
        <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT size=+0><FONT 
        size=+0><FONT color=#0000ff face=Arial size=2><SPAN 
        class=819102015-12042002>The calling-party-identity presentation 
        infromation (CLIP or RPID) is NOT inserted by the user. It is inserted 
        by the network, WITHOUT THE CONSENT of the user. As such, the operator 
        is responsible for the protection of that information in a manner 
        consistent with regulation and the expressed wishes of the 
        user.</SPAN></FONT></FONT></FONT></FONT></DIV>
        <DIV><FONT lang=0 FAMILY="SANSSERIF"><FONT face=Arial><SPAN 
        class=819102015-12042002><SPAN 
        class=765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>
        <DIV><FONT color=#000000 lang=0 FAMILY="SANSSERIF"><FONT 
        color=#0000ff><SPAN class=819102015-12042002></SPAN></FONT></FONT><FONT 
        face=Arial>&nbsp;<SPAN 
        class=765415513-15042002>##########################################################</SPAN></FONT></DIV>
        <DIV><FONT face=Arial><SPAN class=765415513-15042002>No, The CLIP is 
        inserted by the network WITH THE CONSENT of the user. When you subscribe 
        a line from a PSTN service provider, the form you filled indicating 
        whether you want your telephone number to be PUBLIC or PRIVATE. (of 
        course, the default is PUBLIC if you leave it blank.) When you make a 
        call, the PRIVATE or PUBLIC status value is retrieved and set the 
        Address Presentation Restricted Indicator (APRI) field in SS7 
        environment (Note that, CLIR toggles the values), then the terminating 
        side of the network based on the APRI value to determine whether the 
        Calling Party Number shall be delivered to the callee's 
        CPE.</SPAN></FONT></DIV>
        <DIV><FONT face=Arial><SPAN 
        class=765415513-15042002></SPAN></FONT>&nbsp;</DIV>
        <DIV><FONT face=Arial><SPAN class=765415513-15042002>Since we want to 
        push the decision making to the UA, we should standardize the rule such 
        that the UA shall display the From information based on Privacy 
        indication. If we think that too many UAs won't follow this rule, than 
        we should allow the Service Provider of terminating side (not the 
        originating side) to alter the From information before terminating the 
        call to the UA based on the Privacy indication.</SPAN></FONT></DIV>
        <DIV><FONT face=Arial><SPAN 
        class=765415513-15042002></SPAN></FONT>&nbsp;</DIV>
        <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT 
        face=Arial><SPAN class=765415513-15042002>This is my penny 
        thought,</SPAN></FONT></SPAN></FONT></DIV>
        <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT 
        face=Arial><SPAN 
        class=765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
        <DIV><FONT face=Arial><SPAN class=765415513-15042002><FONT 
        face=Arial><SPAN class=765415513-15042002>Mark 
        Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Date: Mon, 15 Apr 2002 12:49:43 EDT
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/15/02 10:45:08 AM Eastern Daylight Time, 
bcampbell@dynamicsoft.com writes:


> You pretty much have to assume that the receiving user sees _anything_ you 
> send his UA, except in the rare case where the user has no control over the 
> UA behavior. Asking a UAS to hide information from its owner is not useful 
> in the generic case.

I don't believe anyone has suggested (I hope) that the receiving user's UA 
should be responsible for hiding the information. It is always the last 
"trusted" entity which must do this, i.e., the "service provider" in Mark 
Chiou's comments. The exception, of course, might be when the terminating UA 
is also "trusted" by the service provider, but I'm not sure of a good example 
of this. The terminating UA would then be like an extension of the "Service 
Provider".

 
> I agree with Dean on this in general, although it really doesn't matter if 
> any _network_ inserted identification is there with the consent of the 
> originating user or not. A privacy request need _only_ apply to 
> network-inserted identification data. The UAC has every opportunity _not_ 
> 

I believe that every identification inserted must be "with consent" of the 
originating user, either explicitly by user action, or implicitly by the fact 
that they made a phone call. And no, a privacy request does not only apply to 
network-inserted identifcation data. (Compare to the ISDN case in which the 
originating user inserts their phone number and also indicates that it is to 
be kept private, i.e., not delivered to the other user. The network verifies 
it and passes it through their trusted network or to another trusted network. 
The requirement in SIP is not different.)

Just to confuse the issue a little more, maybe I should remind everyone at 
this point, that, at least in the US, when the originator says their identity 
should be kept private, it is still delivered at the other end for an "800" 
call. Also, some years ago, Florida created special office codes to override 
the originator's privacy desire when calls terminated to those special office 
codes. I don't know if that stupid capability still exists in Florida.

And no, the UAC does not have the opportunity to not insert an identification 
if the "Service Provider" requires it as a prerequisite for service.
 
> Expecting the network to screen out ID that _I_ provide is a slippery slope. 
> What if I put "call from Ben" in a Subject header? Do you screen that out 
> as well? What if I _tell_ the destination user who I am in an IM associated 
> with the call? What about the media session,?
>  
> 
Providing an ID in a field defined for that prupose (for the benefit of the 
network and required trace) and have the network "screen" it is not a 
slippary slope. It is exactly what must be done and is straight forward.

I hope we have agreed across the board that the network would never screen 
the contents of other fields such as the Subject header.

Mike



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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/15/02 10:45:08 AM Eastern Daylight Time, bcampbell@dynamicsoft.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">You pretty much have to assume that the receiving user sees _anything_ you send his UA, except in the rare case where the user has no control over the UA behavior. Asking a UAS to hide information from its owner is not useful in the generic case.</FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I don't believe anyone has suggested (I hope) that the receiving user's UA should be responsible for hiding the information. It is always the last "trusted" entity which must do this, i.e., the "service provider" in Mark Chiou's comments. The exception, of course, might be when the terminating UA is also "trusted" by the service provider, but I'm not sure of a good example of this. The terminating UA would then be like an extension of the "Service Provider".
<BR>
<BR> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I agree with Dean on this in general, although it really doesn't matter if any _network_ inserted identification is there with the consent of the originating user or not. A privacy request need _only_ apply to network-inserted identification data. The UAC has every opportunity _not_ to insert identification if it does not want to. </BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I believe that every identification inserted must be "with consent" of the originating user, either explicitly by user action, or implicitly by the fact that they made a phone call. And no, a privacy request does not only apply to network-inserted identifcation data. (Compare to the ISDN case in which the originating user inserts their phone number and also indicates that it is to be kept private, i.e., not delivered to the other user. The network verifies it and passes it through their trusted network or to another trusted network. The requirement in SIP is not different.)
<BR>
<BR>Just to confuse the issue a little more, maybe I should remind everyone at this point, that, at least in the US, when the originator says their identity should be kept private, it is still delivered at the other end for an "800" call. Also, some years ago, Florida created special office codes to override the originator's privacy desire when calls terminated to those special office codes. I don't know if that stupid capability still exists in Florida.
<BR>
<BR>And no, the UAC does not have the opportunity to not insert an identification if the "Service Provider" requires it as a prerequisite for service.
<BR> 
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Expecting the network to screen out ID that _I_ provide is a slippery slope. What if I put "call from Ben" in a Subject header? Do you screen that out as well? What if I _tell_ the destination user who I am in an IM associated with the call? What about the media session,?</FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></FONT><FONT  COLOR="#0000ff" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0"> </FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR></BLOCKQUOTE>
<BR>Providing an ID in a field defined for that prupose (for the benefit of the network and required trace) and have the network "screen" it is not a slippary slope. It is exactly what must be done and is straight forward.
<BR>
<BR>I hope we have agreed across the board that the network would never screen the contents of other fields such as the Subject header.
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/12/02 4:07:34 PM Eastern Daylight Time, 
hgs@cs.columbia.edu writes:


> 
> I don't think anticipating regulation is particularly helpful here.
> Anybody can say anything about what some legislature somewhere may
> legislate at some point in the future.
> 

I wasn't speaking of "anticipating regulation" but rather current legislation 
that will eventually applied to SIP-based telephony whenever someone wants to 
sell a regular telephony service using SIP in place of what is currently 
provided over circuit switched technology. It will take a very simple action 
by a state utility commission to declare that its rules for circuit switched 
also must apply to packet-based telephony. It would be foolish to ignore this 
eventuality.


> I wouldn't be too surprised if, instead of PSTN legislation, some of the
> email spam legislation finds its way into SIP behavior, e.g., having to
> declare a truthful return address, an indication of the call intent
> (like some of the laws and bills calling for adding ADV to the subject
> line) or including an opt-out link (header?). If we could make progress
> there, I think we'd do more for "privacy" than hiding a user name.
> 
> 

Just like the law today for FAX (in the US) - every call must be identified 
with the source. Yes, I suspect that this will become required for SIP, not 
necessarily by law, but by any responsible service provider as a prerequisite 
for setting up a session.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/12/02 4:07:34 PM Eastern Daylight Time, hgs@cs.columbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>I don't think anticipating regulation is particularly helpful here.
<BR>Anybody can say anything about what some legislature somewhere may
<BR>legislate at some point in the future.
<BR></BLOCKQUOTE>
<BR>
<BR>I wasn't speaking of "anticipating regulation" but rather current legislation that will eventually applied to SIP-based telephony whenever someone wants to sell a regular telephony service using SIP in place of what is currently provided over circuit switched technology. It will take a very simple action by a state utility commission to declare that its rules for circuit switched also must apply to packet-based telephony. It would be foolish to ignore this eventuality.
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I wouldn't be too surprised if, instead of PSTN legislation, some of the
<BR>email spam legislation finds its way into SIP behavior, e.g., having to
<BR>declare a truthful return address, an indication of the call intent
<BR>(like some of the laws and bills calling for adding ADV to the subject
<BR>line) or including an opt-out link (header?). If we could make progress
<BR>there, I think we'd do more for "privacy" than hiding a user name.
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>Just like the law today for FAX (in the US) - every call must be identified with the source. Yes, I suspect that this will become required for SIP, not necessarily by law, but by any responsible service provider as a prerequisite for setting up a session.
<BR>
<BR>Mike
<BR></FONT></HTML>

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From: "David R. Oran" <oran@cisco.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 12:38:52 -0400
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Brian Rosen wrote:
>If we could get consensus that trace-with-anonymity is the problem,
then > we can
> work on solutions.  Until we get to something as straightforward as 
> that,
> we are going to spin a lot of wheels.

That's always been the prime motivator for me, plus the secondary use of
NAI for call return to an anonymous caller.

Dave


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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 12:08:34 -0500
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Message
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Mark
Watson
  Sent: Monday, April 15, 2002 11:50 AM
  To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
  Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Rejecting non-anonymous requests when anonymity is required is all very
well, but a 2543 client will not know what to do in order to get its call
through. It will require user intervention. For every call. How will the
user know when anonymity is required ? The user does not want to be forced
into anonymity unless absolutely necessary, so they will always try
non-anonymous first.

  I'm not sure how a 2543 client will know how to request privacy in the
first place. And I assume that a user knows whether or not he wishes to make
an anonymous call. I cannot think of a _technical_ solution for helping a
user decide if a call should be anonymous or not.

   It's not a question of 'badly behaved clients'. The client is
well-behaved and following the users instructions to be 'non-anonymous' -
sure, the client can prevent its identity being revealed, but the user does
not want this. The SUBSCRIBER wants anonymity, not the user.

  I'm not sure what you mean--who is the subscriber, if not the user (i.e.
the person trying to place an anonymous call)?  (This may be the source of
my confusion in the previous paragraph. I was using the term "user" and
"subscriber" interchangeably.)

  You could standardise a way of communicating the subscriber's wishes to
the UA, and then enforce them. But how does this work for a forwarding user
(or their subscriber) who does not wish to be identified ?

  Presumadly if a UA supports anonymous calling, it must give the subscriber
some way of expressing the desire to do so. I don't see any need to
standardize that. Once the subscriber expresses that desire, it is up to the
UA to do things right (like, don't _automatically_ put user identifiable
information in headers, From or otherwise.)  A UA that pretends to support
anonymity but screws it up fits my definition of "badly behaved."

  ...Mark
    -----Original Message-----
    From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
    Sent: 15 April 2002 17:40
    To: Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis;
Mpierce1@aol.com; sip@ietf.org
    Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


    Yes, it is a long thread :-)

    I pretty much agree with your original assessment for when the network
authenticates From, although I would propose a better behavior would be to
just disallow a request with a non-anonymous From header _and_ with privacy
requested. This approach is more backwards compatible (in case something
involved still freaks out about modified From headers), and a lot simpler to
implement in the network (no B2BUA required, at least for _this_ reason)

    But in general, the requirement that a network must prevent a badly
behaved client from revealing its identity accidentally seems a bit
excessive. It should be sufficient to say that a client _can_ prevent its
identity from being revealed.
      -----Original Message-----
      From: Mark Watson [mailto:mwatson@nortelnetworks.com]
      Sent: Monday, April 15, 2002 11:17 AM
      To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
      Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


      Ben,

      I know this is a long thread, but I have addressed all these points
before. Please see below...
        -----Original Message-----
        From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
        Sent: 15 April 2002 15:44
        To: Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
        Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


        You pretty much have to assume that the receiving user sees
_anything_ you send his UA, except in the rare case where the user has no
control over the UA behavior. Asking a UAS to hide information from its
owner is not useful in the generic case.

        I agree with Dean on this in general, although it really doesn't
matter if any _network_ inserted identification is there with the consent of
the originating user or not. A privacy request need _only_ apply to
network-inserted identification data. The UAC has every opportunity _not_ to
insert identification if it does not want to.
      It doesn't matter what device inserts the information. It matters
whether it is part of the service for the information to be inserted. If it
is a part of the service, then the responsibility lies with the Service
Provider - they defined the service and they are responsible for things the
users do effectively at their behest.

      If I check the From: field I receive from my users is valid, then I am
forcing them to put either their identity or 'Anonymous' in there.

      If there is then a privacy requirement arising from the rights of
ANOTHER PERSON, NOT THE USER, then I have to do something about this - I as
the Service Provider have the responsibility, not the user. An example is
when the *subscriber* has requested anonymity.

      If I let my users use any old From value, then the situation is not so
clear, but I still argue that for fields whose intention is defined for
transport of a particular identity, then the Service Provider has some
responsibility. (cf Special Arrangements in the ISDN).
        Expecting the network to screen out ID that _I_ provide is a
slippery slope. What if I put "call from Ben" in a Subject header? Do you
screen that out as well? What if I _tell_ the destination user who I am in
an IM associated with the call? What about the media session,?

      Personally, I don't think screening out the Subject would be
necessary, although there might be some market demand for services which did
this. Subject is a genuinely transparent field. There is no intention that
it contain your identity, and UAs are unlikely to insert your identity here
automatically.

      ...Mark

         -----Original Message-----
        From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
Chiou, Mark
        Sent: Monday, April 15, 2002 9:28 AM
        To: Dean Willis; Mpierce1@aol.com; sip@ietf.org
        Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


          Dean wrote:
          #####################################
          The point I'm trying to make is that To and From are USER supplied
information. The Service Provider doesn't insert them, inspect them, or have
any responsibility for them. Insisting that the service provider change them
is just like insisting that the service provider change any other
user-supplied information, such as the words I'm typing now. Anything I put
in my To or from field is deliberately and explicitly intended to be
delivered to the terminating side of the dialog, and if as a service
provider you FAIL to do that, then you have DAMAGED my data, and my lawyers
will be in touch!

          The calling-party-identity presentation infromation (CLIP or RPID)
is NOT inserted by the user. It is inserted by the network, WITHOUT THE
CONSENT of the user. As such, the operator is responsible for the protection
of that information in a manner consistent with regulation and the expressed
wishes of the user.
          <snip>
           ##########################################################
          No, The CLIP is inserted by the network WITH THE CONSENT of the
user. When you subscribe a line from a PSTN service provider, the form you
filled indicating whether you want your telephone number to be PUBLIC or
PRIVATE. (of course, the default is PUBLIC if you leave it blank.) When you
make a call, the PRIVATE or PUBLIC status value is retrieved and set the
Address Presentation Restricted Indicator (APRI) field in SS7 environment
(Note that, CLIR toggles the values), then the terminating side of the
network based on the APRI value to determine whether the Calling Party
Number shall be delivered to the callee's CPE.

          Since we want to push the decision making to the UA, we should
standardize the rule such that the UA shall display the From information
based on Privacy indication. If we think that too many UAs won't follow this
rule, than we should allow the Service Provider of terminating side (not the
originating side) to alter the From information before terminating the call
to the UA based on the Privacy indication.

          This is my penny thought,

          Mark Chiou

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<HTML><HEAD><TITLE>Message</TITLE>
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charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D064020117-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Mark =
Watson<BR><B>Sent:</B>=20
  Monday, April 15, 2002 11:50 AM<BR><B>To:</B> 'Ben Campbell'; Chiou, =
Mark;=20
  Dean Willis; Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: =
Summary of=20
  RE: [Sip] Comment, SIP Privacy draft<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D755424716-15042002>Rejecting non-anonymous requests when =
anonymity is=20
  required is all very well, but a 2543 client will not know what to do =
in order=20
  to get its call through. It will require user intervention. For every =
call.=20
  How will the user know when anonymity is required ? The user does not =
want to=20
  be forced into anonymity unless absolutely necessary, so they will =
always try=20
  non-anonymous first.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D755424716-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D064020117-15042002><FONT face=3DArial =
size=3D2>I'm not=20
  sure how a 2543 client will know how to request privacy in the first =
place.=20
  And I assume that a user knows whether or not he wishes to make an =
anonymous=20
  call. I cannot think of&nbsp;a _technical_ solution for helping a user =
decide=20
  if a call should be anonymous or=20
  not.&nbsp;&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D064020117-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D064020117-15042002>&nbsp;</SPAN>It's not =
a question=20
  of 'badly behaved clients'. The client is well-behaved and following =
the users=20
  instructions to be 'non-anonymous' - sure, the client can prevent its =
identity=20
  being revealed, but the user does not want this. The SUBSCRIBER wants=20
  anonymity, not the user.<SPAN class=3D064020117-15042002><FONT =
face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D064020117-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D064020117-15042002><FONT face=3DArial =
size=3D2>I'm not=20
  sure what you mean--who is the subscriber, if not the user (i.e. the =
person=20
  trying to place an anonymous call)?</FONT>&nbsp;<FONT face=3DArial =
size=3D2> (This=20
  may be the source of my confusion in the previous paragraph. I was =
using the=20
  term "user" and "subscriber"=20
  interchangeably.)</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D755424716-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff>You could standardise a way of communicating the =
subscriber's=20
  wishes to the UA, and then enforce them. But how does this work for a=20
  forwarding user (or their subscriber) who does not wish to be =
identified=20
  ?<SPAN class=3D064020117-15042002><FONT face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D064020117-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D755424716-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D064020117-15042002><FONT face=3DArial=20
  size=3D2>Presumadly if a UA supports anonymous calling, it =
must&nbsp;give the=20
  subscriber some way of expressing the desire to do so. I don't see any =
need to=20
  standardize that. Once the&nbsp;subscriber expresses that desire, it =
is up to=20
  the UA&nbsp;to do things right (like, don't _automatically_ put=20
  user&nbsp;identifiable information in headers, From or=20
  otherwise.)</FONT>&nbsp;<FONT face=3DArial size=3D2> A UA that =
pretends to support=20
  anonymity but screws it up fits my definition of "badly=20
  behaved."</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D755424716-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D755424716-15042002>...Mark</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Ben Campbell=20
    [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002=20
    17:40<BR><B>To:</B> Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; =
Dean=20
    Willis; Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: =
Summary of RE:=20
    [Sip] Comment, SIP Privacy draft<BR><BR></DIV></FONT>
    <DIV><SPAN class=3D489291916-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>Yes, it is a long thread :-)</FONT></SPAN></DIV>
    <DIV><SPAN class=3D489291916-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D489291916-15042002><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
    pretty much agree with your original assessment for when the network =

    authenticates From, although I would propose a better behavior would =
be to=20
    just disallow a request with a non-anonymous From header _and_ with =
privacy=20
    requested. This approach is more backwards compatible (in case =
something=20
    involved still freaks out about modified From headers), and a lot =
simpler to=20
    implement in the network (no B2BUA required, at least for _this_=20
    reason)</FONT></SPAN></DIV>
    <DIV><SPAN class=3D489291916-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D489291916-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>But in general, the requirement that a network must prevent =
a badly=20
    behaved client from revealing its identity accidentally seems a bit=20
    excessive. It should be sufficient to say that a client _can_ =
prevent its=20
    identity from being revealed.</FONT></SPAN></DIV>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
      size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
      [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April =
15, 2002=20
      11:17 AM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis;=20
      Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of =
RE: [Sip]=20
      Comment, SIP Privacy draft<BR><BR></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>Ben,</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>I know this is a long thread, but I =
have=20
      addressed all these points before. Please see =
below...</SPAN></FONT></DIV>
      <BLOCKQUOTE dir=3Dltr=20
      style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
        <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
        size=3D2>-----Original Message-----<BR><B>From:</B> Ben Campbell =

        [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002 =

        15:44<BR><B>To:</B> Chiou, Mark; Dean Willis; Mpierce1@aol.com;=20
        sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] =
Comment, SIP=20
        Privacy draft<BR><BR></DIV></FONT>
        <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff=20
        size=3D2>You pretty much have to assume that the receiving user =
sees=20
        _anything_ you send his UA, except in the rare case where the =
user has=20
        no control over the UA behavior. Asking a UAS to hide =
information from=20
        its owner is not useful in the generic case.</FONT></SPAN></DIV>
        <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff=20
        size=3D2></FONT></SPAN>&nbsp;</DIV>
        <DIV><SPAN class=3D435513614-15042002><FONT face=3DArial =
color=3D#0000ff=20
        size=3D2>I agree with Dean on this in general, although it =
really doesn't=20
        matter if any _network_ inserted identification is there with =
the=20
        consent of the originating user or not. A privacy request need =
_only_=20
        apply to network-inserted identification data. The UAC has every =

        opportunity _not_ to insert identification if it does not want =
to.=20
        </FONT></SPAN></DIV></BLOCKQUOTE>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>It doesn't matter what device inserts =
the=20
      information. It matters whether it is part of the service for the=20
      information to be inserted. If it is a part of the service, then =
the=20
      responsibility lies with the Service Provider - they defined the =
service=20
      and they are responsible for things the users do effectively at =
their=20
      behest.</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>If I check the From: field I receive =
from my=20
      users is valid, then I am forcing them to put either their =
identity or=20
      'Anonymous' in there.</SPAN></FONT></DIV>
      <DIV>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>If there is then a privacy requirement =
arising=20
      from the rights of ANOTHER PERSON, NOT THE USER, then I have to do =

      something about this - I as the Service Provider have the =
responsibility,=20
      not the user. An example is when the *subscriber* has requested=20
      anonymity.</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D404191416-15042002>If I let my users use any old From =
value, then=20
      the situation is not so clear, but I still argue that for fields =
whose=20
      intention is defined for transport of a particular identity, then =
the=20
      Service Provider has some responsibility. (cf Special Arrangements =
in the=20
      ISDN).</SPAN></FONT></DIV>
      <BLOCKQUOTE dir=3Dltr=20
      style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
        <DIV><FONT color=3D#0000ff><SPAN =
class=3D435513614-15042002><FONT face=3DArial=20
        size=3D2>Expecting the network to screen out ID that _I_ provide =
is a=20
        slippery slope. What if I put "call from Ben" in a Subject =
header? Do=20
        you screen that out as well? What if I _tell_ the destination =
user who I=20
        am in an IM associated with the call? What about the media=20
        session,?<FONT face=3DVerdana size=3D1><SPAN=20
        =
class=3D404191416-15042002>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></DIV=
>
        <DIV><FONT color=3D#0000ff><SPAN =
class=3D435513614-15042002><FONT face=3DArial=20
        size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
        =
class=3D404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV=
></BLOCKQUOTE>
      <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
      size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
      class=3D404191416-15042002>Personally, I don't think screening out =
the=20
      Subject would be necessary, although there might be some market =
demand for=20
      services which did this.&nbsp;Subject is a genuinely transparent =
field.=20
      There is no intention that it contain your identity, and UAs are =
unlikely=20
      to insert your identity here=20
      automatically.</SPAN></FONT></FONT></SPAN></FONT></DIV>
      <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
      size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
      =
class=3D404191416-15042002></SPAN></FONT></FONT></SPAN></FONT>&nbsp;</DIV=
>
      <DIV><FONT color=3D#0000ff><SPAN class=3D435513614-15042002><FONT =
face=3DArial=20
      size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
      =
class=3D404191416-15042002>...Mark</SPAN></FONT></FONT></SPAN></FONT></DI=
V>
      <BLOCKQUOTE dir=3Dltr=20
      style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
        <DIV><SPAN class=3D435513614-15042002></SPAN><FONT =
face=3DTahoma><FONT=20
        size=3D2><SPAN class=3D435513614-15042002><FONT face=3DArial=20
        color=3D#0000ff></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
        <DIV><FONT face=3DTahoma><FONT size=3D2><SPAN=20
        class=3D435513614-15042002>&nbsp;</SPAN>-----Original=20
        Message-----<BR><B>From:</B> sip-admin@ietf.org=20
        [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Chiou,=20
        Mark<BR><B>Sent:</B> Monday, April 15, 2002 9:28 =
AM<BR><B>To:</B> Dean=20
        Willis; Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: =
Summary of=20
        RE: [Sip] Comment, SIP Privacy draft<BR><BR></DIV></FONT></FONT>
        <BLOCKQUOTE dir=3Dltr=20
        style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: =
#0000ff 2px solid; MARGIN-RIGHT: 0px">
          <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
          class=3D765415513-15042002>Dean wrote:</SPAN></FONT></DIV>
          <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
          =
class=3D765415513-15042002>#####################################</SPAN></=
FONT></DIV>
          <DIV><FONT color=3D#0000ff size=3D2><SPAN =
class=3D765415513-15042002>
          <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
          size=3D+0><FONT size=3D2><SPAN =
class=3D819102015-12042002><FONT face=3DArial=20
          color=3D#0000ff>The point I'm trying to make is that To and =
From are=20
          USER supplied information. The&nbsp;Service Provider doesn't =
insert=20
          them, inspect them, or have any responsibility for them. =
Insisting=20
          that the service provider change them is just =
like&nbsp;insisting that=20
          the service provider change any other user-supplied=20
          information,&nbsp;such as the&nbsp;words I'm typing now. =
Anything I=20
          put in my To or from field is deliberately and explicitly =
intended to=20
          be delivered to the terminating side of the dialog, and if as =
a=20
          service provider you FAIL to do that, then you have DAMAGED my =
data,=20
          and my lawyers will be in=20
          touch!</FONT></SPAN></FONT></FONT></FONT></FONT></DIV>
          <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
          size=3D+0><FONT color=3D#0000ff size=3D2><SPAN=20
          =
class=3D819102015-12042002></SPAN></FONT></FONT></FONT></FONT><FONT=20
          face=3DArial></FONT>&nbsp;</DIV>
          <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =
size=3D+0><FONT=20
          size=3D+0><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
          class=3D819102015-12042002>The calling-party-identity =
presentation=20
          infromation (CLIP or RPID) is NOT inserted by the user. It is =
inserted=20
          by the network, WITHOUT THE CONSENT of the user. As such, the =
operator=20
          is responsible for the protection of that information in a =
manner=20
          consistent with regulation and the expressed wishes of the=20
          user.</SPAN></FONT></FONT></FONT></FONT></DIV>
          <DIV><FONT lang=3D0 FAMILY=3D"SANSSERIF"><FONT =
face=3DArial><SPAN=20
          class=3D819102015-12042002><SPAN=20
          =
class=3D765415513-15042002>&lt;snip&gt;</SPAN></SPAN></FONT></FONT></DIV>=

          <DIV><FONT lang=3D0 color=3D#000000 FAMILY=3D"SANSSERIF"><FONT =

          color=3D#0000ff><SPAN=20
          class=3D819102015-12042002></SPAN></FONT></FONT><FONT=20
          face=3DArial>&nbsp;<SPAN=20
          =
class=3D765415513-15042002>##############################################=
############</SPAN></FONT></DIV>
          <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>No, =
The CLIP is=20
          inserted by the network WITH THE CONSENT of the user. When you =

          subscribe a line from a PSTN service provider, the form you =
filled=20
          indicating whether you want your telephone number to be PUBLIC =
or=20
          PRIVATE. (of course, the default is PUBLIC if you leave it =
blank.)=20
          When you make a call, the PRIVATE or PUBLIC status value is =
retrieved=20
          and set the Address Presentation Restricted Indicator (APRI) =
field in=20
          SS7 environment (Note that, CLIR toggles the values), then the =

          terminating side of the network based on the APRI value to =
determine=20
          whether the Calling Party Number shall be delivered to the =
callee's=20
          CPE.</SPAN></FONT></DIV>
          <DIV><FONT face=3DArial><SPAN=20
          class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
          <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002>Since =
we want to=20
          push the decision making to the UA, we should standardize the =
rule=20
          such that the UA shall display the From information based on =
Privacy=20
          indication. If we think that too many UAs won't follow this =
rule, than=20
          we should allow the Service Provider of terminating side (not =
the=20
          originating side) to alter the From information before =
terminating the=20
          call to the UA based on the Privacy =
indication.</SPAN></FONT></DIV>
          <DIV><FONT face=3DArial><SPAN=20
          class=3D765415513-15042002></SPAN></FONT>&nbsp;</DIV>
          <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =

          face=3DArial><SPAN class=3D765415513-15042002>This is my penny =

          thought,</SPAN></FONT></SPAN></FONT></DIV>
          <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =

          face=3DArial><SPAN=20
          =
class=3D765415513-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
          <DIV><FONT face=3DArial><SPAN class=3D765415513-15042002><FONT =

          face=3DArial><SPAN class=3D765415513-15042002>Mark=20
          =
Chiou&nbsp;</SPAN></FONT></SPAN></FONT></SPAN></FONT></DIV></DIV></BLOCKQ=
UOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'David R. Oran'" <oran@cisco.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>, Mpierce1@aol.com,
        sip@ietf.org, "'sipping@ietf.org'" <sipping@ietf.org>
Subject: Network Asserted Identity (was: RE: Summary of RE: [Sip] Comment,
	 SIP Privacy draft)
Date: Mon, 15 Apr 2002 18:23:26 +0100
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OK, so lets see if we can conclude on _short-term_ Network Asserted identity
requirements in a separate thread. Follow-ups to SIPPING ONLY please.

First, assume a trust domain model as per privacy draft applicability
statement. Entities within a domain trust each other, entities in different
domains are mutually untrusted.

I'll try to build up the requirements, least contraversial first:
NAI1) The ability for network entities to insert a network asserted identity
and pass to other network entities within the same trust domain.
NAI2) The ability for a network entity to pass a (non-private) network
asserted identity directly to a UA
NAI3) Secure transport of network asserted identity across 'untrusted'
networks back to the same trust domain (including between trust 'islands' -
disconnected parts of a single trust domain)
NAI4) Transport of network asserted identity between trust domains (i.e.
transport in one domain of a network asserted identity received from another
domain)

For clarity, 
(a) the requirement for a user to choose one of a number of identities,
whilst retaining privacy, is a 3GPP requirement, but is *out of scope* of
this thread.
(b) Call Trace implemented by encrypting the identity and passing this to
the called user, who can then pass the encrypted identity to the generating
service provider/authorities, is considered a sub-case of NAI3 - the
identity is passed to the 'untrusted' user and back to the original Trust
Domain (the Service Provider) before being passed to the authorities.

I expect the From/To debate will continue, probably to the end of time, but
I don't see a strong relationship with this issue.

If people agree with my statement of the potential requirements, perhaps we
could open discussion on which of these should be considered for immediate
standardisation. The others can be put into the pot for discussion of the
generic privacy requirements.

For my part, I am most concerned about NAI1 and NAI2.

Regards...Mark

> -----Original Message-----
> From: David R. Oran [mailto:oran@cisco.com]
> Sent: 15 April 2002 17:39
> To: 'Rosen, Brian'; Mpierce1@aol.com; sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Brian Rosen wrote:
> >If we could get consensus that trace-with-anonymity is the problem,
> then > we can
> > work on solutions.  Until we get to something as straightforward as 
> > that,
> > we are going to spin a lot of wheels.
> 
> That's always been the prime motivator for me, plus the 
> secondary use of
> NAI for call return to an anonymous caller.
> 
> Dave
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<P><FONT SIZE=3D2>OK, so lets see if we can conclude on _short-term_ =
Network Asserted identity requirements in a separate thread. Follow-ups =
to SIPPING ONLY please.</FONT></P>

<P><FONT SIZE=3D2>First, assume a trust domain model as per privacy =
draft applicability statement. Entities within a domain trust each =
other, entities in different domains are mutually untrusted.</FONT></P>

<P><FONT SIZE=3D2>I'll try to build up the requirements, least =
contraversial first:</FONT>
<BR><FONT SIZE=3D2>NAI1) The ability for network entities to insert a =
network asserted identity and pass to other network entities within the =
same trust domain.</FONT></P>

<P><FONT SIZE=3D2>NAI2) The ability for a network entity to pass a =
(non-private) network asserted identity directly to a UA</FONT>
<BR><FONT SIZE=3D2>NAI3) Secure transport of network asserted identity =
across 'untrusted' networks back to the same trust domain (including =
between trust 'islands' - disconnected parts of a single trust =
domain)</FONT></P>

<P><FONT SIZE=3D2>NAI4) Transport of network asserted identity between =
trust domains (i.e. transport in one domain of a network asserted =
identity received from another domain)</FONT></P>

<P><FONT SIZE=3D2>For clarity, </FONT>
<BR><FONT SIZE=3D2>(a) the requirement for a user to choose one of a =
number of identities, whilst retaining privacy, is a 3GPP requirement, =
but is *out of scope* of this thread.</FONT></P>

<P><FONT SIZE=3D2>(b) Call Trace implemented by encrypting the identity =
and passing this to the called user, who can then pass the encrypted =
identity to the generating service provider/authorities, is considered =
a sub-case of NAI3 - the identity is passed to the 'untrusted' user and =
back to the original Trust Domain (the Service Provider) before being =
passed to the authorities.</FONT></P>

<P><FONT SIZE=3D2>I expect the From/To debate will continue, probably =
to the end of time, but I don't see a strong relationship with this =
issue.</FONT></P>

<P><FONT SIZE=3D2>If people agree with my statement of the potential =
requirements, perhaps we could open discussion on which of these should =
be considered for immediate standardisation. The others can be put into =
the pot for discussion of the generic privacy requirements.</FONT></P>

<P><FONT SIZE=3D2>For my part, I am most concerned about NAI1 and =
NAI2.</FONT>
</P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: David R. Oran [<A =
HREF=3D"mailto:oran@cisco.com">mailto:oran@cisco.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 15 April 2002 17:39</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Rosen, Brian'; Mpierce1@aol.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Brian Rosen wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;If we could get consensus that =
trace-with-anonymity is the problem,</FONT>
<BR><FONT SIZE=3D2>&gt; then &gt; we can</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; work on solutions.&nbsp; Until we get to =
something as straightforward as </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; we are going to spin a lot of =
wheels.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; That's always been the prime motivator for me, =
plus the </FONT>
<BR><FONT SIZE=3D2>&gt; secondary use of</FONT>
<BR><FONT SIZE=3D2>&gt; NAI for call return to an anonymous =
caller.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dave</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 18:32:13 +0100
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Ben,
 
The user and the subscriber are not the same.
 
See
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html
<http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html>
for an explanation from earlier today.
 
...Mark

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<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D144223317-15042002>Ben,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D144223317-15042002>The=20
user and the subscriber are not the same.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D144223317-15042002>See=20
<A=20
href=3D"http://www1.ietf.org/mail-archive/working-groups/sip/current/msg=
04720.html">http://www1.ietf.org/mail-archive/working-groups/sip/current=
/msg04720.html</A>&nbsp;for=20
an explanation from earlier today.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D144223317-15042002>...Mark</SPAN></FONT></DIV></BODY></HTML>

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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 sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Mon Apr 15 14:58:19 2002
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 13:23:12 -0500
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MessageOK, in that case I would expect the subscriber profile to have an
optional entry indicating that only anonymous calls were allowed. If some
user (the subscriber or otherwise) then tries to make a non-anonymous call,
it gets rejected.

It also makes sense that a well crafted UA that supported anonymous calling
would have an option to always be anonymous, to avoid the usual cycle of try
non-anonymous, fail, then try again with anonymous.

I don't see why this is any different than any other rejection due to a
policy violation.
  -----Original Message-----
  From: Mark Watson [mailto:mwatson@nortelnetworks.com]
  Sent: Monday, April 15, 2002 12:32 PM
  To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
  Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Ben,

  The user and the subscriber are not the same.

  See
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html
for an explanation from earlier today.

  ...Mark

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D531211918-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>OK, in=20
that case I would expect the subscriber profile to have an optional =
entry=20
indicating that only anonymous calls were allowed. If some user (the =
subscriber=20
or otherwise) then tries to make a non-anonymous call, it gets=20
rejected.</FONT></SPAN></DIV>
<DIV><SPAN class=3D531211918-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D531211918-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>It=20
also makes sense that a well crafted UA that supported anonymous calling =
would=20
have an option to always be anonymous, to avoid the usual cycle of try=20
non-anonymous, fail, then try again with anonymous.</FONT></SPAN></DIV>
<DIV><SPAN class=3D531211918-15042002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D531211918-15042002><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
don't see why this is any different than any other rejection due to a =
policy=20
violation.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April 15, =
2002=20
  12:32 PM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis;=20
  Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: =
[Sip]=20
  Comment, SIP Privacy draft<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002>Ben,</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002>The user and the subscriber are not the=20
  same.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002>See <A=20
  =
href=3D"http://www1.ietf.org/mail-archive/working-groups/sip/current/msg0=
4720.html">http://www1.ietf.org/mail-archive/working-groups/sip/current/m=
sg04720.html</A>&nbsp;for=20
  an explanation from earlier today.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  =
class=3D144223317-15042002>...Mark</SPAN></FONT></DIV></BLOCKQUOTE></BODY=
></HTML>

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Mon Apr 15 15:10:19 2002
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Message-ID: <A3C2399B2FACD411A54200508BE39C74054F713D@zwcwd00r.europe.nortel.com>
From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 19:45:49 +0100
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Three problems:
1) In practice you might expect the 'policy' to be more flexible than just
'everything anonymous' (e.g. internal calls non-anonymous, external calls
anonymous), although the blanket anonymity would be the most that Service
Providers could be required to provide. Anyway, we do not want the only
solution to this requirement to be to configure the UA as always anonymous.
 
The users wish is that they be identified if possible. The 'policy' solution
means that the only way of finding out if a particular call must be
anonymous is to try and see, and then retry if it fails.
 
2) Even with a well-crafted UA, why is the _user_ expected to re-configure
their client because of the _subscriber's_ wishes. And when the policy
changes, how will we indicate to all the users that they should now
re-configure their clients again ?
 
I hope all the Service Providers wanting to launch SIP services have got
their list of client requirements worked out. If your subscriber asks for
this anonymity service they will not be impressed if they have to
upgrade/re-configure all their clients. If you didn't inform them in advance
that the clients needed these capabilities, then they could refuse to
upgrade and still demand the anonymity service.
 
3) What about a user who has asked their home proxy to forward calls, and
who wishes to remain anonymous - is the inclusion of this persons identity
in the To field just a policy violation ? How many UAs have an option to set
the To field blank ? This would be more of a 'call rejection' service than a
'call forwarding' service. I think that if in trying to get a Data Protected
version of a service, the user finds that they can't have the service at
all, they would be entitled to be a bit miffed.
 
...Mark

-----Original Message-----
From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
Sent: 15 April 2002 19:23
To: Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis;
Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


OK, in that case I would expect the subscriber profile to have an optional
entry indicating that only anonymous calls were allowed. If some user (the
subscriber or otherwise) then tries to make a non-anonymous call, it gets
rejected.
 
It also makes sense that a well crafted UA that supported anonymous calling
would have an option to always be anonymous, to avoid the usual cycle of try
non-anonymous, fail, then try again with anonymous.
 
I don't see why this is any different than any other rejection due to a
policy violation.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Monday, April 15, 2002 12:32 PM
To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Ben,
 
The user and the subscriber are not the same.
 
See
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html
<http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html>
for an explanation from earlier today.
 
...Mark


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<BODY>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002>Three problems:</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>1) 
In practice you might expect the 'policy' to be more flexible than just 
'everything anonymous' (e.g. internal calls non-anonymous, external calls 
anonymous), although the blanket anonymity would be the most that Service 
Providers could be required to provide. Anyway, we do not want the only solution 
to this requirement to be to configure the UA as always 
anonymous.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>The 
users wish is that they be identified if possible. The 'policy' solution means 
that the only way of finding out if a particular call must be anonymous is to 
try and see, and then retry if it fails.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>2) 
Even with a well-crafted UA, why is the _user_ expected to re-configure their 
client because of the _subscriber's_ wishes. And when the policy changes, how 
will we indicate to all the users that they should now re-configure their 
clients again ?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>I 
hope all the Service Providers wanting to launch SIP services have got their 
list of client requirements worked out. If your subscriber asks for this 
anonymity service they will not be impressed if they have to 
upgrade/re-configure all their clients. If you didn't inform them in advance 
that the clients needed these capabilities, then they could refuse to upgrade 
and still demand the anonymity service.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>3) 
What about a user who has asked their home proxy to forward calls, and who 
wishes to remain anonymous - is the inclusion of this persons identity in the To 
field just a policy violation ? How many UAs have an option to set the To field 
blank ? This would be more of a 'call rejection' service than a 'call 
forwarding' service. I think that if in trying to get a Data Protected version 
of a service, the user finds that they can't have the service at all, they would 
be entitled to be a bit miffed.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=455513518-15042002>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ben Campbell 
  [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002 
  19:23<BR><B>To:</B> Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis; 
  Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] 
  Comment, SIP Privacy draft<BR><BR></DIV></FONT>
  <DIV><SPAN class=531211918-15042002><FONT color=#0000ff face=Arial size=2>OK, 
  in that case I would expect the subscriber profile to have an optional entry 
  indicating that only anonymous calls were allowed. If some user (the 
  subscriber or otherwise) then tries to make a non-anonymous call, it gets 
  rejected.</FONT></SPAN></DIV>
  <DIV><SPAN class=531211918-15042002><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=531211918-15042002><FONT color=#0000ff face=Arial size=2>It 
  also makes sense that a well crafted UA that supported anonymous calling would 
  have an option to always be anonymous, to avoid the usual cycle of try 
  non-anonymous, fail, then try again with anonymous.</FONT></SPAN></DIV>
  <DIV><SPAN class=531211918-15042002><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=531211918-15042002><FONT color=#0000ff face=Arial size=2>I 
  don't see why this is any different than any other rejection due to a policy 
  violation.</FONT></SPAN></DIV>
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
    size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
    [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April 15, 2002 
    12:32 PM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis; 
    Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] 
    Comment, SIP Privacy draft<BR><BR></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002>Ben,</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002>The user and the subscriber are not the 
    same.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002>See <A 
    href="http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html">http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html</A>&nbsp;for 
    an explanation from earlier today.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
    class=144223317-15042002>...Mark</SPAN></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 14:08:03 -0500
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Message
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Mark
Watson
  Sent: Monday, April 15, 2002 1:46 PM
  To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
  Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Three problems:
  1) In practice you might expect the 'policy' to be more flexible than just
'everything anonymous' (e.g. internal calls non-anonymous, external calls
anonymous), although the blanket anonymity would be the most that Service
Providers could be required to provide. Anyway, we do not want the only
solution to this requirement to be to configure the UA as always anonymous.

  Certainly not. I only meant this for the situation where the subscriber
has asserted that all calls from his account are to be anonymous, and an
individual user of that account might not know about that policy choice.

  The users wish is that they be identified if possible. The 'policy'
solution means that the only way of finding out if a particular call must be
anonymous is to try and see, and then retry if it fails.

  Which then allows the user to decide whether or not to continue according
to policy, or just not make the call. Otherwise, the network seems to be in
the business of deciding that the user really does not mind if the call is
anonymous. That seems fairly benign in this particular case, but not
generically (what if the policy was _no_ anonymous calls.)

  2) Even with a well-crafted UA, why is the _user_ expected to re-configure
their client because of the _subscriber's_ wishes. And when the policy
changes, how will we indicate to all the users that they should now
re-configure their clients again ?

  They don't have to. Doing so just cuts down on the
reject-due-to-policy-violation-oops-retry cycles. Now, I assume the user
must have some relationship with the subscriber (else how does he have
credentials), and the subscriber _could_ just tell the user. Not everything
has to be automatic.

  I hope all the Service Providers wanting to launch SIP services have got
their list of client requirements worked out. If your subscriber asks for
this anonymity service they will not be impressed if they have to
upgrade/re-configure all their clients. If you didn't inform them in advance
that the clients needed these capabilities, then they could refuse to
upgrade and still demand the anonymity service.

  Well, once the idea of an anonymity service is fully formed (presumably
what we are doing here) it makes sense that it is a feature that a client
can support or not support. (Maybe even a supported or requires tag?) Then
it is reasonable for a service provider to say you can make anonymous calls
if your client supports anonymity. If a subscriber has a policy of _only_
anonymous calls, and a user has a client that cannot make anonymous calls,
then he is SOL.  I don't see why this is any different than any _other_
situation where an optional client feature might be required by SP or
subscriber policy. Again, the alternative is a presumption on the networks
part that it knows what the user _really_ means to do better than the user
does. Such an presumption on policy violations scares me in the generic
sense.

  3) What about a user who has asked their home proxy to forward calls, and
who wishes to remain anonymous - is the inclusion of this persons identity
in the To field just a policy violation ? How many UAs have an option to set
the To field blank ? This would be more of a 'call rejection' service than a
'call forwarding' service. I think that if in trying to get a Data Protected
version of a service, the user finds that they can't have the service at
all, they would be entitled to be a bit miffed.

  ...Mark
    -----Original Message-----
    From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
    Sent: 15 April 2002 19:23
    To: Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; Dean Willis;
Mpierce1@aol.com; sip@ietf.org
    Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


    OK, in that case I would expect the subscriber profile to have an
optional entry indicating that only anonymous calls were allowed. If some
user (the subscriber or otherwise) then tries to make a non-anonymous call,
it gets rejected.

    It also makes sense that a well crafted UA that supported anonymous
calling would have an option to always be anonymous, to avoid the usual
cycle of try non-anonymous, fail, then try again with anonymous.

    I don't see why this is any different than any other rejection due to a
policy violation.
      -----Original Message-----
      From: Mark Watson [mailto:mwatson@nortelnetworks.com]
      Sent: Monday, April 15, 2002 12:32 PM
      To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
      Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


      Ben,

      The user and the subscriber are not the same.

      See
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html
for an explanation from earlier today.

      ...Mark

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Mark =
Watson<BR><B>Sent:</B>=20
  Monday, April 15, 2002 1:46 PM<BR><B>To:</B> 'Ben Campbell'; Chiou, =
Mark; Dean=20
  Willis; Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary =
of RE:=20
  [Sip] Comment, SIP Privacy draft<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002>Three problems:</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff>1) In practice you might expect the 'policy' to be =
more flexible=20
  than just 'everything anonymous' (e.g. internal calls non-anonymous, =
external=20
  calls anonymous), although the blanket anonymity would be the most =
that=20
  Service Providers could be required to provide. Anyway, we do not want =
the=20
  only solution to this requirement to be to configure the UA as always=20
  anonymous.<SPAN class=3D278085518-15042002><FONT face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D278085518-15042002><FONT face=3DArial =
size=3D2>Certainly=20
  not. I only meant this for the situation where the subscriber has =
asserted=20
  that&nbsp;all calls from his account are to be anonymous, and an =
individual=20
  user of that account might not know about that policy=20
  choice.</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff>The users wish is that they be identified if possible. =
The=20
  'policy' solution means that the only way of finding out if a =
particular call=20
  must be anonymous is to try and see, and then retry if it fails.<SPAN=20
  class=3D278085518-15042002><FONT face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D278085518-15042002><FONT face=3DArial =
size=3D2>Which=20
  then allows the user&nbsp;to decide whether or not to continue =
according to=20
  policy, or just not make the call. Otherwise, the network seems to=20
  be&nbsp;in&nbsp;the business of deciding that the user really does=20
  not&nbsp;mind if the call is anonymous. That seems fairly benign in =
this=20
  particular case, but not generically (what if the policy was _no_ =
anonymous=20
  calls.)</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff>2) Even with a well-crafted UA, why is the _user_ =
expected to=20
  re-configure their client because of the _subscriber's_ wishes. And =
when the=20
  policy changes, how will we indicate to all the users that they should =
now=20
  re-configure their clients again ?<SPAN =
class=3D278085518-15042002><FONT=20
  face=3DArial size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D278085518-15042002><FONT face=3DArial =
size=3D2>They=20
  don't have to. Doing so just cuts down on the=20
  reject-due-to-policy-violation-oops-retry cycles. Now, I assume the =
user must=20
  have some relationship with the subscriber (else how&nbsp;does he have =

  credentials), and&nbsp;the subscriber _could_ just tell the user. Not=20
  everything has to be =
automatic.</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff>I hope all the Service Providers wanting to launch SIP =
services=20
  have got their list of client requirements worked out. If your =
subscriber asks=20
  for this anonymity service they will not be impressed if they have to=20
  upgrade/re-configure all their clients. If you didn't inform them in =
advance=20
  that the clients needed these capabilities, then they could refuse to =
upgrade=20
  and still demand the anonymity service.<SPAN =
class=3D278085518-15042002><FONT=20
  face=3DArial size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
  color=3D#0000ff><SPAN class=3D278085518-15042002><FONT face=3DArial =
size=3D2>Well,=20
  once the idea of an anonymity service is fully formed (presumably what =
we are=20
  doing here) it makes sense&nbsp;that it is a feature that a client can =
support=20
  or not support.&nbsp;(Maybe even a supported or requires tag?) Then it =
is=20
  reasonable&nbsp;for a service provider to say you can make anonymous =
calls if=20
  your client supports anonymity.&nbsp;If a subscriber has a policy of =
_only_=20
  anonymous&nbsp;calls, and a user has a client that&nbsp;cannot make =
anonymous=20
  calls, then he is&nbsp;SOL.</FONT>&nbsp;<FONT face=3DArial size=3D2> I =
don't see=20
  why this is any different than any _other_ situation where an optional =
client=20
  feature might be required by SP or subscriber policy. Again, the =
alternative=20
  is a presumption on the networks part that it knows what the user =
_really_=20
  means to do better than the user does. Such an presumption on policy=20
  violations scares me in the generic sense.=20
  </FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D455513518-15042002>3)=20
  What about a user who has asked their home proxy to forward calls, and =
who=20
  wishes to remain anonymous - is the inclusion of this persons identity =
in the=20
  To field just a policy violation ? How many UAs have an option to set =
the To=20
  field blank ? This would be more of a 'call rejection' service than a =
'call=20
  forwarding' service. I think that if in trying to get a Data Protected =
version=20
  of a service, the user finds that they can't have the service at all, =
they=20
  would be entitled to be a bit miffed.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D455513518-15042002>...Mark</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Ben Campbell=20
    [mailto:bcampbell@dynamicsoft.com]<BR><B>Sent:</B> 15 April 2002=20
    19:23<BR><B>To:</B> Watson, Mark [MDN05:EP10:EXCH]; Chiou, Mark; =
Dean=20
    Willis; Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: =
Summary of RE:=20
    [Sip] Comment, SIP Privacy draft<BR><BR></DIV></FONT>
    <DIV><SPAN class=3D531211918-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>OK, in that case I would expect the subscriber profile to =
have an=20
    optional entry indicating that only anonymous calls were allowed. If =
some=20
    user (the subscriber or otherwise) then tries to make a =
non-anonymous call,=20
    it gets rejected.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D531211918-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D531211918-15042002><FONT face=3DArial =
color=3D#0000ff size=3D2>It=20
    also makes sense that a well crafted UA that supported anonymous =
calling=20
    would have an option to always be anonymous, to avoid the usual =
cycle of try=20
    non-anonymous, fail, then try again with =
anonymous.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D531211918-15042002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D531211918-15042002><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
    don't see why this is any different than any other rejection due to =
a policy=20
    violation.</FONT></SPAN></DIV>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
      size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
      [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Monday, April =
15, 2002=20
      12:32 PM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis;=20
      Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of =
RE: [Sip]=20
      Comment, SIP Privacy draft<BR><BR></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002>Ben,</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002>The user and the subscriber are not the =

      same.</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002>See <A=20
      =
href=3D"http://www1.ietf.org/mail-archive/working-groups/sip/current/msg0=
4720.html">http://www1.ietf.org/mail-archive/working-groups/sip/current/m=
sg04720.html</A>&nbsp;for=20
      an explanation from earlier today.</SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      class=3D144223317-15042002></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
      =
class=3D144223317-15042002>...Mark</SPAN></FONT></DIV></BLOCKQUOTE></BLOC=
KQUOTE></BLOCKQUOTE></BODY></HTML>

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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 14:10:21 -0500
Message-ID: <CD110021698980419241042CF576B8F2012BAD3F@EXCALIBUR.santera.com>
Thread-Topic: Summary of RE: [Sip] Comment, SIP Privacy draft
Thread-Index: AcHkrd7uMpKggbQQQDmqZat7KoNtkAAAoLCw
From: "Chiou, Mark" <MChiou@Santera.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Ben Campbell" <bcampbell@dynamicsoft.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
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Mark Watson wrote:
######################
3) What about a user who has asked their home proxy to forward calls, =
and who wishes to remain anonymous - is the inclusion of this persons =
identity in the To field just a policy violation ? How many UAs have an =
option to set the To field blank ? This would be more of a 'call =
rejection' service than a 'call forwarding' service. I think that if in =
trying to get a Data Protected version of a service, the user finds that =
they can't have the service at all, they would be entitled to be a bit =
miffed.
#################### =20
Can we insert the privacy information into the Contact header when 3xx =
response is generated for forwarding (redirecting) calls?  Just a =
thought. =20
=20
Mark Chiou
=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D754340419-15042002>Mark=20
Watson wrote:</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D754340419-15042002>######################</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002>3)=20
What about a user who has asked their home proxy to forward calls, and =
who=20
wishes to remain anonymous - is the inclusion of this persons identity =
in the To=20
field just a policy violation ? How many UAs have an option to set the =
To field=20
blank ? This would be more of a 'call rejection' service than a 'call=20
forwarding' service. I think that if in trying to get a Data Protected =
version=20
of a service, the user finds that they can't have the service at all, =
they would=20
be entitled to be a bit miffed.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
class=3D455513518-15042002><SPAN class=3D754340419-15042002><FONT =
face=3DArial=20
size=3D2>####################&nbsp;
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D754340419-15042002>Can we=20
insert the privacy information into the Contact header when 3xx response =
is=20
generated for forwarding (redirecting) calls?&nbsp; Just a =
thought.&nbsp;=20
</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D754340419-15042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D754340419-15042002>Mark=20
Chiou</SPAN></FONT></DIV></FONT></SPAN></SPAN></FONT></FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From daemon@optimus.ietf.org  Mon Apr 15 17:08:15 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Chiou, Mark'" <MChiou@Santera.com>, <Mpierce1@aol.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 15:30:41 -0500
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Dean wrote:
---- 
The calling-party-identity presentation infromation (CLIP or RPID) is
NOT inserted by the user. It is inserted by the network, WITHOUT THE
CONSENT of the user. As such, the operator is responsible for the
protection of that information in a manner consistent with regulation
and the expressed wishes of the user. 
----

Mark wrote: 
----
No, The CLIP is inserted by the network WITH THE CONSENT of the user.
When you subscribe a line from a PSTN service provider, the form you
filled indicating whether you want your telephone number to be PUBLIC or
PRIVATE. (of course, the default is PUBLIC if you leave it blank.) When
you make a call, the PRIVATE or PUBLIC status value is retrieved and set
the Address Presentation Restricted Indicator (APRI) field in SS7
environment (Note that, CLIR toggles the values), then the terminating
side of the network based on the APRI value to determine whether the
Calling Party Number shall be delivered to the callee's CPE. 
----

Coercion <=> consent.

There is no explicit consent within the signaling of an invidiual
transaction. External contractual issues are outside of the protocol,
and tend to violate the "openness" characteristic of an Internet
protocol. In otherwises, there's a philosophical difference between an
explicit request from the user that a network providing break privacy
for that transaction, and a network that breaks privacy just because I
didn't tell it NOT to.

Note of course, that we're not talking about PSTN provoders, we're
talking IP to IP usage over the publicaly addressable Internet.


Mark also wrote:
----
Since we want to push the decision making to the UA, we should
standardize the rule such that the UA shall display the From information
based on Privacy indication. If we think that too many UAs won't follow
this rule, than we should allow the Service Provider of terminating side
(not the originating side) to alter the From information before
terminating the call to the UA based on the Privacy indication.
----

Personally, the UA I uset he most writes every last SIP header and the
body to the screen, and copies them all to a log file just in case. One
cannot rely on the called party's software to protect the privacy of the
calling party.
 

--
Dean


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From daemon@ns.ietf.org  Mon Apr 15 18:30:52 2002
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Mon, 15 Apr 2002 16:50:45 -0500
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From: "Chiou, Mark" <MChiou@Santera.com>
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Coercion <=> consent.

There is no explicit consent within the signaling of an invidiual
transaction. External contractual issues are outside of the protocol,
and tend to violate the "openness" characteristic of an Internet
protocol. In otherwises, there's a philosophical difference between an
explicit request from the user that a network providing break privacy
for that transaction, and a network that breaks privacy just because I
didn't tell it NOT to.

Note of course, that we're not talking about PSTN provoders, we're
talking IP to IP usage over the publicaly addressable Internet.

<Mark Chiou> So, for IP, all the UAs are defaulted to PRIVATE. When a user wishes to make an anonymous calls, SIP needs an indication for the network (i.e. Service Provider) to identify who is calling and translate or retrieve the identity of the calling from the service agreement, then pass the calling identity within the networks without sending the calling identity to the UA of the called party.   

<snip>

Personally, the UA I uset he most writes every last SIP header and the
body to the screen, and copies them all to a log file just in case. One
cannot rely on the called party's software to protect the privacy of the
calling party.
 
<Mark Chiou> Agree. So we have to separate the needs for Privacy and the needs for Network, such that the privacy is fully controlled by the UA. On the other hand, the Service Providers have the responsibility to obtain the UA's identify for network usage (i.e. Emergency, Law Enforcement, Call trace, and etc) without breaking the UA's privacy. I means the UA's identity, which is needed by the Network, shall not be delivered to the terminating of the UA.

Mark Chiou

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From daemon@ns.ietf.org  Tue Apr 16 02:11:02 2002
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Date: Tue, 16 Apr 2002 08:26:43 +0300
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Chiou, Mark'" <MChiou@Santera.com>, <Mpierce1@aol.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Dean Willis writes:

 > No, The CLIP is inserted by the network WITH THE CONSENT of the user.
 > When you subscribe a line from a PSTN service provider, the form you
 > filled indicating whether you want your telephone number to be PUBLIC or
 > PRIVATE. (of course, the default is PUBLIC if you leave it blank.) When
 > you make a call, the PRIVATE or PUBLIC status value is retrieved and set
 > the Address Presentation Restricted Indicator (APRI) field in SS7
 > environment (Note that, CLIR toggles the values), then the terminating
 > side of the network based on the APRI value to determine whether the
 > Calling Party Number shall be delivered to the callee's CPE. 

i don't understand why we would need anything new for this kind of
privacy purpose.

we already have the Anonymous display string.  if th user wants from
privacy, it uses Anonymous display string from which the sip/ss7 gw
knows what to do.  the same to the other direction.

if the call goes to a sip user, then the network MUST rewrite the from
uri and thus implement a b2bua.

-- juha




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Subject: [Sip] Query regarding CIN Delivery
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According to draft-ietf-sip-isup-03.txt

"  The construction of the From field is dependent on the presence of a
   CIN parameter. If the CIN is not present, then the gateway should
   create a dummy From header containing a SIP URI without a user
   portion which communicates only the hostname of the gateway (e.g.
   'sip:gw.level3.net')."

This works fine in case of a GW-GW scenario as the originating GW will
send only the IP Address in the From field and the terminating GW will
remove the IP Address before sending the CIN to the SCN network.

But in case the call lands on a SIP terminal from Gateway then From field will
not reflect the calling party who is the subscriber of GW, rather the SIP
terminal might just display the IP Address of the GW as the calling party
number on the user's screen.

So Wouldn't it be better to add  sip:Anonymous<restricted@gw.level3.net> ?
This way the SIP User will display the display name as Anonymous.
But ofcourse, a limitation of this will be in the GW-GW scenario, the
terminating
GW might map it to CIN presentation restricted while sending it to the SCN n/w.

Any suggestions??

Regards,
Bishista Paul


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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Chiou, Mark'" <MChiou@Santera.com>,
        Ben Campbell
	 <bcampbell@dynamicsoft.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 16 Apr 2002 10:17:16 +0100
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Why would you trust the UA to obey this ? How could you police this if the
calling UA is getting service through a difefrent service provider from the
forwarding party ?
 
...Mark

-----Original Message-----
From: Chiou, Mark [mailto:MChiou@Santera.com]
Sent: 15 April 2002 20:10
To: Watson, Mark [MDN05:EP10:EXCH]; Ben Campbell; Dean Willis;
Mpierce1@aol.com; sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


Mark Watson wrote:
######################
3) What about a user who has asked their home proxy to forward calls, and
who wishes to remain anonymous - is the inclusion of this persons identity
in the To field just a policy violation ? How many UAs have an option to set
the To field blank ? This would be more of a 'call rejection' service than a
'call forwarding' service. I think that if in trying to get a Data Protected
version of a service, the user finds that they can't have the service at
all, they would be entitled to be a bit miffed.
####################  
Can we insert the privacy information into the Contact header when 3xx
response is generated for forwarding (redirecting) calls?  Just a thought.  
 
Mark Chiou
 


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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=114482009-16042002>Why 
would you trust the UA to obey this ? How could you police this if the calling 
UA is getting service through a difefrent service provider from the forwarding 
party ?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=114482009-16042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=114482009-16042002>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Chiou, Mark 
  [mailto:MChiou@Santera.com]<BR><B>Sent:</B> 15 April 2002 20:10<BR><B>To:</B> 
  Watson, Mark [MDN05:EP10:EXCH]; Ben Campbell; Dean Willis; Mpierce1@aol.com; 
  sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: [Sip] Comment, SIP Privacy 
  draft<BR><BR></DIV></FONT>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=754340419-15042002>Mark 
  Watson wrote:</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=754340419-15042002>######################</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=455513518-15042002>3) 
  What about a user who has asked their home proxy to forward calls, and who 
  wishes to remain anonymous - is the inclusion of this persons identity in the 
  To field just a policy violation ? How many UAs have an option to set the To 
  field blank ? This would be more of a 'call rejection' service than a 'call 
  forwarding' service. I think that if in trying to get a Data Protected version 
  of a service, the user finds that they can't have the service at all, they 
  would be entitled to be a bit miffed.</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
  class=455513518-15042002><SPAN class=754340419-15042002><FONT face=Arial 
  size=2>####################&nbsp; 
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=754340419-15042002>Can 
  we insert the privacy information into the Contact header when 3xx response is 
  generated for forwarding (redirecting) calls?&nbsp; Just a thought.&nbsp; 
  </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=754340419-15042002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=754340419-15042002>Mark 
  Chiou</SPAN></FONT></DIV></FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Tue Apr 16 05:58:29 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Ben wrote:

Which then allows the user to decide whether or not to continue according to
policy, or just not make the call. Otherwise, the network seems to be in the
business of deciding that the user really does not mind if the call is
anonymous. That seems fairly benign in this particular case, but not
generically (what if the policy was _no_ anonymous calls.) 

A 'no anonymous calls' policy would be in breach of the users right to make
anonymous calls. It is not the network that is deciding that the user does
not mind if the call is anonymous, it is the subscriber, and they have a
legal right to do this. The subscribe must be able to control what
information the network reveals about them. The user does not have the right
to ask the network to reveal information that the subscriber does not want
revealed - although the use can of course reveal the information themselves.

2) Even with a well-crafted UA, why is the _user_ expected to re-configure
their client because of the _subscriber's_ wishes. And when the policy
changes, how will we indicate to all the users that they should now
re-configure their clients again ? 
 
They don't have to. Doing so just cuts down on the
reject-due-to-policy-violation-oops-retry cycles. Now, I assume the user
must have some relationship with the subscriber (else how does he have
credentials), and the subscriber _could_ just tell the user. Not everything
has to be automatic. 

Frequent reject-due-to- policy-violation-oops-retry cycles are just
unaceptable for wireless environments. There must be a 'works first time'
solution. If you think the user needs to be informed that their call has
become anonymous, then fine, let's standardise this indication.

I hope all the Service Providers wanting to launch SIP services have got
their list of client requirements worked out. If your subscriber asks for
this anonymity service they will not be impressed if they have to
upgrade/re-configure all their clients. If you didn't inform them in advance
that the clients needed these capabilities, then they could refuse to
upgrade and still demand the anonymity service. 
 
Well, once the idea of an anonymity service is fully formed (presumably what
we are doing here) it makes sense that it is a feature that a client can
support or not support. (Maybe even a supported or requires tag?) Then it is
reasonable for a service provider to say you can make anonymous calls if
your client supports anonymity. If a subscriber has a policy of _only_
anonymous calls, and a user has a client that cannot make anonymous calls,
then he is SOL.  I don't see why this is any different than any _other_
situation where an optional client feature might be required by SP or
subscriber policy. Again, the alternative is a presumption on the networks
part that it knows what the user _really_ means to do better than the user
does. Such an presumption on policy violations scares me in the generic
sense.  

It's different because the 'subscriber policy' feature is not just a groovy
addition to the service, but something that the Service Provide must provide
for the subscriber 'by a simple means' (in the words of the European
legislation) by which they mean I make a phone call to my Service Provider
and ask. Upgrading possibly hundreds of clients might not be considered a
'simple means'.
 
Again, not a question of knowing what the user 'wants' - in this case the
subscribers wishe's overrule the users wishe's - we know the user wants to
reveal their identity but in this case the user is 'SOL'.
 
...Mark

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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=866021009-16042002>Ben 
wrote:</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff><SPAN class=278085518-15042002><FONT face=Arial size=2>Which 
    then allows the user&nbsp;to decide whether or not to continue according to 
    policy, or just not make the call. Otherwise, the network seems to 
    be&nbsp;in&nbsp;the business of deciding that the user really does 
    not&nbsp;mind if the call is anonymous. That seems fairly benign in this 
    particular case, but not generically (what if the policy was _no_ anonymous 
    calls.)</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=866021009-16042002>A 
'no anonymous calls' policy would be in breach of the users right to make 
anonymous calls. It is not the network that is deciding that the user does not 
mind if the call is anonymous, it is the subscriber, and they have a legal right 
to do this. The subscribe must be able to control what information the network 
reveals about them. The user does not have the right to ask the network to 
reveal information that the subscriber does not want revealed - although the use 
can of course reveal the information themselves.</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff>2) Even with a well-crafted UA, why is the _user_ expected to 
    re-configure their client because of the _subscriber's_ wishes. And when the 
    policy changes, how will we indicate to all the users that they should now 
    re-configure their clients again ?<SPAN class=278085518-15042002><FONT 
    face=Arial size=2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff><SPAN 
    class=278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff><SPAN class=278085518-15042002><FONT face=Arial size=2>They 
    don't have to. Doing so just cuts down on the 
    reject-due-to-policy-violation-oops-retry cycles. Now, I assume the user 
    must have some relationship with the subscriber (else how&nbsp;does he have 
    credentials), and&nbsp;the subscriber _could_ just tell the user. Not 
    everything has to be 
    automatic.</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE>
<DIV><SPAN class=455513518-15042002></SPAN><FONT size=1><FONT 
color=#0000ff><FONT face=Verdana><SPAN class=866021009-16042002>Frequent 
reject-due-to-</SPAN><SPAN 
class=866021009-16042002>&nbsp;policy-violation-oops-retry cycles are just 
unaceptable for wireless environments. There must be a 'works first time' 
solution. If you think the user needs to be informed that their call has become 
anonymous, then fine, let's standardise this 
indication.</SPAN></FONT></FONT></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff>I hope all the Service Providers wanting to launch SIP 
    services have got their list of client requirements worked out. If your 
    subscriber asks for this anonymity service they will not be impressed if 
    they have to upgrade/re-configure all their clients. If you didn't inform 
    them in advance that the clients needed these capabilities, then they could 
    refuse to upgrade and still demand the anonymity service.<SPAN 
    class=278085518-15042002><FONT face=Arial 
    size=2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=Verdana size=1><SPAN class=455513518-15042002><FONT 
    color=#0000ff><SPAN 
    class=278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
    class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
    size=2>Well, once the idea of an anonymity service is fully formed 
    (presumably what we are doing here) it makes sense&nbsp;that it is a feature 
    that a client can support or not support.&nbsp;(Maybe even a supported or 
    requires tag?) Then it is reasonable&nbsp;for a service provider to say you 
    can make anonymous calls if your client supports anonymity.&nbsp;If a 
    subscriber has a policy of _only_ anonymous&nbsp;calls, and a user has a 
    client that&nbsp;cannot make anonymous calls, then he 
    is&nbsp;SOL.</FONT>&nbsp;<FONT face=Arial size=2> I don't see why this is 
    any different than any _other_ situation where an optional client feature 
    might be required by SP or subscriber policy. Again, the alternative is a 
    presumption on the networks part that it knows what the user _really_ means 
    to do better than the user does. Such an presumption on policy violations 
    scares me in the generic sense.&nbsp;<FONT face=Verdana size=1><SPAN 
    class=866021009-16042002>&nbsp;</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN class=866021009-16042002>It's different 
because the 'subscriber policy'&nbsp;feature is not just a groovy addition to 
the&nbsp;service,&nbsp;but something that the Service Provide must provide for 
the subscriber 'by a simple means' (in the words of the European legislation) by 
which they mean I make a phone call to my Service Provider and ask. Upgrading 
possibl</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT><FONT 
color=#0000ff><FONT face=Verdana size=1><SPAN class=455513518-15042002><SPAN 
class=278085518-15042002><FONT face=Arial size=2><FONT face=Verdana size=1><SPAN 
class=866021009-16042002>y&nbsp;hundreds of clients might not be considered a 
'simple means'.</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN 
class=866021009-16042002></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN class=866021009-16042002>Again, not a 
question of knowing what the user 'wants' - in this case the subscribers wishe's 
overrule the users wishe's - we know the user wants to reveal their identity but 
in this case the user is 
'SOL'.</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN 
class=866021009-16042002></SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=455513518-15042002><SPAN class=278085518-15042002><FONT face=Arial 
size=2><FONT face=Verdana size=1><SPAN 
class=866021009-16042002>...Mark</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></DIV></BODY></HTML>

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From daemon@optimus.ietf.org  Tue Apr 16 06:52:32 2002
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Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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the current bis-09 text says:

The From header field allows for a display name. A UAC SHOULD use the
display name  Anonymous , along with a syntactically correct, but
otherwise meaningless URI (like sip:thisis@anonymous.invalid), if
the identity of the client is to remain hidden.

this is all that is needed except that in my service the uri must be
meaningful.

most UAs on the market support this.  so fail to see what the issue is.

-- juha


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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From daemon@optimus.ietf.org  Tue Apr 16 07:07:04 2002
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Message-ID: <A3C2399B2FACD411A54200508BE39C74054F7144@zwcwd00r.europe.nortel.com>
From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>,
        "'Ben Campbell'"
	 <bcampbell@dynamicsoft.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 16 Apr 2002 11:31:20 +0100
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The client capability issue was for the case where the user does not wish to
remain anonymous, but the subscriber does.

Without a network modification of the From field then:

1) The user includes their identity, and the session is rejected. There is
no reject code which clearly indicates what action the user/UA should take.
There will be no current UAs which would know to auto-retry with
'Anonymous', so this will require user intervention on all calls.
Auto-retrys are no good for wireless anyway.

2) It was suggested that the client should have a 'always anonymous' option
which the user can set, so that all calls via this subscription will be
anonymous. Again, current clients may or may not have this, so relying on
this to provide the 'subscriber anonymity' service without the retrys of (1)
is not such a good idea unless you have clearly told subscribers - before
they subscribe - that such a capability is needed.

Service Providers need to be well aware of the client capabilities needed to
implement the regulatory services they have to provide, so that they can
make it clear to potential subscribers what those required client
capabilities are.

Even so, I really can't see all the above being realistic. Imagine a
subscriber goes to their service provider and requests anonymity. The first
thing that happens is that all calls through the subscription start failing
with some 'Privacy violation' response code. 'Oh sorry,' says the Service
Provider, 'but you asked for all calls to be anonymous and then carried on
trying to make non-anonymous calls. You'll need to tell all the users to
change their settings to anonymous'.

Whilst heamoraging money due to lost business, as none of their employees
can make calls, the subscriber makes the not unreasonable point that they
could tell all their users to switch to anonymous calls anyway, without
asking the Service Provider, so what is the point of their regulatory right
to an anonymity service ??

It has to work in a manner which is transparent and simple to the users and
subscriber. If you want a client-based solution to this service, then you
need to standardise a way of automatically informing & updating clients
about the subscription anonymity policy. Still doesn't work for forwarding
users.

The alternative is to accept that From/To modification in the network is
required to implement these services & no standardisation is required.

....Mark


> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> Sent: 16 April 2002 11:10
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
> sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> the current bis-09 text says:
> 
> The From header field allows for a display name. A UAC SHOULD use the
> display name  Anonymous , along with a syntactically correct, but
> otherwise meaningless URI (like sip:thisis@anonymous.invalid), if
> the identity of the client is to remain hidden.
> 
> this is all that is needed except that in my service the uri must be
> meaningful.
> 
> most UAs on the market support this.  so fail to see what the 
> issue is.
> 
> -- juha
> 
> 

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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<BODY>

<P><FONT SIZE=3D2>The client capability issue was for the case where =
the user does not wish to remain anonymous, but the subscriber =
does.</FONT>
</P>

<P><FONT SIZE=3D2>Without a network modification of the From field =
then:</FONT>
</P>

<P><FONT SIZE=3D2>1) The user includes their identity, and the session =
is rejected. There is no reject code which clearly indicates what =
action the user/UA should take. There will be no current UAs which =
would know to auto-retry with 'Anonymous', so this will require user =
intervention on all calls. Auto-retrys are no good for wireless =
anyway.</FONT></P>

<P><FONT SIZE=3D2>2) It was suggested that the client should have a =
'always anonymous' option which the user can set, so that all calls via =
this subscription will be anonymous. Again, current clients may or may =
not have this, so relying on this to provide the 'subscriber anonymity' =
service without the retrys of (1) is not such a good idea unless you =
have clearly told subscribers - before they subscribe - that such a =
capability is needed.</FONT></P>

<P><FONT SIZE=3D2>Service Providers need to be well aware of the client =
capabilities needed to implement the regulatory services they have to =
provide, so that they can make it clear to potential subscribers what =
those required client capabilities are.</FONT></P>

<P><FONT SIZE=3D2>Even so, I really can't see all the above being =
realistic. Imagine a subscriber goes to their service provider and =
requests anonymity. The first thing that happens is that all calls =
through the subscription start failing with some 'Privacy violation' =
response code. 'Oh sorry,' says the Service Provider, 'but you asked =
for all calls to be anonymous and then carried on trying to make =
non-anonymous calls. You'll need to tell all the users to change their =
settings to anonymous'.</FONT></P>

<P><FONT SIZE=3D2>Whilst heamoraging money due to lost business, as =
none of their employees can make calls, the subscriber makes the not =
unreasonable point that they could tell all their users to switch to =
anonymous calls anyway, without asking the Service Provider, so what is =
the point of their regulatory right to an anonymity service =
??</FONT></P>

<P><FONT SIZE=3D2>It has to work in a manner which is transparent and =
simple to the users and subscriber. If you want a client-based solution =
to this service, then you need to standardise a way of automatically =
informing &amp; updating clients about the subscription anonymity =
policy. Still doesn't work for forwarding users.</FONT></P>

<P><FONT SIZE=3D2>The alternative is to accept that From/To =
modification in the network is required to implement these services =
&amp; no standardisation is required.</FONT></P>

<P><FONT SIZE=3D2>....Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: jh@lohi.eng.song.fi [<A =
HREF=3D"mailto:jh@lohi.eng.song.fi">mailto:jh@lohi.eng.song.fi</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 16 April 2002 11:10</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; =
Mpierce1@aol.com;</FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; the current bis-09 text says:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The From header field allows for a display =
name. A UAC SHOULD use the</FONT>
<BR><FONT SIZE=3D2>&gt; display name&nbsp; Anonymous , along with a =
syntactically correct, but</FONT>
<BR><FONT SIZE=3D2>&gt; otherwise meaningless URI (like =
sip:thisis@anonymous.invalid), if</FONT>
<BR><FONT SIZE=3D2>&gt; the identity of the client is to remain =
hidden.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; this is all that is needed except that in my =
service the uri must be</FONT>
<BR><FONT SIZE=3D2>&gt; meaningful.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; most UAs on the market support this.&nbsp; so =
fail to see what the </FONT>
<BR><FONT SIZE=3D2>&gt; issue is.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- juha</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From daemon@optimus.ietf.org  Tue Apr 16 07:33:16 2002
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Date: Tue, 16 Apr 2002 13:58:39 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'Ben Campbell'"
	 <bcampbell@dynamicsoft.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mark Watson writes:

 > The client capability issue was for the case where the user does not wish to
 > remain anonymous, but the subscriber does.

what is a subscriber and how does it differ from a user?  where is the
term subscriber defined?

 > 1) The user includes their identity, and the session is rejected. There is
 > no reject code which clearly indicates what action the user/UA should take.
 > There will be no current UAs which would know to auto-retry with
 > 'Anonymous', so this will require user intervention on all calls.
 > Auto-retrys are no good for wireless anyway.

this whole hassle seems to originate from your subscriber business.  i
don't see any need to a subscriber that is not the same as the user.  is
this again some 3gpp thing?

 > 2) It was suggested that the client should have a 'always anonymous' option
 > which the user can set, so that all calls via this subscription will be
 > anonymous. Again, current clients may or may not have this, so relying on
 > this to provide the 'subscriber anonymity' service without the retrys of (1)
 > is not such a good idea unless you have clearly told subscribers - before
 > they subscribe - that such a capability is needed.

as i said, most client that i have see do already have this "always
anonymous" option.  i don't see any need to complicate the sip specs for
those clients that don't because the implementation is trivial. 

 > Service Providers need to be well aware of the client capabilities needed to
 > implement the regulatory services they have to provide, so that they can
 > make it clear to potential subscribers what those required client
 > capabilities are.

i have no problem with that.  already know we need to test the clients
and give recommendations to users.

 > Even so, I really can't see all the above being realistic. Imagine a
 > subscriber goes to their service provider and requests anonymity. The first
 > thing that happens is that all calls through the subscription start failing
 > with some 'Privacy violation' response code. 'Oh sorry,' says the Service
 > Provider, 'but you asked for all calls to be anonymous and then carried on
 > trying to make non-anonymous calls. You'll need to tell all the users to
 > change their settings to anonymous'.

this is no problem, because the user requests anonymity by using
Anonymous in the display string.  it is not a subscription option.  all
subscriptions look the same in this regard.

 > Whilst heamoraging money due to lost business, as none of their employees
 > can make calls, the subscriber makes the not unreasonable point that they
 > could tell all their users to switch to anonymous calls anyway, without
 > asking the Service Provider, so what is the point of their regulatory right
 > to an anonymity service ??

it makes no sense to try to emulate stupid pstn phones using with sip
UAs.  we can and should assume that a sip UA is a user configurable
beast.  therefore, the user itself can decide either permanently or on a
call by call basis if it wants anonymity or not.

 > It has to work in a manner which is transparent and simple to the users and
 > subscriber. If you want a client-based solution to this service, then you
 > need to standardise a way of automatically informing & updating clients
 > about the subscription anonymity policy. Still doesn't work for forwarding
 > users.

as i have said, there is no such thing as subscription anonymity policy.

-- juha


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@ns.ietf.org  Tue Apr 16 09:00:50 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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> what is a subscriber and how does it differ from a user?  where is the
> term subscriber defined?

This thread has been going for a while now, and as you said in the rest of
your message, the concept of a subscriber as separate from the user is
pretty important to the arguments, which is why I keep re-describing it. 

The subscriber is just one example of a person who has an interest in a
session who does not have a 'user agent' involved in the session to
represent them. My contention is that the easiest way to represent this
person in the session is through services in the network.

See
http://www1.ietf.org/mail-archive/working-groups/sip/current/msg04720.html
for a description of why the subscriber is different from the user.

In terms of definition, the definition from the European Directive on Data
Protection intelecommunications (97/66/EC) says:

(a) 'subscriber` shall mean any natural or legal person who or which is
party to a contract with the provider of publicly available
telecommunications services for the supply of such services;
(b) 'user` shall mean any natural person using a publicly available
telecommunications service, for private or business purposes, without
necessarily having subscribed to this service;


There is a lot of information about this and data protection in
telecommunications in general at http://www.cdt.org/privacy/eudirective/

Finally, I just spotted that this directive is due to be replaced by a
revision which is explicitly intended to cover new technologies
http://europa.eu.int/eur-lex/en/com/pdf/2000/en_500PC0385.pdf. I would not
advise any Service Provider to ignore this stuff.

Regards,

Mark

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<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>&gt; what is a subscriber and how does it differ =
from a user?&nbsp; where is the</FONT>
<BR><FONT SIZE=3D2>&gt; term subscriber defined?</FONT>
</P>

<P><FONT SIZE=3D2>This thread has been going for a while now, and as =
you said in the rest of your message, the concept of a subscriber as =
separate from the user is pretty important to the arguments, which is =
why I keep re-describing it. </FONT></P>

<P><FONT SIZE=3D2>The subscriber is just one example of a person who =
has an interest in a session who does not have a 'user agent' involved =
in the session to represent them. My contention is that the easiest way =
to represent this person in the session is through services in the =
network.</FONT></P>

<P><FONT SIZE=3D2>See <A =
HREF=3D"http://www1.ietf.org/mail-archive/working-groups/sip/current/msg=
04720.html" =
TARGET=3D"_blank">http://www1.ietf.org/mail-archive/working-groups/sip/c=
urrent/msg04720.html</A> for a description of why the subscriber is =
different from the user.</FONT></P>

<P><FONT SIZE=3D2>In terms of definition, the definition from the =
European Directive on Data Protection intelecommunications (97/66/EC) =
says:</FONT></P>

<P><FONT SIZE=3D2>(a) 'subscriber` shall mean any natural or legal =
person who or which is party to a contract with the provider of =
publicly available telecommunications services for the supply of such =
services;</FONT></P>

<P><FONT SIZE=3D2>(b) 'user` shall mean any natural person using a =
publicly available telecommunications service, for private or business =
purposes, without necessarily having subscribed to this =
service;</FONT></P>
<BR>

<P><FONT SIZE=3D2>There is a lot of information about this and data =
protection in telecommunications in general at <A =
HREF=3D"http://www.cdt.org/privacy/eudirective/" =
TARGET=3D"_blank">http://www.cdt.org/privacy/eudirective/</A></FONT></P>=


<P><FONT SIZE=3D2>Finally, I just spotted that this directive is due to =
be replaced by a revision which is explicitly intended to cover new =
technologies <A =
HREF=3D"http://europa.eu.int/eur-lex/en/com/pdf/2000/en_500PC0385.pdf" =
TARGET=3D"_blank">http://europa.eu.int/eur-lex/en/com/pdf/2000/en_500PC0=
385.pdf</A>. I would not advise any Service Provider to ignore this =
stuff.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

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</HTML>
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From daemon@ns.ietf.org  Tue Apr 16 09:09:26 2002
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Date: Tue, 16 Apr 2002 15:42:06 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
In-Reply-To: <A3C2399B2FACD411A54200508BE39C74054F7147@zwcwd00r.europe.nortel.com>
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Mark Watson writes:

 > The subscriber is just one example of a person who has an interest in a
 > session who does not have a 'user agent' involved in the session to
 > represent them. My contention is that the easiest way to represent this
 > person in the session is through services in the network.

i still fail to see the difference between a user and subscriber. in
order for the user to use my sip service, the user MUST first subscribe
the service.  the user may have any number of sip identities (uris)
associated with the subscription.

 > (a) 'subscriber` shall mean any natural or legal person who or which is
 > party to a contract with the provider of publicly available
 > telecommunications services for the supply of such services;

fine.

 > (b) 'user` shall mean any natural person using a publicly available
 > telecommunications service, for private or business purposes, without
 > necessarily having subscribed to this service;

i don't understand what the above could mean.

 a user can't use my sip service without being first subscribed to it,
 i.e., the user must have a username and a password that identify a
 valid subscription.  i don't care who made the subscription, but in
 general it is a VERY bad idea to pass a given username/password to many
 people. if you do it then there is no way to figure out who actually
 made a call.

-- juha


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From daemon@ns.ietf.org  Tue Apr 16 09:26:40 2002
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From: Juan-Carlos.Rojas@alcatel.fr
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To: Gonzalo.Camarillo@lmf.ericsson.se
Cc: sip@ietf.org
Date: Tue, 16 Apr 2002 14:57:25 +0200
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Hello Gonzalo,

Is there any project to create an internet-draft (or to update the existing
one) for the preconditions mechanism currently defined in the manyfolks
draft, but using SDPng (and so, XML based) ?

Thank you for your answer

Best regards
Juan Carlos



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From daemon@ns.ietf.org  Tue Apr 16 09:27:23 2002
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Subject: Re: [Sip] Re: Offer by the callee
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Date: Tue, 16 Apr 2002 14:53:40 +0200
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Hi Gonzalo,

Thank you very much, indeed this is compliant with the offer/answer draft.

Best regards
Juan Carlos





Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se> on 08/04/2002
16:22:20

To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
cc:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject:  Re: [Sip] Re: Offer by the callee


Hello Juan Carlos,

I have been thinking more about your scenario, and I think that sending
back in the first answer a c=0.0.0.0 line would be the right thing to
do.
This way, even though you have returned an answer with preconditions,
the other party will not perform resource reservation, since it does not
have an IP address to send media to.

Then, when you are ready to send an new offer, you will send it in an
UPDATE. The flow would look as follows:

                        INVITE (offer-1)
------------------------------------------------------------->
       183 (answer-to-offer-1, c=0.0.0.0)
<-------------------------------------------------------------
                             PRACK
------------------------------------------------------------->
                             200 PRACK
<-------------------------------------------------------------
                  UPDATE (offer-2)
<-------------------------------------------------------------
            200 UPDATE (answer-to-offer-2)
------------------------------------------------------------->

        <========== Resource Reservation ==========>

Best regards,

Gonzalo




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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Perhaps an example will help:

The subscriber is a company - the company has made a contract with you as
service provider for the provision of the service.

The user is an employee of the company - the user may be 'subscriber' in the
technical sense to your system, but they are not the subscriber as defined
in the definition below - they do not have any contract with you, the
contract is with the company.

If you insist on individual contracts with the employees of a company to
which you offer service, then you are equating user and subscriber, but I
doubt the company or its employees would be very happy with this mode of
operation.

...Mark

> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> Sent: 16 April 2002 13:42
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
> sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Mark Watson writes:
> 
>  > The subscriber is just one example of a person who has an 
> interest in a
>  > session who does not have a 'user agent' involved in the session to
>  > represent them. My contention is that the easiest way to 
> represent this
>  > person in the session is through services in the network.
> 
> i still fail to see the difference between a user and subscriber. in
> order for the user to use my sip service, the user MUST first 
> subscribe
> the service.  the user may have any number of sip identities (uris)
> associated with the subscription.
> 
>  > (a) 'subscriber` shall mean any natural or legal person 
> who or which is
>  > party to a contract with the provider of publicly available
>  > telecommunications services for the supply of such services;
> 
> fine.
> 
>  > (b) 'user` shall mean any natural person using a publicly available
>  > telecommunications service, for private or business 
> purposes, without
>  > necessarily having subscribed to this service;
> 
> i don't understand what the above could mean.
> 
>  a user can't use my sip service without being first subscribed to it,
>  i.e., the user must have a username and a password that identify a
>  valid subscription.  i don't care who made the subscription, but in
>  general it is a VERY bad idea to pass a given 
> username/password to many
>  people. if you do it then there is no way to figure out who actually
>  made a call.
> 
> -- juha
> 
> 

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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<P><FONT SIZE=3D2>Perhaps an example will help:</FONT>
</P>

<P><FONT SIZE=3D2>The subscriber is a company - the company has made a =
contract with you as service provider for the provision of the =
service.</FONT></P>

<P><FONT SIZE=3D2>The user is an employee of the company - the user may =
be 'subscriber' in the technical sense to your system, but they are not =
the subscriber as defined in the definition below - they do not have =
any contract with you, the contract is with the company.</FONT></P>

<P><FONT SIZE=3D2>If you insist on individual contracts with the =
employees of a company to which you offer service, then you are =
equating user and subscriber, but I doubt the company or its employees =
would be very happy with this mode of operation.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: jh@lohi.eng.song.fi [<A =
HREF=3D"mailto:jh@lohi.eng.song.fi">mailto:jh@lohi.eng.song.fi</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 16 April 2002 13:42</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; =
Mpierce1@aol.com;</FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark Watson writes:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; The subscriber is just one example =
of a person who has an </FONT>
<BR><FONT SIZE=3D2>&gt; interest in a</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; session who does not have a 'user =
agent' involved in the session to</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; represent them. My contention is =
that the easiest way to </FONT>
<BR><FONT SIZE=3D2>&gt; represent this</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; person in the session is through =
services in the network.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; i still fail to see the difference between a =
user and subscriber. in</FONT>
<BR><FONT SIZE=3D2>&gt; order for the user to use my sip service, the =
user MUST first </FONT>
<BR><FONT SIZE=3D2>&gt; subscribe</FONT>
<BR><FONT SIZE=3D2>&gt; the service.&nbsp; the user may have any number =
of sip identities (uris)</FONT>
<BR><FONT SIZE=3D2>&gt; associated with the subscription.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; (a) 'subscriber` shall mean any =
natural or legal person </FONT>
<BR><FONT SIZE=3D2>&gt; who or which is</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; party to a contract with the =
provider of publicly available</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; telecommunications services for the =
supply of such services;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; fine.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; (b) 'user` shall mean any natural =
person using a publicly available</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; telecommunications service, for =
private or business </FONT>
<BR><FONT SIZE=3D2>&gt; purposes, without</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; &gt; necessarily having subscribed to =
this service;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; i don't understand what the above could =
mean.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; a user can't use my sip service without =
being first subscribed to it,</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; i.e., the user must have a username and a =
password that identify a</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; valid subscription.&nbsp; i don't care =
who made the subscription, but in</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; general it is a VERY bad idea to pass a =
given </FONT>
<BR><FONT SIZE=3D2>&gt; username/password to many</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; people. if you do it then there is no way =
to figure out who actually</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; made a call.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- juha</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Chen, Xin (Xin)" <xchen2@lucent.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>,
        Juan-Carlos.Rojas@alcatel.fr
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] Re: Offer by the callee
Date: Tue, 16 Apr 2002 14:11:33 +0100
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Hi Gonzalo,

I don't want to open the debating again,

but this solution you provided assumes the QoS reservation protocol has to
use the IP address of both sides, it again reminds me about RSVP.

I thought we agreed that manyfolks is independent with any QoS reservation
protocol, therefore please don't provide solution on SIP level based on our
guessing on what happens IP level.

Thanks 

Xin Chen

Lucent Technologies
Tel: +44 1793 883137
Mobile: +44 7799 034668


-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: 08 April 2002 15:22
To: Juan-Carlos.Rojas@alcatel.fr
Cc: Jonathan Rosenberg; sip@ietf.org
Subject: Re: [Sip] Re: Offer by the callee


Hello Juan Carlos,

I have been thinking more about your scenario, and I think that sending
back in the first answer a c=0.0.0.0 line would be the right thing to
do. 
This way, even though you have returned an answer with preconditions,
the other party will not perform resource reservation, since it does not
have an IP address to send media to.

Then, when you are ready to send an new offer, you will send it in an
UPDATE. The flow would look as follows:

                        INVITE (offer-1)
------------------------------------------------------------->
       183 (answer-to-offer-1, c=0.0.0.0)    
<-------------------------------------------------------------
                             PRACK
------------------------------------------------------------->
                             200 PRACK
<-------------------------------------------------------------
                  UPDATE (offer-2)
<-------------------------------------------------------------
            200 UPDATE (answer-to-offer-2)
------------------------------------------------------------->

        <========== Resource Reservation ==========>

Best regards,

Gonzalo

Juan-Carlos.Rojas@alcatel.fr wrote:
> 
> Hi Gonzalo,
> 
> Thank you for your answer
> You are right, but sending a held SDP does not prevent the calling party
to
> start resource reservation, but only to send media.
> I agree with you that normally, a port number to zero indicates refusal of
> the media stream. Nevertheless, according to Offer/Answer draft, all port
> numbers equal to zero is used also to announce capabilities (section 9),
> and very often it is used to say just "don't use this media".
> 
> What I need here is to give to the answerer the possibility to say "Mr.the
> Offerer, I'm not rejecting your Invite, but I'd like to send you a new
> offer right now, so it is useless to start resource reservation before
> receiving this new offer"
> 
> Best regards
> Juan Carlos
> 
> Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>@ietf.org on
> 20/03/2002 05:01:02
> 
> Sent by:  sip-admin@ietf.org
> 
> To:   Juan-Carlos ROJAS/FR/ALCATEL@ALCATEL
> cc:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
> Subject:  [Sip] Re: Offer by the callee
> 
> Hi,
> 
> Setting the port number to zero means refusal of the stream, so you are
> risking that the caller sends a CANCEL, since you have refused all the
> media streams. A better solution would be to send a held SDP
> (a=inactive).
> 
> Regards,
> 
> Gonzalo
> 
> Juan-Carlos.Rojas@alcatel.fr wrote:
> >
> > Hello,
> >
> > I'm trying to connect the offer/answer, update and manyfolks drafts for
> the
> > following scenario:
> > - Alice sends an INVITE to Bob with "offer-1"
> > - Bob wants to make an "offer-2" to Alice, before the completion of the
> > session establishment
> > - Bob would like to ask Alice to start resource reservation only based
on
> > the acceptation of "offer-2", and not before (in fact Bob knows that it
> is
> > useless to start resource reservation based on offer-1, because he knows
> he
> > will send immediatly a new offer-2,  but he is obliged to wait for Prack
> > before)
> >
> > The Update draft defines the response 155 to allow the answerer (the
> callee
> > in our case) to request "Mr. the calling, please *send me* a new offer",
> > but there is no way for the same callee to announce "Mr. the calling, *I
> > want to send you* a new offer right now".
> > So, one possible way to achieve that is that the callee sends an answer
> to
> > the original offer but disabling all the media (in order to avoid to
> start
> > resource reservation), and next proposing a new offer, as described in
> the
> > call flow below (assume for the example that they are using end-to-end
> > resource reservation, but please don't generate a new debate on this
> > topic):
> >
> >    Alice
> > Bob
> >                                       INVITE (offer-1)
> >
----------------------------------------------------------------->
> >       183 (answer-to-offer-1, all port numbers to zero)
> >       <-----------------------------------------------------------------
> >                                          PRACK
> >
----------------------------------------------------------------->
> >                                       200 PRACK
> >       <-----------------------------------------------------------------
> >                  UPDATE (offer-2, status-type=e2e)
> >       <-----------------------------------------------------------------
> >                   200 UPDATE (answer-to-offer-2)
> >
----------------------------------------------------------------->
> >
> >    <========== Resource Reservation ==========>
> >
> >            UPDATE (current-status = desired-status)
> >
----------------------------------------------------------------->
> >                                       200 UPDATE
> >       <-----------------------------------------------------------------
> >                                         200 INVITE
> >       <-----------------------------------------------------------------
> >                                               ACK
> >
----------------------------------------------------------------->
> >
> > Is this scenario compliant with the drafts Offer/Answer, Update and
> > Manyfolks ?
> >
> > Thank you for your answer
> > Best regards
> > Juan Carlos
> 
> --
> Gonzalo Camarillo                    Phone :   +1 212 939 71 71
> Columbia University                  Mobile:  +358 40 702 35 35
> 472 Computer Science Building        Fax   :  +358  9 299 30 52
> 1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
> New York, NY 10027
> USA                              Gonzalo.Camarillo@ericsson.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension for Registering Non-Adjacent Contacts
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-path-03.txt
	Pages		: 11
	Date		: 15-Apr-02
	
The REGISTER function is used in a SIP system primarily to associate
a temporary contact address with an address-of-record.  This contact
is generally in the form of a URI, such as Contact:
&ltsip:alice@pc33.atlanta.com> and is generally dynamic and
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from the user's home network to the registered UA must traverse these
proxies.  The REGISTER method itself does not give us a mechanism to
discover and record this sequence of proxies in the registrar for
future use.  This document defines an extension header, 'Path' which
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Mark Watson" <mwatson@nortelnetworks.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        "Dean Willis" <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Message
  -----Original Message-----
  From: Mark Watson [mailto:mwatson@nortelnetworks.com]
  Sent: Tuesday, April 16, 2002 4:21 AM
  To: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
  Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft


  Ben wrote:
      Which then allows the user to decide whether or not to continue
according to policy, or just not make the call. Otherwise, the network seems
to be in the business of deciding that the user really does not mind if the
call is anonymous. That seems fairly benign in this particular case, but not
generically (what if the policy was _no_ anonymous calls.)
  A 'no anonymous calls' policy would be in breach of the users right to
make anonymous calls. It is not the network that is deciding that the user
does not mind if the call is anonymous, it is the subscriber, and they have
a legal right to do this. The subscribe must be able to control what
information the network reveals about them. The user does not have the right
to ask the network to reveal information that the subscriber does not want
revealed - although the use can of course reveal the information themselves.

  I was referring to a no anonymous calls policy on the part of the
subscriber.
      2) Even with a well-crafted UA, why is the _user_ expected to
re-configure their client because of the _subscriber's_ wishes. And when the
policy changes, how will we indicate to all the users that they should now
re-configure their clients again ?

      They don't have to. Doing so just cuts down on the
reject-due-to-policy-violation-oops-retry cycles. Now, I assume the user
must have some relationship with the subscriber (else how does he have
credentials), and the subscriber _could_ just tell the user. Not everything
has to be automatic.
  Frequent reject-due-to- policy-violation-oops-retry cycles are just
unaceptable for wireless environments. There must be a 'works first time'
solution. If you think the user needs to be informed that their call has
become anonymous, then fine, let's standardise this indication.

  A wireless network is one in which the network tends to excercise a lot
more control over the UA than might be typical for other environments. It
this case, the network is likely to be able to push configuration to the
phone itself (i.e. the config for "what default privacy option to use".
(This of course makes the b2bua changing From headers more feasible as well,
as wireless providers tend to have fewer qualms about asserting this level
of control.)
      I hope all the Service Providers wanting to launch SIP services have
got their list of client requirements worked out. If your subscriber asks
for this anonymity service they will not be impressed if they have to
upgrade/re-configure all their clients. If you didn't inform them in advance
that the clients needed these capabilities, then they could refuse to
upgrade and still demand the anonymity service.

      Well, once the idea of an anonymity service is fully formed
(presumably what we are doing here) it makes sense that it is a feature that
a client can support or not support. (Maybe even a supported or requires
tag?) Then it is reasonable for a service provider to say you can make
anonymous calls if your client supports anonymity. If a subscriber has a
policy of _only_ anonymous calls, and a user has a client that cannot make
anonymous calls, then he is SOL.  I don't see why this is any different than
any _other_ situation where an optional client feature might be required by
SP or subscriber policy. Again, the alternative is a presumption on the
networks part that it knows what the user _really_ means to do better than
the user does. Such an presumption on policy violations scares me in the
generic sense.
  It's different because the 'subscriber policy' feature is not just a
groovy addition to the service, but something that the Service Provide must
provide for the subscriber 'by a simple means' (in the words of the European
legislation) by which they mean I make a phone call to my Service Provider
and ask. Upgrading possibly hundreds of clients might not be considered a
'simple means'.

  Again, not a question of knowing what the user 'wants' - in this case the
subscribers wishe's overrule the users wishe's - we know the user wants to
reveal their identity but in this case the user is 'SOL'.

  ...Mark

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Tuesday, April 16, =
2002=20
  4:21 AM<BR><B>To:</B> 'Ben Campbell'; Chiou, Mark; Dean Willis;=20
  Mpierce1@aol.com; sip@ietf.org<BR><B>Subject:</B> RE: Summary of RE: =
[Sip]=20
  Comment, SIP Privacy draft<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D866021009-16042002>Ben wrote:</SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff><SPAN class=3D278085518-15042002><FONT =
face=3DArial size=3D2>Which=20
      then allows the user&nbsp;to decide whether or not to continue =
according=20
      to policy, or just not make the call. Otherwise, the network seems =
to=20
      be&nbsp;in&nbsp;the business of deciding that the user really does =

      not&nbsp;mind if the call is anonymous. That seems fairly benign =
in this=20
      particular case, but not generically (what if the policy was _no_=20
      anonymous=20
  =
calls.)</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV></BLOCKQUOTE></BLOC=
KQUOTE>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D866021009-16042002><FONT=20
  color=3D#0000ff>A 'no anonymous calls' policy would be in breach of =
the users=20
  right to make anonymous calls. It is not the network that is deciding =
that the=20
  user does not mind if the call is anonymous, it is the subscriber, and =
they=20
  have a legal right to do this. The subscribe must be able to control =
what=20
  information the network reveals about them. The user does not have the =
right=20
  to ask the network to reveal information that the subscriber does not =
want=20
  revealed - although the use can of course reveal the information=20
  themselves.<SPAN class=3D683130613-16042002><FONT face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D866021009-16042002><FONT=20
  color=3D#0000ff><SPAN=20
  class=3D683130613-16042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D866021009-16042002><FONT=20
  color=3D#0000ff><SPAN class=3D683130613-16042002><FONT face=3DArial =
size=3D2>I was=20
  referring to a no anonymous calls policy on the part of the=20
  subscriber.</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff>2) Even with a well-crafted UA, why is the _user_ =
expected=20
      to re-configure their client because of the _subscriber's_ wishes. =
And=20
      when the policy changes, how will we indicate to all the users =
that they=20
      should now re-configure their clients again ?<SPAN=20
      class=3D278085518-15042002><FONT face=3DArial=20
      size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff><SPAN=20
      =
class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff><SPAN class=3D278085518-15042002><FONT =
face=3DArial size=3D2>They=20
      don't have to. Doing so just cuts down on the=20
      reject-due-to-policy-violation-oops-retry cycles. Now, I assume =
the user=20
      must have some relationship with the subscriber (else =
how&nbsp;does he=20
      have credentials), and&nbsp;the subscriber _could_ just tell the =
user. Not=20
      everything has to be=20
      =
automatic.</FONT>&nbsp;</SPAN></FONT></SPAN></FONT></DIV></BLOCKQUOTE></B=
LOCKQUOTE>
  <DIV><SPAN class=3D455513518-15042002></SPAN><FONT size=3D1><FONT=20
  face=3DVerdana><FONT color=3D#0000ff><SPAN =
class=3D866021009-16042002>Frequent=20
  reject-due-to-</SPAN><SPAN=20
  class=3D866021009-16042002>&nbsp;policy-violation-oops-retry cycles =
are just=20
  unaceptable for wireless environments. There must be a 'works first =
time'=20
  solution. If you think the user needs to be informed that their call =
has=20
  become anonymous, then fine, let's standardise this indication.<SPAN=20
  class=3D683130613-16042002><FONT face=3DArial=20
  size=3D2>&nbsp;</FONT></SPAN></SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT size=3D1><FONT face=3DVerdana><FONT color=3D#0000ff><SPAN=20
  class=3D866021009-16042002><SPAN=20
  =
class=3D683130613-16042002></SPAN></SPAN></FONT></FONT></FONT>&nbsp;</DIV=
>
  <DIV><FONT size=3D1><FONT face=3DVerdana><FONT color=3D#0000ff><SPAN=20
  class=3D866021009-16042002><SPAN class=3D683130613-16042002><FONT =
face=3DArial=20
  size=3D2>A wireless network is one in which the network tends to =
excercise a lot=20
  more control over the UA than might be typical for other environments. =

  It&nbsp;this case, the network is likely to be able to push =
configuration to=20
  the phone itself (i.e. the config for "what default privacy option to=20
  use".&nbsp;(This of course makes the b2bua changing From headers more =
feasible=20
  as well, as wireless providers tend to have&nbsp;fewer&nbsp;qualms =
about=20
  asserting this level of=20
  control.)</FONT>&nbsp;</SPAN></SPAN></FONT></FONT></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px">
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff>I hope all the Service Providers wanting to launch =
SIP=20
      services have got their list of client requirements worked out. If =
your=20
      subscriber asks for this anonymity service they will not be =
impressed if=20
      they have to upgrade/re-configure all their clients. If you didn't =
inform=20
      them in advance that the clients needed these capabilities, then =
they=20
      could refuse to upgrade and still demand the anonymity =
service.<SPAN=20
      class=3D278085518-15042002><FONT face=3DArial=20
      size=3D2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
      <DIV><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><FONT=20
      color=3D#0000ff><SPAN=20
      =
class=3D278085518-15042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
      class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
      size=3D2>Well, once the idea of an anonymity service is fully =
formed=20
      (presumably what we are doing here) it makes sense&nbsp;that it is =
a=20
      feature that a client can support or not support.&nbsp;(Maybe even =
a=20
      supported or requires tag?) Then it is reasonable&nbsp;for a =
service=20
      provider to say you can make anonymous calls if your client =
supports=20
      anonymity.&nbsp;If a subscriber has a policy of _only_=20
      anonymous&nbsp;calls, and a user has a client that&nbsp;cannot =
make=20
      anonymous calls, then he is&nbsp;SOL.</FONT>&nbsp;<FONT =
face=3DArial size=3D2>=20
      I don't see why this is any different than any _other_ situation =
where an=20
      optional client feature might be required by SP or subscriber =
policy.=20
      Again, the alternative is a presumption on the networks part that =
it knows=20
      what the user _really_ means to do better than the user does. Such =
an=20
      presumption on policy violations scares me in the generic=20
      sense.&nbsp;<FONT face=3DVerdana size=3D1><SPAN=20
      =
class=3D866021009-16042002>&nbsp;</SPAN></FONT></FONT></SPAN></SPAN></FON=
T></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE>
  <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN =
class=3D866021009-16042002>It's different=20
  because the 'subscriber policy'&nbsp;feature is not just a groovy =
addition to=20
  the&nbsp;service,&nbsp;but something that the Service Provide must =
provide for=20
  the subscriber 'by a simple means' (in the words of the European =
legislation)=20
  by which they mean I make a phone call to my Service Provider and ask. =

  Upgrading possibl</SPAN></FONT></FONT></SPAN></SPAN></FONT><FONT =
face=3DVerdana=20
  size=3D1><SPAN class=3D455513518-15042002><SPAN =
class=3D278085518-15042002><FONT=20
  face=3DArial size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D866021009-16042002>y&nbsp;hundreds of clients might not be =
considered a=20
  'simple means'.<SPAN class=3D683130613-16042002><FONT face=3DArial=20
  =
size=3D2>&nbsp;</FONT></SPAN></SPAN></FONT></FONT></SPAN></SPAN></FONT></=
FONT><FONT=20
  color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN =
class=3D455513518-15042002><SPAN=20
  class=3D278085518-15042002><FONT face=3DArial size=3D2><FONT =
face=3DVerdana=20
  size=3D1><SPAN class=3D866021009-16042002><SPAN=20
  =
class=3D683130613-16042002>&nbsp;</SPAN></SPAN></FONT></FONT></SPAN></SPA=
N></FONT></FONT></DIV>
  <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  =
class=3D866021009-16042002></SPAN></FONT></FONT></SPAN></SPAN></FONT></FO=
NT>&nbsp;</DIV>
  <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN =
class=3D866021009-16042002>Again, not a=20
  question of knowing what the user 'wants' - in this case the =
subscribers=20
  wishe's overrule the users wishe's - we know the user wants to reveal =
their=20
  identity but in this case the user is=20
  'SOL'.</SPAN></FONT></FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  =
class=3D866021009-16042002></SPAN></FONT></FONT></SPAN></SPAN></FONT></FO=
NT>&nbsp;</DIV>
  <DIV><FONT color=3D#0000ff><FONT face=3DVerdana size=3D1><SPAN=20
  class=3D455513518-15042002><SPAN class=3D278085518-15042002><FONT =
face=3DArial=20
  size=3D2><FONT face=3DVerdana size=3D1><SPAN=20
  =
class=3D866021009-16042002>...Mark</SPAN></FONT></FONT></SPAN></SPAN></FO=
NT></FONT></DIV></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_00E7_01C1E51E.6964D540--


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From daemon@ns.ietf.org  Tue Apr 16 10:04:41 2002
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Message-ID: <15548.8990.883536.879546@harjus.eng.song.fi>
Date: Tue, 16 Apr 2002 16:11:58 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mark Watson writes:

 > The subscriber is a company - the company has made a contract with you as
 > service provider for the provision of the service.

 > The user is an employee of the company - the user may be 'subscriber' in the
 > technical sense to your system, but they are not the subscriber as defined
 > in the definition below - they do not have any contract with you, the
 > contract is with the company.

i don't see how "who pays for the service for a set of users" has any
relevance to the privacy discussion.

 > If you insist on individual contracts with the employees of a company to
 > which you offer service, then you are equating user and subscriber, but I
 > doubt the company or its employees would be very happy with this mode of
 > operation.

i do insist individual usernames and passwords.  if you do not have
that, then there is no way to figure out who made the calls, which is a
VERY big problem both for the company and for system in general.

-- juha


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From daemon@ns.ietf.org  Tue Apr 16 10:05:31 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "Rohan Mahy" <rohan@cisco.com>, <sip@ietf.org>, <sipping@ietf.org>,
        "James M. Polk" <jmpolk@cisco.com>
References: <4.1.20020415103657.016784e0@localhost>
Subject: Re: [Sip] Re: [Sipping] SIP/SIPPING Interim Meeting
Date: Tue, 16 Apr 2002 08:23:53 -0500
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I don't recall whether this has been explicitly mentioned, but the =
Pulver SIP Summit occurs in conjunction with Interop. So many of the SIP =
people of the world are already in Vegas this week for SIP Summit.  We =
might as well get together formally.

And yes, I was able to book a room at the Riviera, 0.6 miles from the =
convention center, for $90 a night, on the web yesterday afternoon.

--
Dean

  ----- Original Message -----=20
  From: James M. Polk=20
  To: Rohan Mahy ; sip@ietf.org ; sipping@ietf.org=20
  Sent: Monday, April 15, 2002 11:10 AM
  Subject: [Sip] Re: [Sipping] SIP/SIPPING Interim Meeting


  At 01:39 PM 4/13/2002 -0700, Rohan Mahy wrote:
  >Hi,
  >
  >I'd like to announce a joint SIP/SIPPING interim meeting May 6-7 in =
Las
  >Vegas, Nevada. =20

  You chose the week of InterOp Las Vegas, plan on having the meeting =
*at* the same location as that convention, and have really confirmed =
that there are rooms available at the hotel attached to the convention =
center?!

  Seems even if you've picked a location that does have the room, that =
you couldn't have picked a more hectic week or place to have it in =
(except maybe CES week in Januarys)...

  Or was this a plan to have during InterOp?

  >this date and location were changed (with AD approval)
  >as a fallback due to conflicts with the previously proposed time and =
place
  >(Boston, the following week).

  *************************************
  "People generally demand more respect for their own rights than they =
are willing to allow for others"


  James M. Polk
  Consulting Engineer
  Office of the CTO

  Cisco Systems
  2200 East President George Bush Turnpike=20
  Richardson, TX  75082 USA
  w) 972.813.5208
  f)  972.813.5280
  www.cisco.com=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I don't recall whether this has been =
explicitly=20
mentioned, but the Pulver SIP Summit occurs in conjunction with Interop. =
So many=20
of the SIP people of the world are already in Vegas this week for SIP=20
Summit.&nbsp; We might as well get together formally.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>And yes, I was able to book a room at =
the Riviera,=20
0.6 miles from the convention center, for $90 a night, on the web =
yesterday=20
afternoon.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>--<BR>Dean</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Djmpolk@cisco.com href=3D"mailto:jmpolk@cisco.com">James M. =
Polk</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A title=3Drohan@cisco.com=20
  href=3D"mailto:rohan@cisco.com">Rohan Mahy</A> ; <A =
title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">sip@ietf.org</A> ; <A =
title=3Dsipping@ietf.org=20
  href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Monday, April 15, 2002 =
11:10=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Sip] Re: [Sipping] =
SIP/SIPPING=20
  Interim Meeting</DIV>
  <DIV><FONT face=3DArial size=3D2></FONT><BR></DIV>At 01:39 PM =
4/13/2002 -0700,=20
  Rohan Mahy wrote:<BR>&gt;Hi,<BR>&gt;<BR>&gt;I'd like to announce a =
joint=20
  SIP/SIPPING interim meeting May 6-7 in Las<BR>&gt;Vegas, Nevada.&nbsp; =

  <BR><BR>You chose the week of<U> InterOp Las Vegas</U>, plan on having =
the=20
  meeting *at* the same location as that convention, and have really =
confirmed=20
  that there are rooms available at the hotel attached to the convention =

  center?!<BR><BR>Seems even if you've picked a location that does have =
the=20
  room, that you couldn't have picked a more hectic week or place to =
have it in=20
  (except maybe CES week in Januarys)...<BR><BR>Or was this a plan to =
have=20
  during InterOp?<BR><BR>&gt;this date and location were changed (with =
AD=20
  approval)<BR>&gt;as a fallback due to conflicts with the previously =
proposed=20
  time and place<BR>&gt;(Boston, the following week).<BR>
  <DIV align=3Dcenter>*************************************<BR>"People =
generally=20
  demand more respect for their own rights than they are willing to =
allow for=20
  others"<BR><BR></DIV>James M. Polk<BR>Consulting Engineer<BR>Office of =
the=20
  CTO<BR><BR>Cisco Systems<BR>2200 East President George Bush Turnpike=20
  <BR>Richardson, TX&nbsp; 75082 USA<BR>w) 972.813.5208<BR>f)&nbsp;=20
  972.813.5280<BR><A href=3D"http://www.cisco.com/"=20
  eudora=3D"autourl">www.cisco.com</A> </BLOCKQUOTE></BODY></HTML>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>, "Sipping \(E-mail\)" <sipping@ietf.org>
Date: Tue, 16 Apr 2002 08:30:13 -0500
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Subject: [Sip] Request: Posting Formats -- Use Plain Text
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I've noticed a lot of people posting messages in Microsoft Rich Text or
MHTML formats.

This is painful and hard to reply to, and makes it hard to follow the
thread of conversation. And it makes the messages nearly IMPOSSIBLE to
read on smaller-format devices.

Please:

1) Set your email program up to use plain text if at all possible. There
were guidelines for this on IETF Announce last year.

2) Don't just copy in the whole body of a message when you're responding
to a small section of it. Just include the relevant text and an
attribution to the original author -- like "Dean wrote:".

Thanks.

--
Dean

We could probably dig up the USENET etiquette FAQ if anyone would find
that useful.


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From daemon@ns.ietf.org  Tue Apr 16 10:15:16 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Juha wrote:
> i don't see how "who pays for the service for a set of users" has any
> relevance to the privacy discussion.
> 

Then you should take this up with the European Commission, and other
legislative organisations in the rest of the world.

The identity of the user is considered by them to (potentially) reveal
Personal Data of the subscriber. Personal Data is protected by 'Data
Protection' laws which you are free to disobey if you don't mind the fines.

[It's not 'who pays', is it who has legally contracted for provision of the
service (usually the same person who pays).]

> i do insist individual usernames and passwords.  if you do not have
> that, then there is no way to figure out who made the calls, 
> which is a
> VERY big problem both for the company and for system in general.

The requirement is not about usernames, passwords, UAs, SIP, packets, IP,
clients, proxies, networks etc. etc. This requirement is about services,
service providers, Personal Data, subscribers, users and the protection of
their data according to the law.

...Mark

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charset=3Dus-ascii">
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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
</HEAD>
<BODY>
<BR>

<P><FONT SIZE=3D2>Juha wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; i don't see how &quot;who pays for the service =
for a set of users&quot; has any</FONT>
<BR><FONT SIZE=3D2>&gt; relevance to the privacy discussion.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>Then you should take this up with the European =
Commission, and other legislative organisations in the rest of the =
world.</FONT></P>

<P><FONT SIZE=3D2>The identity of the user is considered by them to =
(potentially) reveal Personal Data of the subscriber. Personal Data is =
protected by 'Data Protection' laws which you are free to disobey if =
you don't mind the fines.</FONT></P>

<P><FONT SIZE=3D2>[It's not 'who pays', is it who has legally =
contracted for provision of the service (usually the same person who =
pays).]</FONT></P>

<P><FONT SIZE=3D2>&gt; i do insist individual usernames and =
passwords.&nbsp; if you do not have</FONT>
<BR><FONT SIZE=3D2>&gt; that, then there is no way to figure out who =
made the calls, </FONT>
<BR><FONT SIZE=3D2>&gt; which is a</FONT>
<BR><FONT SIZE=3D2>&gt; VERY big problem both for the company and for =
system in general.</FONT>
</P>

<P><FONT SIZE=3D2>The requirement is not about usernames, passwords, =
UAs, SIP, packets, IP, clients, proxies, networks etc. etc. This =
requirement is about services, service providers, Personal Data, =
subscribers, users and the protection of their data according to the =
law.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

</BODY>
</HTML>
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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Tue Apr 16 10:15:35 2002
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From: "Salva Rey Calatayud" <salreyca@teleco.upv.es>
To: sip@ietf.org
Date: Sun, 16 Apr 2000 15:48:06 +0200
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Hi Gonzalo,

	This is Salva Rey, from Siemens CT in Munich. How are you 
doing?

	I wanted to comment you a couple of things about manyfolks.


	For what I see in manyfolks it seems that two aspects are 
being defined jointly:

	1. QoS capability exchange signalling and interaction with the 
offer/answer model(through methods and headers)	

	2. QoS requirements (in SDP fields)

	Point #1 will be always managed by the UA, but maybe #2 is 
let to the application...

	this has never been mentioned, but are we aware of this?
	will manyfolks provide any support for QoS capability exchange 
signalling in case that the format for media stream descriptor is 
other than sdp, let's say here sdpng or xml?

	my concern is that maybe these two aspects are woven too 
tightly, and later on manyfolks cannot be used if a format different 
to sdp is used.

regards,
Salva


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Tue Apr 16 10:23:34 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 16 Apr 2002 14:43:49 +0100
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A 'no anonymous calls' policy would be in breach of the users right to make
anonymous calls. It is not the network that is deciding that the user does
not mind if the call is anonymous, it is the subscriber, and they have a
legal right to do this. The subscribe must be able to control what
information the network reveals about them. The user does not have the right
to ask the network to reveal information that the subscriber does not want
revealed - although the use can of course reveal the information themselves.

 
I was referring to a no anonymous calls policy on the part of the
subscriber.  

I know you were. I no anonymous calls policy on the part of the _subscriber_
would breach the _users_ right to make anonymous calls. 

Frequent reject-due-to- policy-violation-oops-retry cycles are just
unaceptable for wireless environments. There must be a 'works first time'
solution. If you think the user needs to be informed that their call has
become anonymous, then fine, let's standardise this indication. 
 
A wireless network is one in which the network tends to excercise a lot more
control over the UA than might be typical for other environments. It this
case, the network is likely to be able to push configuration to the phone
itself (i.e. the config for "what default privacy option to use". (This of
course makes the b2bua changing From headers more feasible as well, as
wireless providers tend to have fewer qualms about asserting this level of
control.)  
 

Indeed. If I have to specify special client behaviour not defined in
RFC2543, then I might as well assume bis compliance, and then I can modify
From/To without needing to keep state and modify them back on responses -
easy.
 
However, my personal view is that we should minimise the requirements on SIP
clients in order for them to work on the wireless network. The more
wireless-specific requirements or worse, requirements specific to a
particular wireless system, then the less SIP clients will be developed for
that environment, which is a bad thing.
 
The problem with the 'config push' option is that it makes it hard to
develop more complex policies on behalf of the subscriber (e.g. based on
destination), because the whole policy is dictated by the flexibility of the
configuration that you can push to the user, which is limited by what is
standardised.
 
OK, so we could use 'config push' for the regulatory requirement for an
'always anonymous' service, and do more complex things with B2BUAs. But I
would like a single solution and anyway you are arguing that From/To
modification is NEVER required.
 
...Mark

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<TITLE>Message</TITLE>

<META content="MSHTML 5.00.3315.2870" name=GENERATOR></HEAD>
<BODY>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV><FONT face=Verdana size=1><SPAN class=866021009-16042002><FONT 
    color=#0000ff>A 'no anonymous calls' policy would be in breach of the users 
    right to make anonymous calls. It is not the network that is deciding that 
    the user does not mind if the call is anonymous, it is the subscriber, and 
    they have a legal right to do this. The subscribe must be able to control 
    what information the network reveals about them. The user does not have the 
    right to ask the network to reveal information that the subscriber does not 
    want revealed - although the use can of course reveal the information 
    themselves.<SPAN class=683130613-16042002><FONT face=Arial 
    size=2>&nbsp;</FONT></SPAN></FONT></SPAN></FONT></DIV>
    <DIV><FONT face=Verdana size=1><SPAN class=866021009-16042002><FONT 
    color=#0000ff><SPAN 
    class=683130613-16042002></SPAN></FONT></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
    class=866021009-16042002><SPAN class=683130613-16042002><FONT face=Arial 
    size=2>I was referring to a no anonymous calls policy on the part of the 
    subscriber.</FONT>&nbsp;<SPAN 
    class=278513513-16042002>&nbsp;</SPAN></SPAN></SPAN></FONT></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT face=Verdana size=1><SPAN 
class=866021009-16042002><SPAN class=683130613-16042002><SPAN 
class=278513513-16042002>I know you were. I no anonymous calls policy on the 
part of the _subscriber_ would breach the _users_ right to&nbsp;make anonymous 
calls.&nbsp;</SPAN></SPAN></SPAN></FONT></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <BLOCKQUOTE dir=ltr 
    style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"></BLOCKQUOTE>
    <DIV><SPAN class=455513518-15042002></SPAN><FONT size=1><FONT 
    face=Verdana><FONT color=#0000ff><SPAN class=866021009-16042002>Frequent 
    reject-due-to-</SPAN><SPAN 
    class=866021009-16042002>&nbsp;policy-violation-oops-retry cycles are just 
    unaceptable for wireless environments. There must be a 'works first time' 
    solution. If you think the user needs to be informed that their call has 
    become anonymous, then fine, let's standardise this indication.<SPAN 
    class=683130613-16042002><FONT face=Arial 
    size=2>&nbsp;</FONT></SPAN></SPAN></FONT></FONT></FONT></DIV>
    <DIV><FONT size=1><FONT face=Verdana><FONT color=#0000ff><SPAN 
    class=866021009-16042002><SPAN 
    class=683130613-16042002></SPAN></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff><FONT size=1><FONT face=Verdana><SPAN 
    class=866021009-16042002><SPAN class=683130613-16042002><FONT face=Arial 
    size=2>A wireless network is one in which the network tends to excercise a 
    lot more control over the UA than might be typical for other environments. 
    It&nbsp;this case, the network is likely to be able to push configuration to 
    the phone itself (i.e. the config for "what default privacy option to 
    use".&nbsp;(This of course makes the b2bua changing From headers more 
    feasible as well, as wireless providers tend to have&nbsp;fewer&nbsp;qualms 
    about asserting this level of control.)</FONT>&nbsp;<SPAN 
    class=278513513-16042002>&nbsp;</SPAN></SPAN></SPAN></FONT></FONT></FONT></DIV>
    <DIV><FONT color=#0000ff><FONT size=1><FONT face=Verdana><SPAN 
    class=866021009-16042002><SPAN class=683130613-16042002><SPAN 
    class=278513513-16042002></SPAN></SPAN></SPAN></FONT></FONT></FONT>&nbsp;</DIV></BLOCKQUOTE></BLOCKQUOTE>
<DIV><FONT color=#0000ff><FONT size=1><FONT face=Verdana><SPAN 
class=866021009-16042002><SPAN class=683130613-16042002><SPAN 
class=278513513-16042002>Indeed. If I have to specify special client behaviour 
not defined in&nbsp;RFC2543, then I might as well assume bis compliance, and 
then I can modify From/To without needing to&nbsp;keep state and modify them 
back on responses - easy.</SPAN></SPAN></SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><FONT size=1><FONT face=Verdana><SPAN 
class=866021009-16042002><SPAN class=683130613-16042002><SPAN 
class=278513513-16042002></SPAN></SPAN></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=278513513-16042002>However, my personal view is that we should minimise 
the requirements on SIP clients in order for them to work on the wireless 
network. The more wireless-specific requirements or worse, requirements specific 
to a particular wireless system, then the less SIP clients will be developed for 
that environment, which is a bad thing.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=278513513-16042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=278513513-16042002>The 
problem with the 'config push' option is that it makes it hard to develop more 
complex policies on behalf of the subscriber (e.g. based on destination), 
because the whole policy is dictated by the flexibility of the configuration 
that you can push to the user, which is limited by what is 
standardised.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=278513513-16042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=278513513-16042002>OK, 
so we could use 'config push' for the regulatory requirement for an 'always 
anonymous' service, and do more complex things with B2BUAs. But I would like a 
single solution and anyway you are arguing that From/To modification is NEVER 
required.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=278513513-16042002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=278513513-16042002>...Mark</SPAN></FONT></DIV></BODY></HTML>

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Date: Tue, 16 Apr 2002 16:45:21 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
In-Reply-To: <A3C2399B2FACD411A54200508BE39C74054F714A@zwcwd00r.europe.nortel.com>
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Mark Watson writes:

 > The identity of the user is considered by them to (potentially) reveal
 > Personal Data of the subscriber. Personal Data is protected by 'Data
 > Protection' laws which you are free to disobey if you don't mind the fines.

huh.  i have never said that i don't honor someone's privacy wish.

 > The requirement is not about usernames, passwords, UAs, SIP, packets, IP,
 > clients, proxies, networks etc. etc. This requirement is about services,
 > service providers, Personal Data, subscribers, users and the protection of
 > their data according to the law.

fine and i don't have any problem with any of that.  the display string
does protect the privacy of the user according to his/her wish either on
call by call basis or for all calls.

-- juha


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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark"
	 <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
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Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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> fine and i don't have any problem with any of that.  the 
> display string
> does protect the privacy of the user according to his/her 
> wish either on
> call by call basis or for all calls.
> 

You have to honour the wishes of the SUBSCRIBER as well. I've explained why,
but that does not seem to be not enough for you, so look up the law
(97/66/EC) on the EU website (http://www.europa.eu.int/)

For clarity, I am not saying that I know this law to be 100% applicable to
SIP services - I am not a law court, so I could not do this - I am arguing
that it is likely that this or very similar requirements will apply, so we
should know how to deal with them.

The proposed revision of this document contains considerations about the
unsatisfactory situation with respect to email and Data Protection and urges
the electronic communications industry to pay more attention to Data
Protection. There are fairly explicit 'threats' that if things remain (in
their view) unsatisfactory, then further legislative measures will be
considered.

Paying attention to this stuff at this stage of system development is one
way that we *avoid* the need for *more* regulations to be imposed.

...Mark

------_=_NextPart_001_01C1E54E.B8FD9B84
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
</HEAD>
<BODY>
<BR>

<P><FONT SIZE=3D2>&gt; fine and i don't have any problem with any of =
that.&nbsp; the </FONT>
<BR><FONT SIZE=3D2>&gt; display string</FONT>
<BR><FONT SIZE=3D2>&gt; does protect the privacy of the user according =
to his/her </FONT>
<BR><FONT SIZE=3D2>&gt; wish either on</FONT>
<BR><FONT SIZE=3D2>&gt; call by call basis or for all calls.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>You have to honour the wishes of the SUBSCRIBER as =
well. I've explained why, but that does not seem to be not enough for =
you, so look up the law (97/66/EC) on the EU website (<A =
HREF=3D"http://www.europa.eu.int/" =
TARGET=3D"_blank">http://www.europa.eu.int/</A>)</FONT></P>

<P><FONT SIZE=3D2>For clarity, I am not saying that I know this law to =
be 100% applicable to SIP services - I am not a law court, so I could =
not do this - I am arguing that it is likely that this or very similar =
requirements will apply, so we should know how to deal with =
them.</FONT></P>

<P><FONT SIZE=3D2>The proposed revision of this document contains =
considerations about the unsatisfactory situation with respect to email =
and Data Protection and urges the electronic communications industry to =
pay more attention to Data Protection. There are fairly explicit =
'threats' that if things remain (in their view) unsatisfactory, then =
further legislative measures will be considered.</FONT></P>

<P><FONT SIZE=3D2>Paying attention to this stuff at this stage of =
system development is one way that we *avoid* the need for *more* =
regulations to be imposed.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

</BODY>
</HTML>
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From daemon@ns.ietf.org  Tue Apr 16 10:43:18 2002
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From: "David R. Oran" <oran@cisco.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>, <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 16 Apr 2002 09:59:59 -0400
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Mark,

I must admit that I am not following at all what you are saying. As far
as I know, the privacy laws and regulations are meant to protect
personal information disclosed by the person *to his service provider*,
either via the act of subscribing, and/or by the act of invoking the
service.

I think that the vast majority of the information the user puts in SIP
messages (either header or body) are no more "information disclosed to
the service provider" than the contents of files he FTPs, or dirty
pictures in a jpeg MIME body. They are information disclosed TO THE
RECIPIENT, which the service provider just happens to see going by, or
acts on to provide services for the user. If it were required that the
service provider not disclose this information, they'd all be getting
sued for not encrypting all the traffic going by. The exception are
things specifically for the consumption of the service provider, such as
proxy-authenticate tokens and the like.

I think the information that is of relevance to this discussion is stuff
that was *NOT* inserted by the user, but was instead inserted by the
service provider based on things the user had previously disclosed to
the service provider. These are the things which in fact do need some
level of protection, which is why we have this complicated applicability
statement on the privacy draft and why (I at least) would argue that the
only reasonable design is one in which any identifying information
inserted by the service provider be encrypted or otherwise protected
from disclosure to other parties (including the callee) unless the
customer explicitly requests it not to be.

This last, is of course, just a matter of the default setting of the
privacy: header, which can be handled by default UA configuration and
need not be the subject of protocol standardization.

Dave.


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Mark
Watson
Sent: Tuesday, April 16, 2002 9:31 AM
To: 'jh@lohi.eng.song.fi'
Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft




Juha wrote: 
> i don't see how "who pays for the service for a set of users" has any 
> relevance to the privacy discussion. 
> 
Then you should take this up with the European Commission, and other
legislative organisations in the rest of the world.
The identity of the user is considered by them to (potentially) reveal
Personal Data of the subscriber. Personal Data is protected by 'Data
Protection' laws which you are free to disobey if you don't mind the
fines.
[It's not 'who pays', is it who has legally contracted for provision of
the service (usually the same person who pays).]
> i do insist individual usernames and passwords.  if you do not have 
> that, then there is no way to figure out who made the calls, 
> which is a 
> VERY big problem both for the company and for system in general. 
The requirement is not about usernames, passwords, UAs, SIP, packets,
IP, clients, proxies, networks etc. etc. This requirement is about
services, service providers, Personal Data, subscribers, users and the
protection of their data according to the law.
...Mark 


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From daemon@ns.ietf.org  Tue Apr 16 11:41:15 2002
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: <brian.rosen@maroni.com>, <rohan@cisco.com>, <jo@ipdialog.com>
Date: Tue, 16 Apr 2002 10:01:30 -0500
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Subject: [Sip] Closure on Message draft? (was RE: Correction:SIP Message Draft Fast WGLC)
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As of this morning, the only feedback I have on the message draft is some
nits from Rohan. Are we done?

If I don't get additional negative feedback by midnight tonight (CDT), I
will cut a new release fixing the nits, and recommend we send it to the IESG
for IETF last call.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Friday, April 12, 2002 2:17 PM
> To: sip@ietf.org
> Cc: brian.rosen@maroni.com; rohan@cisco.com; jo@ipdialog.com;
> bcampbell@dynamicsoft.com
> Subject: Correction:SIP Message Draft Fast WGLC
>
>
>
> Oops.
>
> Please review :
>
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-message-02.txt
>
> Until the -02 makes it into the draft repository. They're not as fast as
> I'm wishing for (ok, light speed is too slow, while we're at it!)
>
> Thanks,
>
> --
> Dean
>
>
> Original message:
> -------------
> We need to quickly review the SIM Message Method Extension. This draft
> has been widely discussed, iterated several times in SIMPLE, and is on
> its 2nd revision in SIP. The draft does not appear to be controversial.
>
> See:
>
> http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt
>
> We expect to move to IETF last call next week, so please raise any
> issues immediately.
>
> Ben Campbell will coordinate for the draft editors.
>
> -----------------


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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/16/02 10:09:55 AM Eastern Daylight Time, oran@cisco.com 
writes:


> I think the information that is of relevance to this discussion is stuff
> that was *NOT* inserted by the user, but was instead inserted by the
> service provider based on things the user had previously disclosed to
> the service provider. These are the things which in fact do need some
> level of protection, which is why we have this complicated applicability
> statement on the privacy draft and why (I at least) would argue that the
> only reasonable design is one in which any identifying information
> inserted by the service provider be encrypted or otherwise protected
> from disclosure to other parties (including the callee) unless the
> customer explicitly requests it not to be.
> 
> 

I think you have to add to the first sentence above "and information inserted 
by the user that is required by the Service Provider as a prerequisite for 
service". If the service provider requires that the user enter a valid ID, 
then it must be protected the same as an ID that the Service Provider 
inserted.

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/16/02 10:09:55 AM Eastern Daylight Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I think the information that is of relevance to this discussion is stuff
<BR>that was *NOT* inserted by the user, but was instead inserted by the
<BR>service provider based on things the user had previously disclosed to
<BR>the service provider. These are the things which in fact do need some
<BR>level of protection, which is why we have this complicated applicability
<BR>statement on the privacy draft and why (I at least) would argue that the
<BR>only reasonable design is one in which any identifying information
<BR>inserted by the service provider be encrypted or otherwise protected
<BR>from disclosure to other parties (including the callee) unless the
<BR>customer explicitly requests it not to be.
<BR>
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I think you have to add to the first sentence above "and information inserted by the user that is required by the Service Provider as a prerequisite for service". If the service provider requires that the user enter a valid ID, then it must be protected the same as an ID that the Service Provider inserted.
<BR>
<BR>Mike
<BR></FONT></HTML>

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From daemon@ns.ietf.org  Tue Apr 16 12:52:53 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'David R. Oran'" <oran@cisco.com>, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Tue, 16 Apr 2002 17:17:54 +0100
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Dave,

I agree entirely with what you have written below, with three
clarifications:

1) If the user is *required* by the Service Provider to insert certain
information, then, for this discussion, it is the same as the Service
Provider inserting it - i.e. it doesn't matter which piece of physical
equipment inserts it if the information *has to be there* for the service to
work. In Juha's service the From field is like this, because unless you put
in 'anonymous' you MUST put your correct identity.

2) We must consider the *subscriber's* privacy request as well as the
user's. The subscriber may be a *different person* from the user. Either we
have to automatically get the subscriber's privacy policy to the UA, or the
network has to modify the fields which were network inserted, or inserted by
the user 'on behalf of' the network as per (1)

3) Personally, I go further than (1) and argue that information *expected*
(but not strictly *required*) by the Service Provider has similar status.
The SHOULD in the SIP spec wrt the contents of From/To is enough to mean
that a credible subscriber anonymity service should consider these fields.

So, my argument is that EITHER modification of the From/To fields may be
required in the network OR standardisation is required to reliably get the
subscriber's policy to the UA & deal with other similar problems.

To date I've seen no credible proposals for the latter, so I'm arguing we
should assume the former.

**The effect for the IETF SIP group** is only that it becomes a
'non-assumption' that the From/To fields are immutable within the network.
Future capabilities should be designed with this non-assumption wherever
possible, so that they are not broken by devices which do this. This is all
I am arguing about.

There will, of course, be some services which *are* broken by such devices.

...Mark

> -----Original Message-----
> From: David R. Oran [mailto:oran@cisco.com]
> Sent: 16 April 2002 15:00
> To: Watson, Mark [MDN05:EP10:EXCH]; sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> Mark,
> 
> I must admit that I am not following at all what you are 
> saying. As far
> as I know, the privacy laws and regulations are meant to protect
> personal information disclosed by the person *to his service 
> provider*,
> either via the act of subscribing, and/or by the act of invoking the
> service.
> 
> I think that the vast majority of the information the user puts in SIP
> messages (either header or body) are no more "information disclosed to
> the service provider" than the contents of files he FTPs, or dirty
> pictures in a jpeg MIME body. They are information disclosed TO THE
> RECIPIENT, which the service provider just happens to see going by, or
> acts on to provide services for the user. If it were required that the
> service provider not disclose this information, they'd all be getting
> sued for not encrypting all the traffic going by. The exception are
> things specifically for the consumption of the service 
> provider, such as
> proxy-authenticate tokens and the like.
> 
> I think the information that is of relevance to this 
> discussion is stuff
> that was *NOT* inserted by the user, but was instead inserted by the
> service provider based on things the user had previously disclosed to
> the service provider. These are the things which in fact do need some
> level of protection, which is why we have this complicated 
> applicability
> statement on the privacy draft and why (I at least) would 
> argue that the
> only reasonable design is one in which any identifying information
> inserted by the service provider be encrypted or otherwise protected
> from disclosure to other parties (including the callee) unless the
> customer explicitly requests it not to be.
> 
> This last, is of course, just a matter of the default setting of the
> privacy: header, which can be handled by default UA configuration and
> need not be the subject of protocol standardization.
> 
> Dave.
> 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Mark
> Watson
> Sent: Tuesday, April 16, 2002 9:31 AM
> To: 'jh@lohi.eng.song.fi'
> Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
> sip@ietf.org
> Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> 
> 
> 
> 
> Juha wrote: 
> > i don't see how "who pays for the service for a set of 
> users" has any 
> > relevance to the privacy discussion. 
> > 
> Then you should take this up with the European Commission, and other
> legislative organisations in the rest of the world.
> The identity of the user is considered by them to (potentially) reveal
> Personal Data of the subscriber. Personal Data is protected by 'Data
> Protection' laws which you are free to disobey if you don't mind the
> fines.
> [It's not 'who pays', is it who has legally contracted for 
> provision of
> the service (usually the same person who pays).]
> > i do insist individual usernames and passwords.  if you do not have 
> > that, then there is no way to figure out who made the calls, 
> > which is a 
> > VERY big problem both for the company and for system in general. 
> The requirement is not about usernames, passwords, UAs, SIP, packets,
> IP, clients, proxies, networks etc. etc. This requirement is about
> services, service providers, Personal Data, subscribers, users and the
> protection of their data according to the law.
> ...Mark 
> 
> 

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<P><FONT SIZE=3D2>Dave,</FONT>
</P>

<P><FONT SIZE=3D2>I agree entirely with what you have written below, =
with three clarifications:</FONT>
</P>

<P><FONT SIZE=3D2>1) If the user is *required* by the Service Provider =
to insert certain information, then, for this discussion, it is the =
same as the Service Provider inserting it - i.e. it doesn't matter =
which piece of physical equipment inserts it if the information *has to =
be there* for the service to work. In Juha's service the From field is =
like this, because unless you put in 'anonymous' you MUST put your =
correct identity.</FONT></P>

<P><FONT SIZE=3D2>2) We must consider the *subscriber's* privacy =
request as well as the user's. The subscriber may be a *different =
person* from the user. Either we have to automatically get the =
subscriber's privacy policy to the UA, or the network has to modify the =
fields which were network inserted, or inserted by the user 'on behalf =
of' the network as per (1)</FONT></P>

<P><FONT SIZE=3D2>3) Personally, I go further than (1) and argue that =
information *expected* (but not strictly *required*) by the Service =
Provider has similar status. The SHOULD in the SIP spec wrt the =
contents of From/To is enough to mean that a credible subscriber =
anonymity service should consider these fields.</FONT></P>

<P><FONT SIZE=3D2>So, my argument is that EITHER modification of the =
From/To fields may be required in the network OR standardisation is =
required to reliably get the subscriber's policy to the UA &amp; deal =
with other similar problems.</FONT></P>

<P><FONT SIZE=3D2>To date I've seen no credible proposals for the =
latter, so I'm arguing we should assume the former.</FONT>
</P>

<P><FONT SIZE=3D2>**The effect for the IETF SIP group** is only that it =
becomes a 'non-assumption' that the From/To fields are immutable within =
the network. Future capabilities should be designed with this =
non-assumption wherever possible, so that they are not broken by =
devices which do this. This is all I am arguing about.</FONT></P>

<P><FONT SIZE=3D2>There will, of course, be some services which *are* =
broken by such devices.</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: David R. Oran [<A =
HREF=3D"mailto:oran@cisco.com">mailto:oran@cisco.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 16 April 2002 15:00</FONT>
<BR><FONT SIZE=3D2>&gt; To: Watson, Mark [MDN05:EP10:EXCH]; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mark,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I must admit that I am not following at all =
what you are </FONT>
<BR><FONT SIZE=3D2>&gt; saying. As far</FONT>
<BR><FONT SIZE=3D2>&gt; as I know, the privacy laws and regulations are =
meant to protect</FONT>
<BR><FONT SIZE=3D2>&gt; personal information disclosed by the person =
*to his service </FONT>
<BR><FONT SIZE=3D2>&gt; provider*,</FONT>
<BR><FONT SIZE=3D2>&gt; either via the act of subscribing, and/or by =
the act of invoking the</FONT>
<BR><FONT SIZE=3D2>&gt; service.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I think that the vast majority of the =
information the user puts in SIP</FONT>
<BR><FONT SIZE=3D2>&gt; messages (either header or body) are no more =
&quot;information disclosed to</FONT>
<BR><FONT SIZE=3D2>&gt; the service provider&quot; than the contents of =
files he FTPs, or dirty</FONT>
<BR><FONT SIZE=3D2>&gt; pictures in a jpeg MIME body. They are =
information disclosed TO THE</FONT>
<BR><FONT SIZE=3D2>&gt; RECIPIENT, which the service provider just =
happens to see going by, or</FONT>
<BR><FONT SIZE=3D2>&gt; acts on to provide services for the user. If it =
were required that the</FONT>
<BR><FONT SIZE=3D2>&gt; service provider not disclose this information, =
they'd all be getting</FONT>
<BR><FONT SIZE=3D2>&gt; sued for not encrypting all the traffic going =
by. The exception are</FONT>
<BR><FONT SIZE=3D2>&gt; things specifically for the consumption of the =
service </FONT>
<BR><FONT SIZE=3D2>&gt; provider, such as</FONT>
<BR><FONT SIZE=3D2>&gt; proxy-authenticate tokens and the like.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I think the information that is of relevance to =
this </FONT>
<BR><FONT SIZE=3D2>&gt; discussion is stuff</FONT>
<BR><FONT SIZE=3D2>&gt; that was *NOT* inserted by the user, but was =
instead inserted by the</FONT>
<BR><FONT SIZE=3D2>&gt; service provider based on things the user had =
previously disclosed to</FONT>
<BR><FONT SIZE=3D2>&gt; the service provider. These are the things =
which in fact do need some</FONT>
<BR><FONT SIZE=3D2>&gt; level of protection, which is why we have this =
complicated </FONT>
<BR><FONT SIZE=3D2>&gt; applicability</FONT>
<BR><FONT SIZE=3D2>&gt; statement on the privacy draft and why (I at =
least) would </FONT>
<BR><FONT SIZE=3D2>&gt; argue that the</FONT>
<BR><FONT SIZE=3D2>&gt; only reasonable design is one in which any =
identifying information</FONT>
<BR><FONT SIZE=3D2>&gt; inserted by the service provider be encrypted =
or otherwise protected</FONT>
<BR><FONT SIZE=3D2>&gt; from disclosure to other parties (including the =
callee) unless the</FONT>
<BR><FONT SIZE=3D2>&gt; customer explicitly requests it not to =
be.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This last, is of course, just a matter of the =
default setting of the</FONT>
<BR><FONT SIZE=3D2>&gt; privacy: header, which can be handled by =
default UA configuration and</FONT>
<BR><FONT SIZE=3D2>&gt; need not be the subject of protocol =
standardization.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dave.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: sip-admin@ietf.org [<A =
HREF=3D"mailto:sip-admin@ietf.org">mailto:sip-admin@ietf.org</A>] On =
Behalf Of Mark</FONT>
<BR><FONT SIZE=3D2>&gt; Watson</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Tuesday, April 16, 2002 9:31 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'jh@lohi.eng.song.fi'</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; =
Mpierce1@aol.com;</FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: Summary of RE: [Sip] Comment, SIP =
Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Juha wrote: </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; i don't see how &quot;who pays for the =
service for a set of </FONT>
<BR><FONT SIZE=3D2>&gt; users&quot; has any </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; relevance to the privacy discussion. =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Then you should take this up with the European =
Commission, and other</FONT>
<BR><FONT SIZE=3D2>&gt; legislative organisations in the rest of the =
world.</FONT>
<BR><FONT SIZE=3D2>&gt; The identity of the user is considered by them =
to (potentially) reveal</FONT>
<BR><FONT SIZE=3D2>&gt; Personal Data of the subscriber. Personal Data =
is protected by 'Data</FONT>
<BR><FONT SIZE=3D2>&gt; Protection' laws which you are free to disobey =
if you don't mind the</FONT>
<BR><FONT SIZE=3D2>&gt; fines.</FONT>
<BR><FONT SIZE=3D2>&gt; [It's not 'who pays', is it who has legally =
contracted for </FONT>
<BR><FONT SIZE=3D2>&gt; provision of</FONT>
<BR><FONT SIZE=3D2>&gt; the service (usually the same person who =
pays).]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; i do insist individual usernames and =
passwords.&nbsp; if you do not have </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that, then there is no way to figure out =
who made the calls, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; which is a </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; VERY big problem both for the company and =
for system in general. </FONT>
<BR><FONT SIZE=3D2>&gt; The requirement is not about usernames, =
passwords, UAs, SIP, packets,</FONT>
<BR><FONT SIZE=3D2>&gt; IP, clients, proxies, networks etc. etc. This =
requirement is about</FONT>
<BR><FONT SIZE=3D2>&gt; services, service providers, Personal Data, =
subscribers, users and the</FONT>
<BR><FONT SIZE=3D2>&gt; protection of their data according to the =
law.</FONT>
<BR><FONT SIZE=3D2>&gt; ...Mark </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <Mpierce1@aol.com>, <oran@cisco.com>, <mwatson@nortelnetworks.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/16/02 10:09:55 AM Eastern Daylight Time,
oran@cisco.com writes:
---------
I think the information that is of relevance to this discussion is stuff
that was *NOT* inserted by the user, but was instead inserted by the
service provider based on things the user had previously disclosed to
the service provider. These are the things which in fact do need some
level of protection, which is why we have this complicated applicability
statement on the privacy draft and why (I at least) would argue that the
only reasonable design is one in which any identifying information
inserted by the service provider be encrypted or otherwise protected
from disclosure to other parties (including the callee) unless the
customer explicitly requests it not to be.
__________

I think Dave's generally right, with the one caveat noted by Mark below:

And Mark responded:
------------------
I think you have to add to the first sentence above "and information
inserted by the user that is required by the Service Provider as a
prerequisite for service". If the service provider requires that the
user enter a valid ID, then it must be protected the same as an ID that
the Service Provider inserted.
------------------

I agree mostly with Mark here -- which is why I'm continuing to push
back on an approach that "requires" the From: field to be specific
user-identifying information or suggests that it is more significant
than a "display name" which might optionally be used by some nodes for
some sort of lightweight filtering and displayed at the end point.

From bis:

-----
8.1.1.3 From The From header field indicates the logical identity of the
initiator of the request, possibly the user's address-of-record. Like
the To header field, it contains a URI and optionally a display name. It
is used by SIP elements to determine which processing rules to apply to
a request (for example, automatic call rejection). As such, it is very
important that the From URI not contain IP addresses or the FQDN of the
host on which the UA is running, since these are not logical names. 

The From header field allows for a display name. A UAC SHOULD use the
display name "Anonymous", along with a syntactically correct, but
otherwise meaningless URI (like sip:thisis@anonymous.invalid), if the
identity of the client is to remain hidden. 
-----

Note that the strongest requirements statement about usage here is a
"SHOULD" class statement on anonymity, and guidelines include "possibly
the user's address-of-record" on content in general. The SIP spec does
NOT mandate that the From field: contain any specific information. This
is in some measure intended to prevent the regulatoryistic (I just made
that up) abuse of the field. 

Let us also consider the phrase "very important that the From URI not
contain IP addresses or the FQDN of the host on which the UA is running,
since these are not logical names." This clearly indicates that the From
field is intended to provide a LOGICAL reference, not a DIRECT
reference. "Logical" is a broad term, and implies that the field should
be completed in a manner logically consistent with the role in which the
completor expects to use that field or have that field used by target
nodes.

--
Dean


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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, Mpierce1@aol.com,
        oran@cisco.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Dean wrote
> From bis:
> 
> -----
> 8.1.1.3 From The From header field ...
> It is used by SIP elements to determine which processing rules 
> to apply to a request (for example, automatic call rejection)....
> -----
> 
> Note that the strongest requirements statement about usage here is a
> "SHOULD" class statement on anonymity, and guidelines include 
> "possibly
> the user's address-of-record" on content in general. The SIP spec does
> NOT mandate that the From field: contain any specific 
> information. This
> is in some measure intended to prevent the regulatoryistic (I 
> just made
> that up) abuse of the field. 

It doesn't mandate, sure, but the *implication* is important. The fact that
it might be used for services, processing rules, etc., gives (some) people a
strong feeling that it ought to be accurate.

Of course, on the Internet, no one can hear you scream 'you lied about your
identity!'

In a Service Provider environment people have different expectations, which
is why even SHOULD level requirements like that above lead to a feeling that
the Service Provider must check the value of the field.

...Mark

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<TITLE>RE: Summary of RE: [Sip] Comment, SIP Privacy draft</TITLE>
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<BR>
<BR>

<P><FONT SIZE=3D2>Dean wrote</FONT>
<BR><FONT SIZE=3D2>&gt; From bis:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----</FONT>
<BR><FONT SIZE=3D2>&gt; 8.1.1.3 From The From header field ...</FONT>
<BR><FONT SIZE=3D2>&gt; It is used by SIP elements to determine which =
processing rules </FONT>
<BR><FONT SIZE=3D2>&gt; to apply to a request (for example, automatic =
call rejection)....</FONT>
<BR><FONT SIZE=3D2>&gt; -----</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Note that the strongest requirements statement =
about usage here is a</FONT>
<BR><FONT SIZE=3D2>&gt; &quot;SHOULD&quot; class statement on =
anonymity, and guidelines include </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;possibly</FONT>
<BR><FONT SIZE=3D2>&gt; the user's address-of-record&quot; on content =
in general. The SIP spec does</FONT>
<BR><FONT SIZE=3D2>&gt; NOT mandate that the From field: contain any =
specific </FONT>
<BR><FONT SIZE=3D2>&gt; information. This</FONT>
<BR><FONT SIZE=3D2>&gt; is in some measure intended to prevent the =
regulatoryistic (I </FONT>
<BR><FONT SIZE=3D2>&gt; just made</FONT>
<BR><FONT SIZE=3D2>&gt; that up) abuse of the field. </FONT>
</P>

<P><FONT SIZE=3D2>It doesn't mandate, sure, but the *implication* is =
important. The fact that it might be used for services, processing =
rules, etc., gives (some) people a strong feeling that it ought to be =
accurate.</FONT></P>

<P><FONT SIZE=3D2>Of course, on the Internet, no one can hear you =
scream 'you lied about your identity!'</FONT>
</P>

<P><FONT SIZE=3D2>In a Service Provider environment people have =
different expectations, which is why even SHOULD level requirements =
like that above lead to a feeling that the Service Provider must check =
the value of the field.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

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</HTML>
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From daemon@ns.ietf.org  Tue Apr 16 15:07:00 2002
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
References: <12f.fe535f3.29ed9446@aol.com>
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> I think the information that is of relevance to this discussion is stuff 
>   that was *NOT* inserted by the user, but was instead inserted by the 
>   service provider based on things the user had previously disclosed to 
>   the service provider. These are the things which in fact do need some 
>   level of protection, which is why we have this complicated applicability 
>   statement on the privacy draft and why (I at least) would argue that the 
>   only reasonable design is one in which any identifying information 
>   inserted by the service provider be encrypted or otherwise protected 
>   from disclosure to other parties (including the callee) unless the 
>   customer explicitly requests it not to be. 
> 
> 
> 
> I think you have to add to the first sentence above "and information inserted by the user that is required by the
> Service Provider as a prerequisite for service". If the service provider requires that the user enter a valid ID, then it
> must be protected the same as an ID that the Service Provider inserted. 

No reasonable provider should put that requirement on From, for the
reasons Dean mentioned. This leaves the proxy-authentication data.
Interesting question is whether the proxy-authentication data needs to
be stripped after it has been "used", as it may well apply to several
down-stream proxies.

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From daemon@ns.ietf.org  Tue Apr 16 15:40:46 2002
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From: "ruhiya  mahalati" <ruhiya@rediffmail.com>
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Subject: [Sip] SIP protocol failure scenarios
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Hi,

I am a graduate student and am working on the control memory, 
design and proof of correctness for SIP as my course project. I 
had a few doubts and would be grateful if anyone could answer 
them.

1: How is a duplicate BYE or a CANCEL generated, is it at timeout 
or only whenever the application requests?

2: Are there any cases may be because of lost messages or delayed 
or incorrect messages which may result in the failure of the 
protocol ie exceptional cases which the protocol is not capable of 
handling.

3:CSEQ feild is incremented for each request within a dialog.Is it 
increased for the responses also? and will the retansmission of a 
request contain the same CSEQ value or an incremented one?

thanks and regards,
ruhiya



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From daemon@optimus.ietf.org  Tue Apr 16 21:08:43 2002
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Date: Tue, 16 Apr 2002 17:38:33 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: Mark Watson <mwatson@nortelnetworks.com>
cc: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, <Mpierce1@aol.com>,
        <sip@ietf.org>
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mark,

I have a real problem with the way you have defined or envisioned a
privacy service.  Please see below...

On Tue, 16 Apr 2002, Mark Watson wrote:
> The client capability issue was for the case where the user does not wish to
> remain anonymous, but the subscriber does.
>
> Without a network modification of the From field then:
>
> 1) The user includes their identity, and the session is rejected. There is
> no reject code which clearly indicates what action the user/UA should take.
> There will be no current UAs which would know to auto-retry with
> 'Anonymous', so this will require user intervention on all calls.
> Auto-retrys are no good for wireless anyway.
>
> 2) It was suggested that the client should have a 'always anonymous' option
> which the user can set, so that all calls via this subscription will be
> anonymous. Again, current clients may or may not have this, so relying on
> this to provide the 'subscriber anonymity' service without the retrys of (1)
> is not such a good idea unless you have clearly told subscribers - before
> they subscribe - that such a capability is needed.
>
> Service Providers need to be well aware of the client capabilities needed to
> implement the regulatory services they have to provide, so that they can
> make it clear to potential subscribers what those required client
> capabilities are.
>
> Even so, I really can't see all the above being realistic. Imagine a
> subscriber goes to their service provider and requests anonymity.

A user who wants privacy of their self-revealed information has two very
reasonable choices, both of which are possible using baseline SIP (either
2543 or 3261).  Note that this is a very separate problem from
network-asserted identity used for PSTN interop, call trace, and call
return services.

1) The subscriber can configure their user agent not to reveal most types
of private information.

2) A provider can insert an anonymizing B2BUA (with or without media
anonymity) into the signaling path before forwarding outside its network.

I do not buy the argument that we need the network to provide this
service, but that it is somehow a burden for the network to implement this
service using a B2BUA.

thanks,
-rohan


> The first
> thing that happens is that all calls through the subscription start failing
> with some 'Privacy violation' response code. 'Oh sorry,' says the Service
> Provider, 'but you asked for all calls to be anonymous and then carried on
> trying to make non-anonymous calls. You'll need to tell all the users to
> change their settings to anonymous'.
>
> Whilst heamoraging money due to lost business, as none of their employees
> can make calls, the subscriber makes the not unreasonable point that they
> could tell all their users to switch to anonymous calls anyway, without
> asking the Service Provider, so what is the point of their regulatory right
> to an anonymity service ??
>
> It has to work in a manner which is transparent and simple to the users and
> subscriber. If you want a client-based solution to this service, then you
> need to standardise a way of automatically informing & updating clients
> about the subscription anonymity policy. Still doesn't work for forwarding
> users.
>
> The alternative is to accept that From/To modification in the network is
> required to implement these services & no standardisation is required.
>
> ....Mark
>
>
> > -----Original Message-----
> > From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> > Sent: 16 April 2002 11:10
> > To: Watson, Mark [MDN05:EP10:EXCH]
> > Cc: 'Ben Campbell'; Chiou, Mark; Dean Willis; Mpierce1@aol.com;
> > sip@ietf.org
> > Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
> >
> >
> > the current bis-09 text says:
> >
> > The From header field allows for a display name. A UAC SHOULD use the
> > display name  Anonymous , along with a syntactically correct, but
> > otherwise meaningless URI (like sip:thisis@anonymous.invalid), if
> > the identity of the client is to remain hidden.
> >
> > this is all that is needed except that in my service the uri must be
> > meaningful.
> >
> > most UAs on the market support this.  so fail to see what the
> > issue is.
> >
> > -- juha
> >
> >
>


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From daemon@optimus.ietf.org  Tue Apr 16 23:13:16 2002
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Subject: was Re: Screened FromL and Privacy
Subject: was Re: Summary of RE: [Sip] Comment, SIP Privacy draft
Subject: was lots of others, too

All this discussion of proxy modification of the From and To headers,
and the need for a B2BUA, seems to me to be a short-term problem only.

RFC3261 (2543bis-09) specifies matching of dialog-ids based only
on Call-ID and tags, not on display-name or on addr-spec.  So
when all endpoints are compliant with RFC3261, there will be no
problem with a proxy modifying the display-name or addr-spec in
the From header.

It is only during a transition period, when endpoints that conform
to RFC2543 are supported as well as newer ones.  How long will
this be?  How long will a proxy need to handle a UAC that doesn't
supply a Contact header?  How long will a proxy need to handle
a Record-Route header without the "lr" tag?  How long will a
UAS need to handle a BYE received prior to sending 2xx?  How long 
will the comparison be coded into the proxies and UAs looking 
for "z9hG4bK"?

In all cases, my hope is not too long.  And, at the point when
these "conversion aids" no longer need to be present in SIP
implementations, it will be safe for proxies to modify the From
header to provide anonymity, if the subscriber, or the user, or the
regulator, (or others?), says that it should.

In cases like 3GPP, where (IIRC) release 5 does not include
IMS connections outside of the wireless operator's network, the
problem is even simpler.  Since 3GPP required RFC3261 everywhere,
modification of the From header can be done immediately.  I would
hope this transition period and the need for "conversion aids" would
be over before 3GPP release 6 is deployed.

My conclusion from all of this is that 3GPP can require the UE to
insert a valid user identifier in the From header, and the S-CSCF can
change it if privacy is desired, on a session-by-session basis,
based on whatever rules the S-CSCF wants to follow.

Comments?

Bill Marshall
wtm@research.att.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Message-ID: <A3C2399B2FACD411A54200508BE39C74054F7157@zwcwd00r.europe.nortel.com>
From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>,
        "'Ben Campbell'"
	 <bcampbell@dynamicsoft.com>,
        "Chiou, Mark" <MChiou@Santera.com>,
        Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com,
        sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Wed, 17 Apr 2002 10:12:49 +0100
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Rohan wrote:

> I do not buy the argument that we need the network to provide this
> service, but that it is somehow a burden for the network to 
> implement this
> service using a B2BUA.
> 

This is not what I have been arguing. I agree that this type of anonymity
service will require more than a plain proxy (which by some definitions
means it's a B2BUA).

I know this will break (some) services.

I want to minimise the (future) services that break, by avoiding the
assumption that such devices do not exist.

Regards...Mark

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<P><FONT SIZE=3D2>Rohan wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; I do not buy the argument that we need the =
network to provide this</FONT>
<BR><FONT SIZE=3D2>&gt; service, but that it is somehow a burden for =
the network to </FONT>
<BR><FONT SIZE=3D2>&gt; implement this</FONT>
<BR><FONT SIZE=3D2>&gt; service using a B2BUA.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>This is not what I have been arguing. I agree that =
this type of anonymity service will require more than a plain proxy =
(which by some definitions means it's a B2BUA).</FONT></P>

<P><FONT SIZE=3D2>I know this will break (some) services.</FONT>
</P>

<P><FONT SIZE=3D2>I want to minimise the (future) services that break, =
by avoiding the assumption that such devices do not exist.</FONT>
</P>

<P><FONT SIZE=3D2>Regards...Mark</FONT>
</P>

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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Apr 17 06:18:15 2002
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From: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
Organization: OY LM Ericsson AB
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To: William Marshall <wtm@research.att.com>
CC: sip@ietf.org
Subject: Re: [Sip] z9hG4bK forever ???
References: <200204170247.WAA40877@fish.research.att.com>
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Hi Bill:

This is just to state that you made a very good summary of the situation
with 3GPP and I agree with your final conclusion.

If we don't go for such a solution we may risk the case when neither the
user nor the network (or regulator) required privacy. If we require the
endpoints to start sessions anonymously, just in case...  then the 
session will be anonymous even when not needed.

Therefore, I would like to see a consensus on Bill's proposal: Users
identify theirselves in the From (UAC), the proxy MAY anonymize the 
From if the policy (or the regulator) requires it.

/Miguel

William Marshall wrote:
> 
> Subject: was Re: Screened FromL and Privacy
> Subject: was Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> Subject: was lots of others, too
> 
> All this discussion of proxy modification of the From and To headers,
> and the need for a B2BUA, seems to me to be a short-term problem only.
> 
> RFC3261 (2543bis-09) specifies matching of dialog-ids based only
> on Call-ID and tags, not on display-name or on addr-spec.  So
> when all endpoints are compliant with RFC3261, there will be no
> problem with a proxy modifying the display-name or addr-spec in
> the From header.
> 
> It is only during a transition period, when endpoints that conform
> to RFC2543 are supported as well as newer ones.  How long will
> this be?  How long will a proxy need to handle a UAC that doesn't
> supply a Contact header?  How long will a proxy need to handle
> a Record-Route header without the "lr" tag?  How long will a
> UAS need to handle a BYE received prior to sending 2xx?  How long
> will the comparison be coded into the proxies and UAs looking
> for "z9hG4bK"?
> 
> In all cases, my hope is not too long.  And, at the point when
> these "conversion aids" no longer need to be present in SIP
> implementations, it will be safe for proxies to modify the From
> header to provide anonymity, if the subscriber, or the user, or the
> regulator, (or others?), says that it should.
> 
> In cases like 3GPP, where (IIRC) release 5 does not include
> IMS connections outside of the wireless operator's network, the
> problem is even simpler.  Since 3GPP required RFC3261 everywhere,
> modification of the From header can be done immediately.  I would
> hope this transition period and the need for "conversion aids" would
> be over before 3GPP release 6 is deployed.
> 
> My conclusion from all of this is that 3GPP can require the UE to
> insert a valid user identifier in the From header, and the S-CSCF can
> change it if privacy is desired, on a session-by-session basis,
> based on whatever rules the S-CSCF wants to follow.
> 
> Comments?
> 
> Bill Marshall
> wtm@research.att.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Miguel-Angel Garcia                     Oy LM Ericsson AB
                                        Jorvas, Finland
mailto:Miguel.A.Garcia@ericsson.com     Phone:  +358 9 299 3553
mailto:Miguel.A.Garcia@piuha.net        Mobile: +358 40 5140002

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Apr 17 06:24:27 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] z9hG4bK forever ???
Date: Wed, 17 Apr 2002 10:56:14 +0100
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Bill,

Indeed, this approach seems good to me too. But expect people to argue the
following points:
1) Proxies as defined in RFC3261 cannot modify the From field. Sure, it is a
small extension, but strictly speaking it is more than a pure proxy does.
2) This modification does break some services. It would be best to avoid it
except when necessary e.g. if the *user* requires privacy, then still put
Anonymous in the From field.
3) verification of the From field is either 'required for consistent service
in a Service Provider environment' (myself, probably most of 3GPP, Juha,
others ? ...) OR 'contrary to the definition of the From field as an
end-to-end user asserted field' (Dean, Dave, others ? ...) - we have no
consensus on this.

The only compromise I can see on this is:
a) Those who want to verify the From: field do
b) Those who don't want to don't
c) In the absence of trust relationships the From: field is completely
untrusted (like everything else!). If you have a trust relationship, then as
part of this you will agree whether From: can be trusted or not.
d) The contents of the From field MAY be modified by devices in the network.
So we should design new services and features with this assumption where
possible

(d) is a logical consequence of (a) - don't try and argue out (d) unless you
think you can exclude (a) too.

...Mark

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: 17 April 2002 03:48
> To: sip@ietf.org
> Subject: [Sip] z9hG4bK forever ???
> 
> 
> Subject: was Re: Screened FromL and Privacy
> Subject: was Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> Subject: was lots of others, too
> 
> All this discussion of proxy modification of the From and To headers,
> and the need for a B2BUA, seems to me to be a short-term problem only.
> 
> RFC3261 (2543bis-09) specifies matching of dialog-ids based only
> on Call-ID and tags, not on display-name or on addr-spec.  So
> when all endpoints are compliant with RFC3261, there will be no
> problem with a proxy modifying the display-name or addr-spec in
> the From header.
> 
> It is only during a transition period, when endpoints that conform
> to RFC2543 are supported as well as newer ones.  How long will
> this be?  How long will a proxy need to handle a UAC that doesn't
> supply a Contact header?  How long will a proxy need to handle
> a Record-Route header without the "lr" tag?  How long will a
> UAS need to handle a BYE received prior to sending 2xx?  How long 
> will the comparison be coded into the proxies and UAs looking 
> for "z9hG4bK"?
> 
> In all cases, my hope is not too long.  And, at the point when
> these "conversion aids" no longer need to be present in SIP
> implementations, it will be safe for proxies to modify the From
> header to provide anonymity, if the subscriber, or the user, or the
> regulator, (or others?), says that it should.
> 
> In cases like 3GPP, where (IIRC) release 5 does not include
> IMS connections outside of the wireless operator's network, the
> problem is even simpler.  Since 3GPP required RFC3261 everywhere,
> modification of the From header can be done immediately.  I would
> hope this transition period and the need for "conversion aids" would
> be over before 3GPP release 6 is deployed.
> 
> My conclusion from all of this is that 3GPP can require the UE to
> insert a valid user identifier in the From header, and the S-CSCF can
> change it if privacy is desired, on a session-by-session basis,
> based on whatever rules the S-CSCF wants to follow.
> 
> Comments?
> 
> Bill Marshall
> wtm@research.att.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<P><FONT SIZE=3D2>Bill,</FONT>
</P>

<P><FONT SIZE=3D2>Indeed, this approach seems good to me too. But =
expect people to argue the following points:</FONT>
<BR><FONT SIZE=3D2>1) Proxies as defined in RFC3261 cannot modify the =
From field. Sure, it is a small extension, but strictly speaking it is =
more than a pure proxy does.</FONT></P>

<P><FONT SIZE=3D2>2) This modification does break some services. It =
would be best to avoid it except when necessary e.g. if the *user* =
requires privacy, then still put Anonymous in the From =
field.</FONT></P>

<P><FONT SIZE=3D2>3) verification of the From field is either 'required =
for consistent service in a Service Provider environment' (myself, =
probably most of 3GPP, Juha, others ? ...) OR 'contrary to the =
definition of the From field as an end-to-end user asserted field' =
(Dean, Dave, others ? ...) - we have no consensus on this.</FONT></P>

<P><FONT SIZE=3D2>The only compromise I can see on this is:</FONT>
<BR><FONT SIZE=3D2>a) Those who want to verify the From: field =
do</FONT>
<BR><FONT SIZE=3D2>b) Those who don't want to don't</FONT>
<BR><FONT SIZE=3D2>c) In the absence of trust relationships the From: =
field is completely untrusted (like everything else!). If you have a =
trust relationship, then as part of this you will agree whether From: =
can be trusted or not.</FONT></P>

<P><FONT SIZE=3D2>d) The contents of the From field MAY be modified by =
devices in the network. So we should design new services and features =
with this assumption where possible</FONT></P>

<P><FONT SIZE=3D2>(d) is a logical consequence of (a) - don't try and =
argue out (d) unless you think you can exclude (a) too.</FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: William Marshall [<A =
HREF=3D"mailto:wtm@research.att.com">mailto:wtm@research.att.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>&gt; Sent: 17 April 2002 03:48</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sip] z9hG4bK forever ???</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Subject: was Re: Screened FromL and =
Privacy</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: was Re: Summary of RE: [Sip] Comment, =
SIP Privacy draft</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: was lots of others, too</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; All this discussion of proxy modification of =
the From and To headers,</FONT>
<BR><FONT SIZE=3D2>&gt; and the need for a B2BUA, seems to me to be a =
short-term problem only.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; RFC3261 (2543bis-09) specifies matching of =
dialog-ids based only</FONT>
<BR><FONT SIZE=3D2>&gt; on Call-ID and tags, not on display-name or on =
addr-spec.&nbsp; So</FONT>
<BR><FONT SIZE=3D2>&gt; when all endpoints are compliant with RFC3261, =
there will be no</FONT>
<BR><FONT SIZE=3D2>&gt; problem with a proxy modifying the display-name =
or addr-spec in</FONT>
<BR><FONT SIZE=3D2>&gt; the From header.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; It is only during a transition period, when =
endpoints that conform</FONT>
<BR><FONT SIZE=3D2>&gt; to RFC2543 are supported as well as newer =
ones.&nbsp; How long will</FONT>
<BR><FONT SIZE=3D2>&gt; this be?&nbsp; How long will a proxy need to =
handle a UAC that doesn't</FONT>
<BR><FONT SIZE=3D2>&gt; supply a Contact header?&nbsp; How long will a =
proxy need to handle</FONT>
<BR><FONT SIZE=3D2>&gt; a Record-Route header without the =
&quot;lr&quot; tag?&nbsp; How long will a</FONT>
<BR><FONT SIZE=3D2>&gt; UAS need to handle a BYE received prior to =
sending 2xx?&nbsp; How long </FONT>
<BR><FONT SIZE=3D2>&gt; will the comparison be coded into the proxies =
and UAs looking </FONT>
<BR><FONT SIZE=3D2>&gt; for &quot;z9hG4bK&quot;?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In all cases, my hope is not too long.&nbsp; =
And, at the point when</FONT>
<BR><FONT SIZE=3D2>&gt; these &quot;conversion aids&quot; no longer =
need to be present in SIP</FONT>
<BR><FONT SIZE=3D2>&gt; implementations, it will be safe for proxies to =
modify the From</FONT>
<BR><FONT SIZE=3D2>&gt; header to provide anonymity, if the subscriber, =
or the user, or the</FONT>
<BR><FONT SIZE=3D2>&gt; regulator, (or others?), says that it =
should.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In cases like 3GPP, where (IIRC) release 5 does =
not include</FONT>
<BR><FONT SIZE=3D2>&gt; IMS connections outside of the wireless =
operator's network, the</FONT>
<BR><FONT SIZE=3D2>&gt; problem is even simpler.&nbsp; Since 3GPP =
required RFC3261 everywhere,</FONT>
<BR><FONT SIZE=3D2>&gt; modification of the From header can be done =
immediately.&nbsp; I would</FONT>
<BR><FONT SIZE=3D2>&gt; hope this transition period and the need for =
&quot;conversion aids&quot; would</FONT>
<BR><FONT SIZE=3D2>&gt; be over before 3GPP release 6 is =
deployed.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; My conclusion from all of this is that 3GPP can =
require the UE to</FONT>
<BR><FONT SIZE=3D2>&gt; insert a valid user identifier in the From =
header, and the S-CSCF can</FONT>
<BR><FONT SIZE=3D2>&gt; change it if privacy is desired, on a =
session-by-session basis,</FONT>
<BR><FONT SIZE=3D2>&gt; based on whatever rules the S-CSCF wants to =
follow.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Comments?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Bill Marshall</FONT>
<BR><FONT SIZE=3D2>&gt; wtm@research.att.com</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
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From daemon@optimus.ietf.org  Wed Apr 17 07:49:42 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>, Mpierce1@aol.com
Cc: oran@cisco.com, sip@ietf.org
Subject: RE: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Wed, 17 Apr 2002 12:18:40 +0100
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Henning wrote:


> No reasonable provider should put that requirement on From, for the
> reasons Dean mentioned. This leaves the proxy-authentication data.
> Interesting question is whether the proxy-authentication data needs to
> be stripped after it has been "used", as it may well apply to several
> down-stream proxies.
> 

I believe the view amongst providers in 3GPP is that they would expect such
a requirement to be placed on the From field, if any information is to go in
there at all. They are _mostly_ reasonable in 3GPP :-)

If the proxy-authentication data could include the plain-text (user)name of
the user, and if the user or the subscriber has requested privacy, then the
information certainly could not be passed on.

I suppose you could pass it on to a 'trusted' entity, assuming that this
entity could reliably determine the intersection of the subscribers and
users privacy requirements so it knew what to do next.

Regards,

Mark

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<P><FONT SIZE=3D2>Henning wrote:</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; No reasonable provider should put that =
requirement on From, for the</FONT>
<BR><FONT SIZE=3D2>&gt; reasons Dean mentioned. This leaves the =
proxy-authentication data.</FONT>
<BR><FONT SIZE=3D2>&gt; Interesting question is whether the =
proxy-authentication data needs to</FONT>
<BR><FONT SIZE=3D2>&gt; be stripped after it has been &quot;used&quot;, =
as it may well apply to several</FONT>
<BR><FONT SIZE=3D2>&gt; down-stream proxies.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I believe the view amongst providers in 3GPP is that =
they would expect such a requirement to be placed on the From field, if =
any information is to go in there at all. They are _mostly_ reasonable =
in 3GPP :-)</FONT></P>

<P><FONT SIZE=3D2>If the proxy-authentication data could include the =
plain-text (user)name of the user, and if the user or the subscriber =
has requested privacy, then the information certainly could not be =
passed on.</FONT></P>

<P><FONT SIZE=3D2>I suppose you could pass it on to a 'trusted' entity, =
assuming that this entity could reliably determine the intersection of =
the subscribers and users privacy requirements so it knew what to do =
next.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

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From daemon@optimus.ietf.org  Wed Apr 17 08:41:07 2002
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Mark Watson pointed out:
> d) The contents of the From field MAY be modified by devices in the network.
> So we should design new services and features with this assumption where
> possible

I would go even a few steps farther.

We have now identified TWO cases where un-proxy-like behavior will be
done in real networks.  The first, a few months ago, was where a
firewall control proxy modified the Contact header, and possibly 
also modified the addresses and ports in the SDP.  The second is here, 
where a privacy proxy modifies the From header (and quite probably, others).

So my recommendations:
1) It is worthwhile recognizing these situations in draft-ietf-sip-rfc3261bis,
   and including text that limits the message transformations that such
   network elements can perform.  Essentially this means defining the
   previously coined term "transparent B2BUA".

2) New serivces MUST be designed with the assumption that such "transparent
   B2BUAs" will be in the path, minimize the ways they will fail with such
   message transformations, and discuss the recovery action that a UA may
   take in such failure cases. In the same way that new services MUST be
   designed knowing that forking may happen.  (draft-ietf-sip-guidelines?)

An example where (2) has already been followed is draft-ietf-sip-replaces;
it uses to-tag and from-tag, rather than full contents of From and To.

Bill Marshall
wtm@research.att.com

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From daemon@optimus.ietf.org  Wed Apr 17 10:30:46 2002
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: The SIP Negotiate Method
	Author(s)	: S. Parameswar, B. Stucker
	Filename	: draft-spbs-sip-negotiate-01.txt
	Pages		: 
	Date		: 16-Apr-02
	
There is a need to negotiate a multitude of parameters, settings, 
and algorithms when setting up sessions using Session Initiated 
Protocol (SIP). While SIP itself provides mechanisms for 
negotiation of these parameters on a per-session basis through the
use of the INVITE method, it does not provide a ready mechanism
for meta-session negotiation. The closest mechanism provided is 
the REGISTER method, however, this method is directed towards the 
registrar alone, and cannot be used to conduct negotiation of 
parameters between any two arbitrary SIP nodes.

A URL for this Internet-Draft is:
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From daemon@optimus.ietf.org  Wed Apr 17 11:15:43 2002
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Date: Wed, 17 Apr 2002 15:18:02 +0200
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"14.2 UAS Behaviour" 
mentions that a 500 (Server Internal Error) needs to include a
Retry-After header field ico. the 2nd invite is received before the
final response of the 1st INVITE is not sent yet.

"20.33 Retry-After" 
this header field can be used with 503, 404, 413, 480, 486, 600 or 603.

I assume that this list should include the 500 too?

[rfc2543bis-09]

Thanks,
Diether.


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From daemon@optimus.ietf.org  Wed Apr 17 12:36:34 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "William Marshall" <wtm@research.att.com>, <sip@ietf.org>
References: <200204171210.IAA33585@fish.research.att.com>
Subject: Re: [Sip] z9hG4bK forever ???
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Bill said:
> We have now identified TWO cases where un-proxy-like behavior will be
> done in real networks.  The first, a few months ago, was where a
> firewall control proxy modified the Contact header, and possibly
> also modified the addresses and ports in the SDP.  The second is here,
> where a privacy proxy modifies the From header (and quite probably,
others).

Proxies in any form cannot modify bodies, and B2BUAs are likely to break
things if they try to. SIP security relies greatly on smime-protected
bodies.

What you fail to distinguish is that like identity, there are two sorts
of privacy:

1) UA asserted
2) Network asserted

UA asserted privacy requires that the network NOT tamper with what the
UA has done, nor insert additional information that compromises the
privacy of the UA.

Network asserted privacy  requires a full, non-transparent B2BUA. I see
this as an OPTIONAL service, EXTERNAL to the proxy-like CSCF elements.
In other words, an anonymizer application.

--
Dean




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From daemon@optimus.ietf.org  Wed Apr 17 12:58:52 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "William Marshall" <wtm@research.att.com>, <sip@ietf.org>
References: <200204170247.WAA40877@fish.research.att.com>
Subject: Re: [Sip] z9hG4bK forever ???
Date: Wed, 17 Apr 2002 11:35:38 -0500
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Baill said:
> All this discussion of proxy modification of the From and To headers,
> and the need for a B2BUA, seems to me to be a short-term problem only.
>
> RFC3261 (2543bis-09) specifies matching of dialog-ids based only
> on Call-ID and tags, not on display-name or on addr-spec.  So
> when all endpoints are compliant with RFC3261, there will be no
> problem with a proxy modifying the display-name or addr-spec in
> the From header.

I disagree. I don't want the network mangling my user-to-user headers
unless I've specifically requested it to do so. Specifically, the
non-authenticated, non-mangled, non-protected From: field offers utility
which is DIFFERENT than that provided by network authenticated calling
party ID, and I don't want to sacrifice that utility just because this
is a capability the widely-deployed PSTN (as opposed to say Q.SIG)
didn't have . . .

--
Dean




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From daemon@ns.ietf.org  Wed Apr 17 16:20:12 2002
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FYI...

In addition to the PDF draft, javadoc API and a jar file should be 
available from the JCP site in a day or two.

Anders

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The Expert Group of the following Specification development effort

      JSR-000116 SIP Servlet API

has released a draft specification for Public Review.

    http://jcp.org/jsr/stage/public.jsp

The Public Review closes 9 May 2002.

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From daemon@ns.ietf.org  Wed Apr 17 16:34:38 2002
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Date: Wed, 17 Apr 2002 15:07:00 -0500
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] Cookies or State
In-reply-to: 
 <313680C9A886D511A06000204840E1CF57CE71@whq-msgusr-02.pit.comms.marconi.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, sip@ietf.org
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>The question is, should we 
> base it on State or Cookies?

I like the cookies better, since they mirror a successful web approach.

>To our 
> consternation, the same IP claims were then made against Cookies!
Consternation is the right word.

Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Rosen, Brian
> Sent: Wednesday, April 17, 2002 2:28 PM
> To: 'sip@ietf.org'
> Subject: [Sip] Cookies or State
> 
> 
> There are two drafts that define how to save state in a UA 
> for subsequent use by a Proxy Server - State and Cookies.
> 
> State is part of the DCS work.  Cookies arose when IP assertions were 
> made against State.  Cookies attempted to get around the IP 
> issues by defining it as exactly like HTTP cookies.  To our 
> consternation, the same IP claims were then made against Cookies!
> 
> We need to get this capability.  The question is, should we 
> base it on State or Cookies?
> 
> In my personal (biased) opinion we are better off with 
> Cookies. If anyone did want to fight the IP claim, you would 
> clearly be better off with 75% same language as HTTP cookies.
> 
> Brian
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


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From daemon@ns.ietf.org  Wed Apr 17 16:49:33 2002
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>, <sip@ietf.org>
Subject: RE: [Sip] Cookies or State
Date: Wed, 17 Apr 2002 15:26:49 -0500
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Cookie!! (says the blue furry guy...)

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Rosen,
> Brian
> Sent: Wednesday, April 17, 2002 2:28 PM
> To: 'sip@ietf.org'
> Subject: [Sip] Cookies or State
> 
> 
> There are two drafts that define how to save state in a UA for subsequent
> use by a Proxy Server - State and Cookies.
> 
> State is part of the DCS work.  Cookies arose when IP assertions were 
> made against State.  Cookies attempted to get around the IP issues
> by defining it as exactly like HTTP cookies.  To our consternation,
> the same IP claims were then made against Cookies!
> 
> We need to get this capability.  The question is, should we
> base it on State or Cookies?
> 
> In my personal (biased) opinion we are better off with Cookies.
> If anyone did want to fight the IP claim, you would clearly
> be better off with 75% same language as HTTP cookies.
> 
> Brian
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Apr 17 17:02:00 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Wed, 17 Apr 2002 15:27:37 -0400
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Subject: [Sip] Cookies or State
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There are two drafts that define how to save state in a UA for subsequent
use by a Proxy Server - State and Cookies.

State is part of the DCS work.  Cookies arose when IP assertions were 
made against State.  Cookies attempted to get around the IP issues
by defining it as exactly like HTTP cookies.  To our consternation,
the same IP claims were then made against Cookies!

We need to get this capability.  The question is, should we
base it on State or Cookies?

In my personal (biased) opinion we are better off with Cookies.
If anyone did want to fight the IP claim, you would clearly
be better off with 75% same language as HTTP cookies.

Brian

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Apr 17 17:30:06 2002
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Subject: RE: [Sip] Cookies or State
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> In my personal (biased) opinion we are better off with Cookies.
> If anyone did want to fight the IP claim, you would clearly
> be better off with 75% same language as HTTP cookies.

There are some good reasons to stick with an HTTP cookies approach, although I don't have a strong preference as long as we pick one. IANAL, but I'm sure the IPR issue won't go away by copying HTTP cookie language. On the other hand, appropriate substitution of normative references might deflect the claim. In any case, kludge-free SIP state mechanisms are long overdue.

--Andy Zmolek 
    Technology & Standards Engineer 
      CTO Standards 
        Avaya Inc. 

            zmolek@avaya.com 
              +1 720 444 4001 








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From daemon@ns.ietf.org  Wed Apr 17 19:44:23 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
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Subject: [Sip] new drafts relating to privacy and identity concepts posted
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Please review:

http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-privacy-longter
m-00.txt

http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-identity-00.txt

They've been sent into the internet-drafts repository and should appear
there shortly.

--
Dean


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Subject: Re: [Sip] Cookies or State
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In a message dated 4/17/02 3:32:15 PM Eastern Daylight Time, 
Brian.Rosen@marconi.com writes:


> There are two drafts that define how to save state in a UA for subsequent
> use by a Proxy Server - State and Cookies.
> 
> State is part of the DCS work.  Cookies arose when IP assertions were 
> made against State.  Cookies attempted to get around the IP issues
> by defining it as exactly like HTTP cookies.  To our consternation,
> the same IP claims were then made against Cookies!
> 
> We need to get this capability.  The question is, should we
> base it on State or Cookies?
> 
> In my personal (biased) opinion we are better off with Cookies.
> If anyone did want to fight the IP claim, you would clearly
> be better off with 75% same language as HTTP cookies.
> 

I presume the reference for "Cookies" is still draft-willis-sip-cookies-00 of 
July 2001. I haven't seen any further discussion on this draft or responses 
to the comments I submitted Nov 19. From that draft, it appears there are 
significant security concerns with the use of "Cookies" in a fashion similar 
to HTTP as proposed in this draft.

State would appear to have fewer security concerns if it is simply a value 
stored in the UA for the beniefit of the proxy to fetch and use later. It is 
presumed that the lack of a standard definition of this state value would 
preclude its use by a different proxy than the one which stored it (at least 
by a proxy of a different design).

It continues to escape me why it would not be better for the proxy to simply 
stored this "state" in its own storage, rather than using every UA as a 
distributed storage device. This concept seems to go back to the days when 
memory was limited (the first telephone switch I was involved in designing 
had 64k bytes of memory!) When one can buy 512M bytes of memory for less than 
$100, this should no longer be an issue. The "state" draft seems to indicate 
that the only reason for doing this is so that "proxy servers can remain 
stateless for the duration of the call", but I have never heard a reason why 
this is desirable. (Note that bis-09 defines the term "call-stateful proxy".) 
Why don't we just admit that it is required in some cases to keep the call 
state in the proxy and stop trying to invent ways around this, all of which 
have security and performance problems?

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/17/02 3:32:15 PM Eastern Daylight Time, Brian.Rosen@marconi.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">There are two drafts that define how to save state in a UA for subsequent
<BR>use by a Proxy Server - State and Cookies.
<BR>
<BR>State is part of the DCS work. &nbsp;Cookies arose when IP assertions were 
<BR>made against State. &nbsp;Cookies attempted to get around the IP issues
<BR>by defining it as exactly like HTTP cookies. &nbsp;To our consternation,
<BR>the same IP claims were then made against Cookies!
<BR>
<BR>We need to get this capability. &nbsp;The question is, should we
<BR>base it on State or Cookies?
<BR>
<BR>In my personal (biased) opinion we are better off with Cookies.
<BR>If anyone did want to fight the IP claim, you would clearly
<BR>be better off with 75% same language as HTTP cookies.
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I presume the reference for "Cookies" is still draft-willis-sip-cookies-00 of July 2001. I haven't seen any further discussion on this draft or responses to the comments I submitted Nov 19. From that draft, it appears there are significant security concerns with the use of "Cookies" in a fashion similar to HTTP as proposed in this draft.
<BR>
<BR>State would appear to have fewer security concerns if it is simply a value stored in the UA for the beniefit of the proxy to fetch and use later. It is presumed that the lack of a standard definition of this state value would preclude its use by a different proxy than the one which stored it (at least by a proxy of a different design).
<BR>
<BR>It continues to escape me why it would not be better for the proxy to simply stored this "state" in its own storage, rather than using every UA as a distributed storage device. This concept seems to go back to the days when memory was limited (the first telephone switch I was involved in designing had 64k bytes of memory!) When one can buy 512M bytes of memory for less than $100, this should no longer be an issue. The "state" draft seems to indicate that the only reason for doing this is so that "proxy servers can remain stateless for the duration of the call", but I have never heard a reason why this is desirable. (Note that bis-09 defines the term "call-stateful proxy".) Why don't we just admit that it is required in some cases to keep the call state in the proxy and stop trying to invent ways around this, all of which have security and performance problems?
<BR>
<BR>Mike
<BR></FONT></HTML>

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In a message dated 4/16/02 2:39:47 PM Eastern Daylight Time, 
hgs@cs.columbia.edu writes:


> > [Oran] I think the information that is of relevance to this discussion is 
> stuff 
> >   that was *NOT* inserted by the user, but was instead inserted by the 
> >   service provider based on things the user had previously disclosed to 
> >   the service provider. These are the things which in fact do need some 
> >   level of protection, which is why we have this complicated 
> applicability 
> >   statement on the privacy draft and why (I at least) would argue that 
> the 
> >   only reasonable design is one in which any identifying information 
> >   inserted by the service provider be encrypted or otherwise protected 
> >   from disclosure to other parties (including the callee) unless the 
> >   customer explicitly requests it not to be. 
> > 
> > 
> > 
> > [Mike Pierce] I think you have to add to the first sentence above "and 
> information inserted by the user that is required by the
> > Service Provider as a prerequisite for service". If the service provider 
> requires that the user enter a valid ID, then it
> > must be protected the same as an ID that the Service Provider inserted. 
> 
> [HGS] No reasonable provider should put that requirement on From, for the
> reasons Dean mentioned. This leaves the proxy-authentication data.
> Interesting question is whether the proxy-authentication data needs to
> be stripped after it has been "used", as it may well apply to several
> down-stream proxies.
> 

I think this shows how far we are from any agreement on anything. In his 
e-mail of Apr 15, Dean "mostly" agreed with my statement (although crediting 
it to Mark).

While my statement was "If the service provider requires..." and I believe 
there have been a number of comments that certain service providers must 
require this to support mandated tracing, etc., your comment argues that "No 
reasonable provider should put that requirement on From..."

The issue again is what type of "service provider" is being talked about. It 
is clear that some may require this (a valid ID in From) and some may not. So 
my statement is true. If it's not the "From" field that must be provided, it 
is whatever field the discussion ends up with that provides the functionality 
that many presumed was the purpose of "From".

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/16/02 2:39:47 PM Eastern Daylight Time, hgs@cs.columbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt; [Oran] I think the information that is of relevance to this discussion is stuff 
<BR>&gt; &nbsp;&nbsp;that was *NOT* inserted by the user, but was instead inserted by the 
<BR>&gt; &nbsp;&nbsp;service provider based on things the user had previously disclosed to 
<BR>&gt; &nbsp;&nbsp;the service provider. These are the things which in fact do need some 
<BR>&gt; &nbsp;&nbsp;level of protection, which is why we have this complicated applicability 
<BR>&gt; &nbsp;&nbsp;statement on the privacy draft and why (I at least) would argue that the 
<BR>&gt; &nbsp;&nbsp;only reasonable design is one in which any identifying information 
<BR>&gt; &nbsp;&nbsp;inserted by the service provider be encrypted or otherwise protected 
<BR>&gt; &nbsp;&nbsp;from disclosure to other parties (including the callee) unless the 
<BR>&gt; &nbsp;&nbsp;customer explicitly requests it not to be. 
<BR>&gt; 
<BR>&gt; 
<BR>&gt; 
<BR>&gt; [Mike Pierce] I think you have to add to the first sentence above "and information inserted by the user that is required by the
<BR>&gt; Service Provider as a prerequisite for service". If the service provider requires that the user enter a valid ID, then it
<BR>&gt; must be protected the same as an ID that the Service Provider inserted. 
<BR>
<BR>[HGS] No reasonable provider should put that requirement on From, for the
<BR>reasons Dean mentioned. This leaves the proxy-authentication data.
<BR>Interesting question is whether the proxy-authentication data needs to
<BR>be stripped after it has been "used", as it may well apply to several
<BR>down-stream proxies.
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>I think this shows how far we are from any agreement on anything. In his e-mail of Apr 15, Dean "mostly" agreed with my statement (although crediting it to Mark).
<BR>
<BR>While my statement was "If the service provider requires..." and I believe there have been a number of comments that certain service providers must require this to support mandated tracing, etc., your comment argues that "No reasonable provider should put that requirement on From..."
<BR>
<BR>The issue again is what type of "service provider" is being talked about. It is clear that some may require this (a valid ID in From) and some may not. So my statement is true. If it's not the "From" field that must be provided, it is whatever field the discussion ends up with that provides the functionality that many presumed was the purpose of "From".
<BR>
<BR>Mike
<BR></FONT></HTML>

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Sorry for the cross-post.

I've submitted some new Internet-Drafts on SIP privacy and identity. I hope
that these drafts can serve as a good starting point for discussion about
long-term cryptographic approaches to privacy and identity that will be
generally serviceable for the SIP protocol. As Dean's note to the SIP WG
list suggested, they are now available on softarmor.

Any comments on the drafts would be appreciated, and I hope we'll have a
chance to discuss them at the interim meeting.

The abstracts of the two drafts follow:

Privacy: This document defines new mechanisms for the Session Initiation
Protocol (SIP) in support of privacy. Specifically, guidelines are provided
for the creation of messages that do not divulge personal identity
information. A new "privacy service" logical role for intermediaries is
defined to answer some privacy requirements that user agents cannot satisfy
themselves. Finally, means are presented by which a user can request
particular functions from a privacy service. 

Identity: The existing mechanisms for expressing identity in the Session
Initiation Protocol do not currently permit an administrative domain to
securely verify the identity of the originator of a request. This document
recommends practices and conventions for authenticating end users, and
proposes a way to distribute secure authenticated identities within SIP
messages. 

Jon Peterson
NeuStar, Inc.

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From daemon@ns.ietf.org  Wed Apr 17 20:13:19 2002
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Message-ID: <041601c1e66a$8391b210$133fed0c@C1893415A>
From: "Dean Willis" <dean.willis@softarmor.com>
To: <Mpierce1@aol.com>
Cc: <sip@ietf.org>
References: <94.24e62d2e.29ef5d6f@aol.com>
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
Date: Wed, 17 Apr 2002 18:49:28 -0500
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M said:
---------
I think this shows how far we are from any agreement on anything. In his =
e-mail of Apr 15, Dean "mostly" agreed with my statement (although =
crediting it to Mark).=20
--------

whoops, sorry -- I guess I'll stick to initials. Detangling mhtml =
postings is hard.


M said also:
--------
While my statement was "If the service provider requires..." and I =
believe there have been a number of comments that certain service =
providers must require this to support mandated tracing, etc., your =
comment argues that "No reasonable provider should put that requirement =
on From..."=20

The issue again is what type of "service provider" is being talked =
about. It is clear that some may require this (a valid ID in From) and =
some may not. So my statement is true. If it's not the "From" field that =
must be provided, it is whatever field the discussion ends up with that =
provides the functionality that many presumed was the purpose of "From". =

---------

And speaking of discussion, it looks like Jon Peterson just published a =
new draft on this very topic.

--
Dean


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
FAMILY=3D"SANSSERIF"></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>M said:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>---------</FONT></DIV>
<DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
FAMILY=3D"SANSSERIF">I think this shows how far we are from any =
agreement on=20
anything. In his e-mail of Apr 15, Dean "mostly" agreed with my =
statement=20
(although crediting it to Mark). <BR>--------</FONT></FONT></DIV>
<DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
FAMILY=3D"SANSSERIF"></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
FAMILY=3D"SANSSERIF">whoops, sorry -- I guess I'll stick to initials. =
Detangling=20
mhtml postings is hard.</FONT></FONT></DIV>
<DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
FAMILY=3D"SANSSERIF">&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>M said also:</DIV>
<DIV>--------<BR>While my statement was "If the service provider =
requires..."=20
and I believe there have been a number of comments that certain service=20
providers must require this to support mandated tracing, etc., your =
comment=20
argues that "No reasonable provider should put that requirement on =
From..."=20
<BR><BR>The issue again is what type of "service provider" is being =
talked=20
about. It is clear that some may require this (a valid ID in From) and =
some may=20
not. So my statement is true. If it's not the "From" field that must be=20
provided, it is whatever field the discussion ends up with that provides =
the=20
functionality that many presumed was the purpose of "From". =
<BR>---------</DIV>
<DIV>&nbsp;</DIV>
<DIV>And speaking of discussion, it looks like Jon Peterson just =
published a new=20
draft on this very topic.</DIV>
<DIV>&nbsp;</DIV>
<DIV>--</DIV>
<DIV>Dean<BR></DIV></FONT></FONT></BODY></HTML>

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From daemon@ns.ietf.org  Wed Apr 17 20:20:43 2002
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To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
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	 <sip@ietf.org>
Subject: RE: [Sip] Cookies or State
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My vote is for cookie as well - it is more generic and might be used for
other applications other than those in the DCS state draft.

Andrew 

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Wednesday, April 17, 2002 2:28 PM
> To: 'sip@ietf.org'
> Subject: [Sip] Cookies or State
> 
> 
> There are two drafts that define how to save state in a UA 
> for subsequent
> use by a Proxy Server - State and Cookies.
> 
> State is part of the DCS work.  Cookies arose when IP assertions were 
> made against State.  Cookies attempted to get around the IP issues
> by defining it as exactly like HTTP cookies.  To our consternation,
> the same IP claims were then made against Cookies!
> 
> We need to get this capability.  The question is, should we
> base it on State or Cookies?
> 
> In my personal (biased) opinion we are better off with Cookies.
> If anyone did want to fight the IP claim, you would clearly
> be better off with 75% same language as HTTP cookies.
> 
> Brian
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@ns.ietf.org  Wed Apr 17 20:29:10 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <Mpierce1@aol.com>, <Brian.Rosen@marconi.com>, <sip@ietf.org>
References: <f5.1a7959cd.29ef5d68@aol.com>
Subject: Re: [Sip] Cookies or State
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  ----- Original Message -----=20
  From: Mpierce1@aol.com=20
  I presume the reference for "Cookies" is still =
draft-willis-sip-cookies-00 of July 2001. I haven't seen any further =
discussion on this draft or responses to the comments I submitted Nov =
19. From that draft, it appears there are significant security concerns =
with the use of "Cookies" in a fashion similar to HTTP as proposed in =
this draft.=20

  State would appear to have fewer security concerns if it is simply a =
value stored in the UA for the beniefit of the proxy to fetch and use =
later. It is presumed that the lack of a standard definition of this =
state value would preclude its use by a different proxy than the one =
which stored it (at least by a proxy of a different design).=20

  It continues to escape me why it would not be better for the proxy to =
simply stored this "state" in its own storage, rather than using every =
UA as a distributed storage device. This concept seems to go back to the =
days when memory was limited (the first telephone switch I was involved =
in designing had 64k bytes of memory!) When one can buy 512M bytes of =
memory for less than $100, this should no longer be an issue. The =
"state" draft seems to indicate that the only reason for doing this is =
so that "proxy servers can remain stateless for the duration of the =
call", but I have never heard a reason why this is desirable. (Note that =
bis-09 defines the term "call-stateful proxy".) Why don't we just admit =
that it is required in some cases to keep the call state in the proxy =
and stop trying to invent ways around this, all of which have security =
and performance problems?=20
That is the correct reference, although Brian actually edited the draft =
based on my rough notes . . .

I believe the security concerns are roughly equal -- in both cases, the =
UA is given a hunk of stuff that it has to regurgigate on future =
messages in this dialog.

Both approaches leave the format of the "hunk" open -- it could be =
cryptographically protected, just a hash value matching a table stored =
on the proxy, or almost anything else.

I also believe the security concern is SLIGHTLY less urgent than in =
HTTP. As I understand it, both state and cookie drafts indicate a =
substantially shorter lifetime for the token. In the case of state, the =
lifetime is the duration of the current dialog. In the case of cookie, =
it's the current dialog plus any dialog resulting from redirection or =
referral of the original dialog.=20

HTTP cookies, on the other hand, persist relatively indefiitely and may =
be extracted by any server in the domain of the cookie during any HTTP =
interaction with said server for the duraction of the cookie.

Now it is certainly true that one COULD put sensitive information in =
plain text into a state or cookie token such that other nodes might gain =
access to that information. The cookies draft does mention that this is =
not a good idea.

--
dean


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<STYLE></STYLE>

<META content=3D"MSHTML 6.00.2715.400" name=3DGENERATOR></HEAD>
<BODY>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3DMpierce1@aol.com =
href=3D"mailto:Mpierce1@aol.com">Mpierce1@aol.com</A>=20
  </DIV>
  <DIV><FONT face=3Darial,helvetica><FONT lang=3D0 face=3DArial =
color=3D#000000 size=3D2=20
  FAMILY=3D"SANSSERIF">I presume the reference for "Cookies" is still=20
  draft-willis-sip-cookies-00 of July 2001. I haven't seen any further=20
  discussion on this draft or responses to the comments I submitted Nov =
19. From=20
  that draft, it appears there are significant security concerns with =
the use of=20
  "Cookies" in a fashion similar to HTTP as proposed in this draft.=20
  <BR><BR>State would appear to have fewer security concerns if it is =
simply a=20
  value stored in the UA for the beniefit of the proxy to fetch and use =
later.=20
  It is presumed that the lack of a standard definition of this state =
value=20
  would preclude its use by a different proxy than the one which stored =
it (at=20
  least by a proxy of a different design). <BR><BR>It continues to =
escape me why=20
  it would not be better for the proxy to simply stored this "state" in =
its own=20
  storage, rather than using every UA as a distributed storage device. =
This=20
  concept seems to go back to the days when memory was limited (the =
first=20
  telephone switch I was involved in designing had 64k bytes of memory!) =
When=20
  one can buy 512M bytes of memory for less than $100, this should no =
longer be=20
  an issue. The "state" draft seems to indicate that the only reason for =
doing=20
  this is so that "proxy servers can remain stateless for the duration =
of the=20
  call", but I have never heard a reason why this is desirable. (Note =
that=20
  bis-09 defines the term "call-stateful proxy".) Why don't we just =
admit that=20
  it is required in some cases to keep the call state in the proxy and =
stop=20
  trying to invent ways around this, all of which have security and =
performance=20
  problems? </DIV></BLOCKQUOTE>
<P>That is the correct reference, although Brian actually&nbsp;edited =
the draft=20
based on my rough notes . . .</P>
<P>I believe the security concerns are roughly equal -- in both cases, =
the UA is=20
given a hunk of stuff that it has to regurgigate on future messages in =
this=20
dialog.</P>
<P>Both approaches leave the format of the "hunk" open -- it could be=20
cryptographically protected, just a hash value matching a table stored =
on the=20
proxy, or almost anything else.</P>
<P>I also believe the security concern is SLIGHTLY less urgent than in =
HTTP. As=20
I understand it, both state and cookie drafts indicate a substantially =
shorter=20
lifetime for the token. In the case of state, the lifetime is the =
duration of=20
the&nbsp;current dialog. In the case of cookie, it's the current dialog =
plus any=20
dialog resulting from redirection or referral of the original dialog. =
</P>
<P>HTTP cookies, on the other hand, persist relatively indefiitely and =
may be=20
extracted by any server in the domain of the cookie during any HTTP =
interaction=20
with said server for the duraction of the cookie.</P>
<P>Now it is certainly true that one COULD put sensitive information in =
plain=20
text into a state or cookie token such that other nodes might gain =
access to=20
that information. The cookies draft does mention that this is not a good =

idea.</P>
<P>--<BR>dean</P></FONT></FONT></BODY></HTML>

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From daemon@ns.ietf.org  Wed Apr 17 20:40:43 2002
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To: "Chiou, Mark" <MChiou@Santera.com>
CC: Dean Willis <dean.willis@softarmor.com>, Mpierce1@aol.com, sip@ietf.org
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
References: <CD110021698980419241042CF576B8F2012BAE85@EXCALIBUR.santera.com>
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These kinds of questions regarding subject or message body fall into the
confidentiality category at that point...which is the next step in
providing privacy. The thread to this date has been focusing on identity
privacy as seen in the header fields, not the content of the SIP
message.  

I recommend, IMHO, that these items should be addressed separately, and
that the initial focus on SIP header privacy (except for the content and
related message body) is a good necessary first step to address. The
next obvious step should be the idea of confidentiality of the
content/message body as well as to the media itself (which I believe is
yet another separate issue from teh SIP protocol itself), which I
believe will be subject to required regulation when the smoke clears in
any event. Confidentiality will of course boil down to the SIPS
solution, IPSec possibly, or some other means, but is a separate topic
related to the caller/identification privacy.

When speaking in terms of privacy provided by a service provider, I
believe the act of requesting a UA account to be private to be a
customer subscription process separate and not provided as a SIP
signaling process. Unless my assumption is incorrect, in the event that
it is assumed that SIP will be used as the vehicle to provide the
subscription process and that this is the reason why the need for new
headers is being pursued within the WG. In that case I understand why
this is being pursued but that isnt something that I have seen clearly
in the thread.

I thought that Mark wanted clear direction for the draft to go, select
one of the three choices. The semantics could be hashed out during the
progress of the direction chosen.

"Chiou, Mark" wrote:
> 
> Coercion <=> consent.
> 
> There is no explicit consent within the signaling of an invidiual
> transaction. External contractual issues are outside of the protocol,
> and tend to violate the "openness" characteristic of an Internet
> protocol. In otherwises, there's a philosophical difference between an
> explicit request from the user that a network providing break privacy
> for that transaction, and a network that breaks privacy just because I
> didn't tell it NOT to.
> 
> Note of course, that we're not talking about PSTN provoders, we're
> talking IP to IP usage over the publicaly addressable Internet.
> 
> <Mark Chiou> So, for IP, all the UAs are defaulted to PRIVATE. When a user wishes to make an anonymous calls, SIP needs an indication for the network (i.e. Service Provider) to identify who is calling and translate or retrieve the identity of the calling from the service agreement, then pass the calling identity within the networks without sending the calling identity to the UA of the called party.
> 
> <snip>
> 
> Personally, the UA I uset he most writes every last SIP header and the
> body to the screen, and copies them all to a log file just in case. One
> cannot rely on the called party's software to protect the privacy of the
> calling party.
> 
> <Mark Chiou> Agree. So we have to separate the needs for Privacy and the needs for Network, such that the privacy is fully controlled by the UA. On the other hand, the Service Providers have the responsibility to obtain the UA's identify for network usage (i.e. Emergency, Law Enforcement, Call trace, and etc) without breaking the UA's privacy. I means the UA's identity, which is needed by the Network, shall not be delivered to the terminating of the UA.
> 
> Mark Chiou
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From daemon@ns.ietf.org  Wed Apr 17 21:04:45 2002
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From: Chris Martin <cmartin@dasecurenetworks.com>
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Christian Huitema wrote:
> 
> I have heard many contributors assert that to get real anonymity, one
> will have to use a relay, e.g. an "anonymizing B2BUA". I would like to
> point out a failure in this line of reasoning: anonymizing relays have
> been tried before, for e-mail, and the experience was absolutely not
> conclusive. In fact, two different kind of anonymizers have been tried:
> free ones, that kept absolutely no record, and subscriber-based, which
> kept some records. Both have failed, for different reason.
> 
> The failure mode of a completely free relay is basically spam, and
> similar sort of abuse. There is documentation available on the fate for
> example of the cypherpunk experiment: this set of relays was set up to
> provide true anonymity on the Internet, using a sophisticated mix of
> random redirection and cryptography. In theory, the design was
> excellent. In practice, the system was quickly overwhelmed by porn
> traffic, and the relays dropped one by one out of the system because
> they could not bear the resulting congestion.

> 
> Subscriber based relays are relays that keep some state: they actually
> identify the sender, before washing out the identity in the outgoing
> message. They are more robust to congestion, since they can get some
> funding in the process, but they have a different failure mode: lawsuit.
> Since they keep some records, the records can be subpoenaed, and the
> provider can be forced to disclose the identity of the sending party. In
> fact, in many codes, if the providers refuse to do so, they may be
> deemed an accomplice of whatever action was carried out in the anonymous
> exchange. Many providers have chosen to close their doors rather than
> have to defend more lawsuit -- often after a brush with the Church of
> Scientology.

Privacy, I think, in relation to a service provider is not the same as
anonimity to an Internet based user. Privacy is used to mask the
original user to the called party, and does not prevent law enforcement
from getting the originator of such calls if a law is broken. B2BUA's
implemented by an SP need not contain any user account information to
provide the level of privacy afforded by the B2BUA (topology hiding) as
opposed to the function of privacy which may be implemented by an SP
which may alias the users true identity to <PRIVATE> before sending the
packet on to the B2BUA which can then scrub the IP address. You can
still track the communication to the SP, but then you must use the
proper vehicle to request the proprietary information located in the SP
databases to identify the caller.

How do SP's of PSTN networks cope with lawsuits today when law
enforcement must get this information? I am seriously asking this
question not being sarcastic, anyone know the answer? It seems to me
that the questions have been answered by the the SPs' of the dinosaurs
that we are replacing, we must take some of the lessons learned as a
model for answering these questions dont you think?

> 
> If we really want practical anonymity, it has to be based on the UA, and
> it should not rely on the help of a third party. We should use a
> combination of encryption, pre-paid subscriptions, and anonymous IP
> addresses (e.g. IPv6 privacy addresses.)

I can see this for publicly non-service provider hosted UA's as a
possibility, but for those enterprises, SOHO's serviced by SP's
regulations will be developed that require SP's of these types of
communications to have more stringent rules and processes. The
telecommunications industry today must comply with such regualtions for
call traces, wiretaps etc. Every country has some set of regulations
that place these requirements on SP's of our particular flavor of
communications, which though different from the PSTN world will no doubt
adopt many of those same restrictions and regulations for many similar
aspects. I am not versed in the regulations or telecommunications but I
do know that they exist. Many parties in this list have brought many
such examples to the list, which were of great interest to me. 

I think that implementing it on the UA will be a great idea, and
probably will be implemented by many vendors just as NAT and firewall
traversal solutions are today, but in the SP realm many of these
features will not be required to be active and in many cases will be
restricted by policies other than the SP due to the fact that the SP
will most likely provide a centralized functin for this service in order
to meet regulation as well as to meet security policies implemented by
customers of the SP which would look upon such mechanisms as a bypass of
such policies, not to mention not very scalable in larger environments.

Just my 2 cents
Chris
> 
> -- Christian Huitema
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From daemon@ns.ietf.org  Thu Apr 18 01:59:16 2002
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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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Mpierce1@aol.com writes:

 > The issue again is what type of "service provider" is being talked
 > about. It is clear that some may require this (a valid ID in From)
 > and some may not. 

to me this is the same thing as verifying the customer's source ip
address.  we as a service provider DO check that the source ip address
of packets that we receive from our subscribers really belong to them.
if all service providers would do such checking, dod attacks would be
must more easier to trace down.

-- juha


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From daemon@optimus.ietf.org  Thu Apr 18 02:35:55 2002
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: William Marshall <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] z9hG4bK forever ???
Date: Thu, 18 Apr 2002 07:13:29 +0100
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Bill's argument breaks down in a few important points.

1)	Such a proxy" would not be RFC 3261 compliant.

2)	Release 5 does not preclude interworking with other SIP networks. It
merely does not specify how it occurs. It does not specify how interworking
with R2 might be performed, but I am sure that some wireless network at some
point in the future will specify it. The fact that Release 6 might get round
to a full specification will not invalidate any proprietary interworking
methods. Given the importance that some operators are placing on
interworking with enterprise networks, and the fact that most of the
enterprise SIP terminals out there are closer to RFC 2543 on this one than
to RFC 3261, I do not see it happening. Enterprise terminals also tend to
sit around longer than anyone thinks, and even if they are not attached to
the network, people still expect to pull them out of the cupboard after a
couple of years, plug them into the network, and see them work.

3)	In the longer term, with split mobiles, we cannot guarantee that the
UA part of the split is RFC 3261 conformant. Similar comments apply to the
lifetime of RFC2543 terminals as above.

"Short term" in Bill's argument can be 5 years or more.

Keith



> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: 17 April 2002 03:48
> To: sip@ietf.org
> Subject: [Sip] z9hG4bK forever ???
> 
> 
> Subject: was Re: Screened FromL and Privacy
> Subject: was Re: Summary of RE: [Sip] Comment, SIP Privacy draft
> Subject: was lots of others, too
> 
> All this discussion of proxy modification of the From and To headers,
> and the need for a B2BUA, seems to me to be a short-term problem only.
> 
> RFC3261 (2543bis-09) specifies matching of dialog-ids based only
> on Call-ID and tags, not on display-name or on addr-spec.  So
> when all endpoints are compliant with RFC3261, there will be no
> problem with a proxy modifying the display-name or addr-spec in
> the From header.
> 
> It is only during a transition period, when endpoints that conform
> to RFC2543 are supported as well as newer ones.  How long will
> this be?  How long will a proxy need to handle a UAC that doesn't
> supply a Contact header?  How long will a proxy need to handle
> a Record-Route header without the "lr" tag?  How long will a
> UAS need to handle a BYE received prior to sending 2xx?  How long 
> will the comparison be coded into the proxies and UAs looking 
> for "z9hG4bK"?
> 
> In all cases, my hope is not too long.  And, at the point when
> these "conversion aids" no longer need to be present in SIP
> implementations, it will be safe for proxies to modify the From
> header to provide anonymity, if the subscriber, or the user, or the
> regulator, (or others?), says that it should.
> 
> In cases like 3GPP, where (IIRC) release 5 does not include
> IMS connections outside of the wireless operator's network, the
> problem is even simpler.  Since 3GPP required RFC3261 everywhere,
> modification of the From header can be done immediately.  I would
> hope this transition period and the need for "conversion aids" would
> be over before 3GPP release 6 is deployed.
> 
> My conclusion from all of this is that 3GPP can require the UE to
> insert a valid user identifier in the From header, and the S-CSCF can
> change it if privacy is desired, on a session-by-session basis,
> based on whatever rules the S-CSCF wants to follow.
> 
> Comments?
> 
> Bill Marshall
> wtm@research.att.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Thu Apr 18 02:40:54 2002
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Subject: RE: [Sip] Correction:SIP Message Draft Fast WGLC
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Hi!

I have a question related to the message draft:

What should be used, if only a URL to the content (e.g. http),
but not the content itself is delivered to the receiving UA, so
that it is the receiving users decision, whether to download the
content of not.

This makes sense, if the _bandwidth_is_limited_
and _large_messages_ are sent. This could be even a service
(in some network element, e.g. store and forward server)
provided to the user, to avoid that the channel down to the
user agent gets congested for too long time.

Is e.g. the Call-Info header field a good idea?
Or should the payload be used for this?
Better proposals?

cheers,
 Bernie

> -----Original Message-----
> From: ext Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 12 April, 2002 22:17
> To: sip@ietf.org
> Cc: brian.rosen@maroni.com; rohan@cisco.com; jo@ipdialog.com;
> bcampbell@dynamicsoft.com
> Subject: [Sip] Correction:SIP Message Draft Fast WGLC
> 
> 
> 
> Oops.
> 
> Please review :
> 
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-message-02.txt
> 
> Until the -02 makes it into the draft repository. They're not 
> as fast as
> I'm wishing for (ok, light speed is too slow, while we're at it!)
> 
> Thanks,
> 
> --
> Dean
> 
> 
> Original message:
> -------------
> We need to quickly review the SIM Message Method Extension. This draft
> has been widely discussed, iterated several times in SIMPLE, and is on
> its 2nd revision in SIP. The draft does not appear to be 
> controversial.
> 
> See:
> 
> http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt
> 
> We expect to move to IETF last call next week, so please raise any
> issues immediately.
> 
> Ben Campbell will coordinate for the draft editors.
> 
> -----------------
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'William Marshall'" <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] z9hG4bK forever ???
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Mark Watson writes:

 > The only compromise I can see on this is:
 > a) Those who want to verify the From: field do
 > b) Those who don't want to don't
 > c) In the absence of trust relationships the From: field is completely
 > untrusted (like everything else!). If you have a trust relationship, then as
 > part of this you will agree whether From: can be trusted or not.
 > d) The contents of the From field MAY be modified by devices in the network.
 > So we should design new services and features with this assumption where
 > possible

the above is consistent with what i have tried to say about this topic
and contains nothing new.  this whole thread has thus been more or less
useless.

-- juha


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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        William Marshall
	 <wtm@research.att.com>, sip@ietf.org
Subject: RE: [Sip] z9hG4bK forever ???
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I think we're about at the end of this one.

Two final points from me:
1) Dean's desire to avoid having the From: field mangled unless he has
specifically requested it is reasonable. I still think  that the
subscriber's wishes could overrule this though.

This is no different from the ISDN 'Special Arrangement' where the user
inserts any number they like in the Calling Party Number field. The network
passes it though with no checking, changing, nothing - I think this is
analogous to what Dean wants for the From field.

But the network still removes this information if the subscriber wants
anonymity. The stated purpose of the field is user identity, which implies
Personal Data of the subscriber. This is enough to argue that the network
must remove it if privacy is required, and this is certainly the application
of current legislation to ISDN.

What is the stated purpose of the From field - I do not think it is
sufficiently different for us to be _sure_ that we can avoid the
requirement.

So, the munging may still be required, but I certainly agree it should be
avoided wherever possible.

2) In formal terms, 'B2BUA == NOT Proxy'. We all know there will be/are
devices which behave very much like a proxy, but perform some additional
functions, such as these privacy services. This is fine. Everyone should be
aware that with only two terms defined 'B2BUA' and 'Proxy', such devices
fall into the B2BUA category.

...Mark

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 17 April 2002 17:36
> To: William Marshall; sip@ietf.org
> Subject: Re: [Sip] z9hG4bK forever ???
> 
> 
> Baill said:
> > All this discussion of proxy modification of the From and 
> To headers,
> > and the need for a B2BUA, seems to me to be a short-term 
> problem only.
> >
> > RFC3261 (2543bis-09) specifies matching of dialog-ids based only
> > on Call-ID and tags, not on display-name or on addr-spec.  So
> > when all endpoints are compliant with RFC3261, there will be no
> > problem with a proxy modifying the display-name or addr-spec in
> > the From header.
> 
> I disagree. I don't want the network mangling my user-to-user headers
> unless I've specifically requested it to do so. Specifically, the
> non-authenticated, non-mangled, non-protected From: field 
> offers utility
> which is DIFFERENT than that provided by network authenticated calling
> party ID, and I don't want to sacrifice that utility just because this
> is a capability the widely-deployed PSTN (as opposed to say Q.SIG)
> didn't have . . .
> 
> --
> Dean
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] z9hG4bK forever ???</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>I think we're about at the end of this one.</FONT>
</P>

<P><FONT SIZE=3D2>Two final points from me:</FONT>
<BR><FONT SIZE=3D2>1) Dean's desire to avoid having the From: field =
mangled unless he has specifically requested it is reasonable. I still =
think&nbsp; that the subscriber's wishes could overrule this =
though.</FONT></P>

<P><FONT SIZE=3D2>This is no different from the ISDN 'Special =
Arrangement' where the user inserts any number they like in the Calling =
Party Number field. The network passes it though with no checking, =
changing, nothing - I think this is analogous to what Dean wants for =
the From field.</FONT></P>

<P><FONT SIZE=3D2>But the network still removes this information if the =
subscriber wants anonymity. The stated purpose of the field is user =
identity, which implies Personal Data of the subscriber. This is enough =
to argue that the network must remove it if privacy is required, and =
this is certainly the application of current legislation to =
ISDN.</FONT></P>

<P><FONT SIZE=3D2>What is the stated purpose of the From field - I do =
not think it is sufficiently different for us to be _sure_ that we can =
avoid the requirement.</FONT></P>

<P><FONT SIZE=3D2>So, the munging may still be required, but I =
certainly agree it should be avoided wherever possible.</FONT>
</P>

<P><FONT SIZE=3D2>2) In formal terms, 'B2BUA =3D=3D NOT Proxy'. We all =
know there will be/are devices which behave very much like a proxy, but =
perform some additional functions, such as these privacy services. This =
is fine. Everyone should be aware that with only two terms defined =
'B2BUA' and 'Proxy', such devices fall into the B2BUA =
category.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Dean Willis [<A =
HREF=3D"mailto:dean.willis@softarmor.com">mailto:dean.willis@softarmor.c=
om</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 17 April 2002 17:36</FONT>
<BR><FONT SIZE=3D2>&gt; To: William Marshall; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] z9hG4bK forever ???</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Baill said:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; All this discussion of proxy modification =
of the From and </FONT>
<BR><FONT SIZE=3D2>&gt; To headers,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; and the need for a B2BUA, seems to me to =
be a short-term </FONT>
<BR><FONT SIZE=3D2>&gt; problem only.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; RFC3261 (2543bis-09) specifies matching of =
dialog-ids based only</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; on Call-ID and tags, not on display-name =
or on addr-spec.&nbsp; So</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; when all endpoints are compliant with =
RFC3261, there will be no</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; problem with a proxy modifying the =
display-name or addr-spec in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the From header.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I disagree. I don't want the network mangling =
my user-to-user headers</FONT>
<BR><FONT SIZE=3D2>&gt; unless I've specifically requested it to do so. =
Specifically, the</FONT>
<BR><FONT SIZE=3D2>&gt; non-authenticated, non-mangled, non-protected =
From: field </FONT>
<BR><FONT SIZE=3D2>&gt; offers utility</FONT>
<BR><FONT SIZE=3D2>&gt; which is DIFFERENT than that provided by =
network authenticated calling</FONT>
<BR><FONT SIZE=3D2>&gt; party ID, and I don't want to sacrifice that =
utility just because this</FONT>
<BR><FONT SIZE=3D2>&gt; is a capability the widely-deployed PSTN (as =
opposed to say Q.SIG)</FONT>
<BR><FONT SIZE=3D2>&gt; didn't have . . .</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; Dean</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: HTTP Digest Authentication Using AKA
	Author(s)	: R. Housley, W. Polk
	Filename	: draft-ietf-sip-digest-aka-00.txt
	Pages		: 17
	Date		: 17-Apr-02
	
The Hypertext Transfer Protocol (HTTP) Authentication Framework
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The Authentication and Key Agreement (AKA) mechanism performs user
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From daemon@optimus.ietf.org  Thu Apr 18 10:06:25 2002
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: <bernhard.honeisen@nokia.com>, <sip@ietf.org>
Subject: RE: [Sip] Correction:SIP Message Draft Fast WGLC
Date: Thu, 18 Apr 2002 08:10:57 -0500
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Hi Bernie,

You describe the concept of content indirection. The MESSAGE draft
intentionally does not specify that behavior, as it is really a more general
problem. It is particularly applicable to NOTIFY, and is probably to _any_
method that can carry a body.

Sean Olson recently published a draft collecting requirements for content
indirection. I expect work to continue in that are in the near future.

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> bernhard.honeisen@nokia.com
> Sent: Thursday, April 18, 2002 1:17 AM
> To: sip@ietf.org
> Subject: RE: [Sip] Correction:SIP Message Draft Fast WGLC
>
>
> Hi!
>
> I have a question related to the message draft:
>
> What should be used, if only a URL to the content (e.g. http),
> but not the content itself is delivered to the receiving UA, so
> that it is the receiving users decision, whether to download the
> content of not.
>
> This makes sense, if the _bandwidth_is_limited_
> and _large_messages_ are sent. This could be even a service
> (in some network element, e.g. store and forward server)
> provided to the user, to avoid that the channel down to the
> user agent gets congested for too long time.
>
> Is e.g. the Call-Info header field a good idea?
> Or should the payload be used for this?
> Better proposals?
>
> cheers,
>  Bernie
>
> > -----Original Message-----
> > From: ext Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: 12 April, 2002 22:17
> > To: sip@ietf.org
> > Cc: brian.rosen@maroni.com; rohan@cisco.com; jo@ipdialog.com;
> > bcampbell@dynamicsoft.com
> > Subject: [Sip] Correction:SIP Message Draft Fast WGLC
> >
> >
> >
> > Oops.
> >
> > Please review :
> >
> > http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-message-02.txt
> >
> > Until the -02 makes it into the draft repository. They're not
> > as fast as
> > I'm wishing for (ok, light speed is too slow, while we're at it!)
> >
> > Thanks,
> >
> > --
> > Dean
> >
> >
> > Original message:
> > -------------
> > We need to quickly review the SIM Message Method Extension. This draft
> > has been widely discussed, iterated several times in SIMPLE, and is on
> > its 2nd revision in SIP. The draft does not appear to be
> > controversial.
> >
> > See:
> >
> > http://search.ietf.org/internet-drafts/draft-ietf-sip-message-01.txt
> >
> > We expect to move to IETF last call next week, so please raise any
> > issues immediately.
> >
> > Ben Campbell will coordinate for the draft editors.
> >
> > -----------------
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: HTTP Digest Authentication Using AKA
	Author(s)	: A. Niemi, J. Arkko, V. Torvinen
	Filename	: draft-ietf-sip-digest-aka-00.txt
	Pages		: 17
	Date		: 17-Apr-02
	
The Hypertext Transfer Protocol (HTTP) Authentication Framework
includes two authentication schemes: Basic and Digest.  Both schemes
employ a shared secret based mechanism for access authentication.
The Authentication and Key Agreement (AKA) mechanism performs user
authentication and session key distribution in Universal Mobile
Telecommunications System (UMTS) networks.  AKA is a challenge-
response based mechanism that uses symmetric cryptography.  This memo
specifies an AKA based one-time password generation mechanism for
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From daemon@optimus.ietf.org  Thu Apr 18 12:36:52 2002
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> the above is consistent with what i have tried to say about this topic
> and contains nothing new.  this whole thread has thus been 
> more or less useless.
> 
> -- juha

I couldn't disagree more.

Our WG chairs have been enormously patient with the latest batch of rants and tirades on this subject. The quality of discussion has improved measurably since the one-hour RPID rathole in Minneapolis. Since the union of all proposed requirements is the null set, the most effective path forward will of necessity involve difficult compromise against less important requirements in order to satisfy those that will really matter over time. Frankly, Jon Peterson has done an excellent job of taking up this challenge and laying out some concrete proposals to address it.

While you are certainly entitled to your opinion, it should be abundantly clear by now that this set of issues is important to many in the sip community and isn't going away just because you would like to dismiss it. Based on list traffic, rough consensus *isn't* saying this is useless and it *isn't* saying we can ignore these issues and move on. 

--Andy



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From daemon@optimus.ietf.org  Thu Apr 18 15:57:37 2002
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Subject: [Sip] Re: Cookies or State
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I don't think the question should be one vs. the other.  More a matter
of merging the contents.

If state is chosen, then the syntax should be changed to be
consistent with http cookies, as is recommended in the cookies draft.
Also, requirements about sending state tokens for associated calls
need to be thought through more carefully.

If cookies is chosen, then the "Requires" mechanisms needs to be added, 
so that a proxy will know whether its cookies will be returned.  Also, 
the rules for UA inserting cookies into requests and responses need to be 
more carefully defined - the state draft deals with privacy (that ugly word 
again) concerns about sending cookies when the proxy that needs them
is not on the path of the request or response.

So I think the real question is which of two paths should be taken, since
the destination is the same for both.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Wed, 17 Apr 2002 15:27:37 -0400
Subject: [Sip] Cookies or State

There are two drafts that define how to save state in a UA for subsequent
use by a Proxy Server - State and Cookies.

State is part of the DCS work.  Cookies arose when IP assertions were 
made against State.  Cookies attempted to get around the IP issues
by defining it as exactly like HTTP cookies.  To our consternation,
the same IP claims were then made against Cookies!

We need to get this capability.  The question is, should we
base it on State or Cookies?

In my personal (biased) opinion we are better off with Cookies.
If anyone did want to fight the IP claim, you would clearly
be better off with 75% same language as HTTP cookies.

Brian

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Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft
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In a message dated 4/15/02 12:15:28 PM Eastern Daylight Time, 
huitema@windows.microsoft.com writes:


> I have heard many contributors assert that to get real anonymity, one
> will have to use a relay, e.g. an "anonymizing B2BUA". I would like to
> point out a failure in this line of reasoning: anonymizing relays have
> been tried before, for e-mail, and the experience was absolutely not
> conclusive. In fact, two different kind of anonymizers have been tried:
> free ones, that kept absolutely no record, and subscriber-based, which
> kept some records. Both have failed, for different reason.
> 

What you said in the rest of the e-mail about the failure of anonymizing 
B2BUAs sounded pretty "conclusive" to me. (rather than "not conclusive" as 
you said above). Can we agree that anonymizing B2BUA is not an acceptable 
answer to any privacy requirements?

Mike


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<HTML><FONT FACE=arial,helvetica><FONT  SIZE=2>In a message dated 4/15/02 12:15:28 PM Eastern Daylight Time, huitema@windows.microsoft.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=CITE style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I have heard many contributors assert that to get real anonymity, one
<BR>will have to use a relay, e.g. an "anonymizing B2BUA". I would like to
<BR>point out a failure in this line of reasoning: anonymizing relays have
<BR>been tried before, for e-mail, and the experience was absolutely not
<BR>conclusive. In fact, two different kind of anonymizers have been tried:
<BR>free ones, that kept absolutely no record, and subscriber-based, which
<BR>kept some records. Both have failed, for different reason.
<BR></FONT><FONT  COLOR="#000000" SIZE=3 FAMILY="SANSSERIF" FACE="Arial" LANG="0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR="#000000" SIZE=2 FAMILY="SANSSERIF" FACE="Arial" LANG="0">
<BR>What you said in the rest of the e-mail about the failure of anonymizing B2BUAs sounded pretty "conclusive" to me. (rather than "not conclusive" as you said above). Can we agree that anonymizing B2BUA is not an acceptable answer to any privacy requirements?
<BR>
<BR>Mike
<BR></FONT></HTML>

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From daemon@optimus.ietf.org  Thu Apr 18 18:55:24 2002
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I wanted to speak to some of these points.

I agree that privacy starts with the user agent - populating headers with
appropriately private values and so forth. I also agree that getting
dynamic, 'anonymous' IP addresses is a way to achieve the privacy that one
might otherwise request from an intermediary. I don't agree, however, that
this means we should abandon any line of reasoning that uses intermediaries.
Although the heyday of penet.fi is long-gone, there are still third-party
anonymizer services for SMTP and HTTP in operation today. There are sorts of
anonymity that are valuable and which rely on intermediaries. Sometimes, it
might not be obvious to users how 'private' their IP addresses are
(something you wouldn't want to be wrong about, eh?). Sometimes, users may
not have control over how the headers of their messages are populated.

Even getting privacy from the same service provider from whom you get SIP
registration services makes sense in some cases. If I buy my service from a
very large service provider (MSN, say), then simply revealing my service
provider is unlikely to reveal my identity - if MSN provided anonymization,
I'd feel relatively anonymous using their service. For many sorts of
interactions this may be an adequate amount of privacy - sort of like seeing
the postmark, with its zip code, on a letter without a return address.
Agreed that if the xenuphiles sued MSN for my anonymous wrongdoings, MSN
would undoubtedly pony up my name. But if you are doing things with
third-party-provided anonymity that will bring legal sanctions against you,
then you had better be sure you have a crusading anonymity provider that is
willing to become a political prisoner in your stead - even that is,
however, not entirely unheard of.

So in short, I see no harm in recognizing both anonymous IPs and
intermediaries as means to achieve privacy. As it happens, both are
mentioned in my new privacy draft.

Jon Peterson

NeuStar, Inc.

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Thursday, April 18, 2002 12:40 PM
To: huitema@windows.microsoft.com; sip@ietf.org
Subject: Re: Summary of RE: [Sip] Comment, SIP Privacy draft


In a message dated 4/15/02 12:15:28 PM Eastern Daylight Time,
huitema@windows.microsoft.com writes: 




I have heard many contributors assert that to get real anonymity, one 
will have to use a relay, e.g. an "anonymizing B2BUA". I would like to 
point out a failure in this line of reasoning: anonymizing relays have 
been tried before, for e-mail, and the experience was absolutely not 
conclusive. In fact, two different kind of anonymizers have been tried: 
free ones, that kept absolutely no record, and subscriber-based, which 
kept some records. Both have failed, for different reason. 




What you said in the rest of the e-mail about the failure of anonymizing
B2BUAs sounded pretty "conclusive" to me. (rather than "not conclusive" as
you said above). Can we agree that anonymizing B2BUA is not an acceptable
answer to any privacy requirements? 

Mike 



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<DIV>
<P><FONT size=2>I wanted to speak to some of these points.</FONT></P>
<P><FONT size=2>I agree that privacy starts with the user agent - populating 
headers with appropriately private values and so forth. I also agree that 
getting dynamic, 'anonymous' IP addresses is a way to achieve the privacy that 
one might otherwise request from an intermediary. I don't agree, however, that 
this means we should abandon any line of reasoning that uses intermediaries. 
Although the heyday of penet.fi is long-gone, there are still third-party 
anonymizer services for SMTP and HTTP in operation today. There are sorts of 
anonymity that are valuable and which rely on intermediaries. Sometimes, it 
might not be obvious to use<SPAN class=450182022-18042002>r</SPAN>s how 
'private' their IP addresses are (something you wouldn't want to be wrong about, 
eh?). Sometimes, users may not have control over how the headers of their 
messages are populated.</FONT></P>
<P><FONT size=2>Even getting privacy from the same service provider from whom 
you get SIP registration services makes sense in some cases. If I buy my service 
from a very large service provider (MSN, say), then simply revealing my service 
provider is unlikely to reveal my identity - if MSN provided anonymization, I'd 
feel relatively anonymous using their service. For many sorts of interactions 
this may be an adequate amount of privacy - sort of like see<SPAN 
class=450182022-18042002>ing</SPAN> the postmark, with its zip code, on a letter 
without a return address. Agreed that if the xenuphiles sued MSN for my 
anonymous wrongdoings, MSN would undoubtedly pony up my name. But if you are 
doing things with third-party-provided anonymity that will bring legal sanctions 
against you, then you had better be sure you have a crusading anonymity provider 
that is willing to become a political prisoner in your stead - even that is, 
however, not entirely unheard of.</FONT></P>
<P><FONT size=2>So in short, I see no harm in recognizing both anonymous IPs and 
intermediaries as means to achieve privacy.<SPAN class=450182022-18042002> As it 
happens, both are mentioned in my new privacy draft.</SPAN></FONT></P>
<P><FONT size=2>Jon Peterson</FONT></P>
<P><FONT size=2>NeuStar, Inc.</FONT></P></DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Thursday, April 18, 2002 12:40 
  PM<BR><B>To:</B> huitema@windows.microsoft.com; 
  sip@ietf.org<BR><B>Subject:</B> Re: Summary of RE: [Sip] Comment, SIP Privacy 
  draft<BR><BR></DIV></FONT><FONT face=arial,helvetica><FONT size=2>In a message 
  dated 4/15/02 12:15:28 PM Eastern Daylight Time, huitema@windows.microsoft.com 
  writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px" 
  TYPE="CITE">I have heard many contributors assert that to get real 
    anonymity, one <BR>will have to use a relay, e.g. an "anonymizing B2BUA". I 
    would like to <BR>point out a failure in this line of reasoning: anonymizing 
    relays have <BR>been tried before, for e-mail, and the experience was 
    absolutely not <BR>conclusive. In fact, two different kind of anonymizers 
    have been tried: <BR>free ones, that kept absolutely no record, and 
    subscriber-based, which <BR>kept some records. Both have failed, for 
    different reason. <BR></FONT><FONT face=Arial lang=0 size=3 
    FAMILY="SANSSERIF"></BLOCKQUOTE><BR></FONT><FONT face=Arial lang=0 size=2 
  FAMILY="SANSSERIF"><BR>What you said in the rest of the e-mail about the 
  failure of anonymizing B2BUAs sounded pretty "conclusive" to me. (rather than 
  "not conclusive" as you said above). Can we agree that anonymizing B2BUA is 
  not an acceptable answer to any privacy requirements? <BR><BR>Mike 
<BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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From daemon@optimus.ietf.org  Thu Apr 18 19:32:43 2002
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Date: Thu, 18 Apr 2002 16:06:40 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: sip@ietf.org
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Subject: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Hi,

I'd like to propose a concrete plan to move forward with a solution for
network-asserted identity and privacy.  I'm proposing this as an
individual (who is sick of reading about this topic on the mailing
list).  Below I've listed some of problems that various folks were
apparently trying to solve in this space.

1.  Provide a Network-Asserted ID
 a.   Within a trusted administrative domain or federation of domains
 b.   For interworking with PSTN mechanisms*
 c.   Usable across administrative boundaries

2.  Provide capability for Call Trace
 a.   Within a trusted administrative domain or federation of domains
 b.   For interworking with PSTN mechanisms*
 c.   Usable across administrative boundaries

3.  Provide privacy
 a.   by the user directly withholding information from the network
 b.   in the network for user-provided info at the user's request
 c.   in the network for user-provided info based on network policy
 d.   of network-provided information

4.  Allow for a user-provided "hint" for Network-Asserted ID

*(with gateways in the trusted domain or federation)

Most of the snags and disagreements we have had with the existing
Remote-Party-ID draft (draft-ietf-sip-privacy-04) are related to problems
1c, 2c, and 4.  Even a very scaled-down version of this draft would allow
for trusted intermediaries and user agents (including gateways) to
exchange network asserted identity and provide the capability for call
trace within a domain or federation of trusted domains.  The solution is
as simple as a single header which carries the identity in clear text,
which is added by the first proxy that authenticates the user, and
removed before it leaves the trust boundary.  This would solve
1a, 1b, 2a, and 2b.

Jon Peterson just drafted a Network-Asserted Identity proposal which
requires S/MIME signatures on each message.  This address the whole
category of problems related to 1 and 2 using cryptographic mechanisms.
While you could use this mechanism inside trusted networks, the
mechanism may be too heavyweight to prove practical for use with every
call in every network.  However, this is the only proposed mechanism
to solve 1c, and 2c in an elegant manner.

Jon also took a stab at the "longer-term" privacy requirements.  His draft
discusses how to solve problems 3a and 3b, and could be extended to
support 3d (as proposed in Jon's identity draft).  In addition, a network
has always been able to introduce an anonymizing B2BUA to solve problem
3c.

Finally, we have problem 4.  Some folks have proposed that
UAs should provide a hint using the same mechanism used to
communicate network asserted identity among trusted entities.
Others propose that authentication servers that create a
network asserted identity accept multiple "usernames" during
authentication and use this as the hint.  AFAIK these are the
only two choices, and we can come up with an answer to this
question in a separate thread.  The answer does not fundamentally
change any of the mechanisms we select to solve the other problems.

So, in summary, I'm proposing:

No additional work is required to implement 3c

Three short term deliverables
I) adopt draft-peterson-sip-privacy-longterm as a WG item to solve
   3a and 3b

II) write an extension to I) as a WG item for requesting privacy of
network asserted ID (however it is expressed) to support 3d.  This would
get used by III and IV.

III) pare down the mechanism in draft-ietf-sip-privacy to only address 1a,
1b, 2a, and 2b.  quickly get closure on which mechanism we use for 4, and
include it in the draft.  keep the current applicability statement.

One or more longer term deliverables, including
IV) draft-peterson-sip-identity ... to address a more general way to
solve 1 and 2 including 1c and 2c.  Note that this draft is the first
step toward providing real cryptographic identity services.  I think we
also need to address this topic as a charter item.

We might also provide an informational document describing how to map
privacy and identity between SIP and the PSTN.

hope this helps.
thanks,
-rohan


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From daemon@ns.ietf.org  Fri Apr 19 00:00:33 2002
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From: "Sreerekha Madhava Shenoy" <sreerekha.shenoy@wipro.com>
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hi
  We had a doubt about the significance of "maddr" parameter in the Contact
header.
  When the proxy gets the next hop information from the Location Service,
should it honour the "maddr" (when present) as the next hop address ? Or
should it take the   "host" part of the Contact header ?


Thanks
regards
rekha.


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From daemon@ns.ietf.org  Fri Apr 19 05:04:04 2002
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hi all,

where it's possible find the status of not renewed  and the cause...
of the inclusion in other draft as draft-sip-rfc2543


thanks

sal

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<DIV><FONT face=3DArial size=3D2>hi all,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>where it's possible find the status of =
not=20
renewed&nbsp;&nbsp;and the cause...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>of&nbsp;the inclusion in other draft as =

draft-sip-rfc2543</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>sal</FONT></DIV></BODY></HTML>

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From daemon@ns.ietf.org  Fri Apr 19 05:17:32 2002
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Date: Fri, 19 Apr 2002 11:27:47 +0300
From: Hatice ORBAY <orbayh@se.bel.alcatel.be>
Organization: Alcatel
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Hello,
If a response to  OPTIONS request is an 4xx message (420 or 486 etc...),
then does the UAS wait for an ACK from Proxy ? Or not, ACK is only a
part of the INVITE transaction?

Thanks in advance,
Hatice S. Orbay


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From daemon@ns.ietf.org  Fri Apr 19 05:53:32 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Rohan Mahy'" <rohan@cisco.com>, sip@ietf.org
Subject: RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
Date: Fri, 19 Apr 2002 10:22:36 +0100
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Good plan. I agree.

...Mark

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 19 April 2002 00:07
> To: sip@ietf.org
> Subject: [Sip] Proposed plan for Network-Asserted ID/Privacy
> 
> 
> Hi,
> 
> I'd like to propose a concrete plan to move forward with a 
> solution for
> network-asserted identity and privacy.  I'm proposing this as an
> individual (who is sick of reading about this topic on the mailing
> list).  Below I've listed some of problems that various folks were
> apparently trying to solve in this space.
> 
> 1.  Provide a Network-Asserted ID
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
> 
> 2.  Provide capability for Call Trace
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
> 
> 3.  Provide privacy
>  a.   by the user directly withholding information from the network
>  b.   in the network for user-provided info at the user's request
>  c.   in the network for user-provided info based on network policy
>  d.   of network-provided information
> 
> 4.  Allow for a user-provided "hint" for Network-Asserted ID
> 
> *(with gateways in the trusted domain or federation)
> 
> Most of the snags and disagreements we have had with the existing
> Remote-Party-ID draft (draft-ietf-sip-privacy-04) are related 
> to problems
> 1c, 2c, and 4.  Even a very scaled-down version of this draft 
> would allow
> for trusted intermediaries and user agents (including gateways) to
> exchange network asserted identity and provide the capability for call
> trace within a domain or federation of trusted domains.  The 
> solution is
> as simple as a single header which carries the identity in clear text,
> which is added by the first proxy that authenticates the user, and
> removed before it leaves the trust boundary.  This would solve
> 1a, 1b, 2a, and 2b.
> 
> Jon Peterson just drafted a Network-Asserted Identity proposal which
> requires S/MIME signatures on each message.  This address the whole
> category of problems related to 1 and 2 using cryptographic 
> mechanisms.
> While you could use this mechanism inside trusted networks, the
> mechanism may be too heavyweight to prove practical for use with every
> call in every network.  However, this is the only proposed mechanism
> to solve 1c, and 2c in an elegant manner.
> 
> Jon also took a stab at the "longer-term" privacy 
> requirements.  His draft
> discusses how to solve problems 3a and 3b, and could be extended to
> support 3d (as proposed in Jon's identity draft).  In 
> addition, a network
> has always been able to introduce an anonymizing B2BUA to 
> solve problem
> 3c.
> 
> Finally, we have problem 4.  Some folks have proposed that
> UAs should provide a hint using the same mechanism used to
> communicate network asserted identity among trusted entities.
> Others propose that authentication servers that create a
> network asserted identity accept multiple "usernames" during
> authentication and use this as the hint.  AFAIK these are the
> only two choices, and we can come up with an answer to this
> question in a separate thread.  The answer does not fundamentally
> change any of the mechanisms we select to solve the other problems.
> 
> So, in summary, I'm proposing:
> 
> No additional work is required to implement 3c
> 
> Three short term deliverables
> I) adopt draft-peterson-sip-privacy-longterm as a WG item to solve
>    3a and 3b
> 
> II) write an extension to I) as a WG item for requesting privacy of
> network asserted ID (however it is expressed) to support 3d.  
> This would
> get used by III and IV.
> 
> III) pare down the mechanism in draft-ietf-sip-privacy to 
> only address 1a,
> 1b, 2a, and 2b.  quickly get closure on which mechanism we 
> use for 4, and
> include it in the draft.  keep the current applicability statement.
> 
> One or more longer term deliverables, including
> IV) draft-peterson-sip-identity ... to address a more general way to
> solve 1 and 2 including 1c and 2c.  Note that this draft is the first
> step toward providing real cryptographic identity services.  
> I think we
> also need to address this topic as a charter item.
> 
> We might also provide an informational document describing how to map
> privacy and identity between SIP and the PSTN.
> 
> hope this helps.
> thanks,
> -rohan
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
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<P><FONT SIZE=2>Good plan. I agree.</FONT>
</P>

<P><FONT SIZE=2>...Mark</FONT>
</P>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Rohan Mahy [<A HREF="mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: 19 April 2002 00:07</FONT>
<BR><FONT SIZE=2>&gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Subject: [Sip] Proposed plan for Network-Asserted ID/Privacy</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Hi,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I'd like to propose a concrete plan to move forward with a </FONT>
<BR><FONT SIZE=2>&gt; solution for</FONT>
<BR><FONT SIZE=2>&gt; network-asserted identity and privacy.&nbsp; I'm proposing this as an</FONT>
<BR><FONT SIZE=2>&gt; individual (who is sick of reading about this topic on the mailing</FONT>
<BR><FONT SIZE=2>&gt; list).&nbsp; Below I've listed some of problems that various folks were</FONT>
<BR><FONT SIZE=2>&gt; apparently trying to solve in this space.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 1.&nbsp; Provide a Network-Asserted ID</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; a.&nbsp;&nbsp; Within a trusted administrative domain or federation of domains</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; b.&nbsp;&nbsp; For interworking with PSTN mechanisms*</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; c.&nbsp;&nbsp; Usable across administrative boundaries</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 2.&nbsp; Provide capability for Call Trace</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; a.&nbsp;&nbsp; Within a trusted administrative domain or federation of domains</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; b.&nbsp;&nbsp; For interworking with PSTN mechanisms*</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; c.&nbsp;&nbsp; Usable across administrative boundaries</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 3.&nbsp; Provide privacy</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; a.&nbsp;&nbsp; by the user directly withholding information from the network</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; b.&nbsp;&nbsp; in the network for user-provided info at the user's request</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; c.&nbsp;&nbsp; in the network for user-provided info based on network policy</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; d.&nbsp;&nbsp; of network-provided information</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 4.&nbsp; Allow for a user-provided &quot;hint&quot; for Network-Asserted ID</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; *(with gateways in the trusted domain or federation)</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Most of the snags and disagreements we have had with the existing</FONT>
<BR><FONT SIZE=2>&gt; Remote-Party-ID draft (draft-ietf-sip-privacy-04) are related </FONT>
<BR><FONT SIZE=2>&gt; to problems</FONT>
<BR><FONT SIZE=2>&gt; 1c, 2c, and 4.&nbsp; Even a very scaled-down version of this draft </FONT>
<BR><FONT SIZE=2>&gt; would allow</FONT>
<BR><FONT SIZE=2>&gt; for trusted intermediaries and user agents (including gateways) to</FONT>
<BR><FONT SIZE=2>&gt; exchange network asserted identity and provide the capability for call</FONT>
<BR><FONT SIZE=2>&gt; trace within a domain or federation of trusted domains.&nbsp; The </FONT>
<BR><FONT SIZE=2>&gt; solution is</FONT>
<BR><FONT SIZE=2>&gt; as simple as a single header which carries the identity in clear text,</FONT>
<BR><FONT SIZE=2>&gt; which is added by the first proxy that authenticates the user, and</FONT>
<BR><FONT SIZE=2>&gt; removed before it leaves the trust boundary.&nbsp; This would solve</FONT>
<BR><FONT SIZE=2>&gt; 1a, 1b, 2a, and 2b.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Jon Peterson just drafted a Network-Asserted Identity proposal which</FONT>
<BR><FONT SIZE=2>&gt; requires S/MIME signatures on each message.&nbsp; This address the whole</FONT>
<BR><FONT SIZE=2>&gt; category of problems related to 1 and 2 using cryptographic </FONT>
<BR><FONT SIZE=2>&gt; mechanisms.</FONT>
<BR><FONT SIZE=2>&gt; While you could use this mechanism inside trusted networks, the</FONT>
<BR><FONT SIZE=2>&gt; mechanism may be too heavyweight to prove practical for use with every</FONT>
<BR><FONT SIZE=2>&gt; call in every network.&nbsp; However, this is the only proposed mechanism</FONT>
<BR><FONT SIZE=2>&gt; to solve 1c, and 2c in an elegant manner.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Jon also took a stab at the &quot;longer-term&quot; privacy </FONT>
<BR><FONT SIZE=2>&gt; requirements.&nbsp; His draft</FONT>
<BR><FONT SIZE=2>&gt; discusses how to solve problems 3a and 3b, and could be extended to</FONT>
<BR><FONT SIZE=2>&gt; support 3d (as proposed in Jon's identity draft).&nbsp; In </FONT>
<BR><FONT SIZE=2>&gt; addition, a network</FONT>
<BR><FONT SIZE=2>&gt; has always been able to introduce an anonymizing B2BUA to </FONT>
<BR><FONT SIZE=2>&gt; solve problem</FONT>
<BR><FONT SIZE=2>&gt; 3c.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Finally, we have problem 4.&nbsp; Some folks have proposed that</FONT>
<BR><FONT SIZE=2>&gt; UAs should provide a hint using the same mechanism used to</FONT>
<BR><FONT SIZE=2>&gt; communicate network asserted identity among trusted entities.</FONT>
<BR><FONT SIZE=2>&gt; Others propose that authentication servers that create a</FONT>
<BR><FONT SIZE=2>&gt; network asserted identity accept multiple &quot;usernames&quot; during</FONT>
<BR><FONT SIZE=2>&gt; authentication and use this as the hint.&nbsp; AFAIK these are the</FONT>
<BR><FONT SIZE=2>&gt; only two choices, and we can come up with an answer to this</FONT>
<BR><FONT SIZE=2>&gt; question in a separate thread.&nbsp; The answer does not fundamentally</FONT>
<BR><FONT SIZE=2>&gt; change any of the mechanisms we select to solve the other problems.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; So, in summary, I'm proposing:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; No additional work is required to implement 3c</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Three short term deliverables</FONT>
<BR><FONT SIZE=2>&gt; I) adopt draft-peterson-sip-privacy-longterm as a WG item to solve</FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp;&nbsp; 3a and 3b</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; II) write an extension to I) as a WG item for requesting privacy of</FONT>
<BR><FONT SIZE=2>&gt; network asserted ID (however it is expressed) to support 3d.&nbsp; </FONT>
<BR><FONT SIZE=2>&gt; This would</FONT>
<BR><FONT SIZE=2>&gt; get used by III and IV.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; III) pare down the mechanism in draft-ietf-sip-privacy to </FONT>
<BR><FONT SIZE=2>&gt; only address 1a,</FONT>
<BR><FONT SIZE=2>&gt; 1b, 2a, and 2b.&nbsp; quickly get closure on which mechanism we </FONT>
<BR><FONT SIZE=2>&gt; use for 4, and</FONT>
<BR><FONT SIZE=2>&gt; include it in the draft.&nbsp; keep the current applicability statement.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; One or more longer term deliverables, including</FONT>
<BR><FONT SIZE=2>&gt; IV) draft-peterson-sip-identity ... to address a more general way to</FONT>
<BR><FONT SIZE=2>&gt; solve 1 and 2 including 1c and 2c.&nbsp; Note that this draft is the first</FONT>
<BR><FONT SIZE=2>&gt; step toward providing real cryptographic identity services.&nbsp; </FONT>
<BR><FONT SIZE=2>&gt; I think we</FONT>
<BR><FONT SIZE=2>&gt; also need to address this topic as a charter item.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; We might also provide an informational document describing how to map</FONT>
<BR><FONT SIZE=2>&gt; privacy and identity between SIP and the PSTN.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; hope this helps.</FONT>
<BR><FONT SIZE=2>&gt; thanks,</FONT>
<BR><FONT SIZE=2>&gt; -rohan</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; _______________________________________________</FONT>
<BR><FONT SIZE=2>&gt; Sip mailing list&nbsp; <A HREF="https://www1.ietf.org/mailman/listinfo/sip" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=2>&gt; This list is for NEW development of the core SIP Protocol</FONT>
<BR><FONT SIZE=2>&gt; Use sip-implementors@cs.columbia.edu for questions on current sip</FONT>
<BR><FONT SIZE=2>&gt; Use sipping@ietf.org for new developments on the application of sip</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
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To: Hatice ORBAY <orbayh@se.bel.alcatel.be>
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Subject: Re: [Sip] ACK for 4xx in response to OPTIONS
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For 3xx+ responses ACK is part of the INVITE transaction, only for 2xx
response its not a part of INVITE transaction.

Also, for responses to requests other than INVITE, ACK is not sent.

Ajit

Hatice ORBAY wrote:

> Hello,
> If a response to  OPTIONS request is an 4xx message (420 or 486 etc...),
> then does the UAS wait for an ACK from Proxy ? Or not, ACK is only a
> part of the INVITE transaction?
>
> Thanks in advance,
> Hatice S. Orbay
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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Hi,

ACK is used with INVITEs ONLY. It may, or may not, depending on the status
code in the response, be part of the INVITE transaction itself, or a
separate transaction.

See the spec for more details...

Regards,

Christer Holmberg
Ericsson Finland

Hatice ORBAY wrote:

> Hello,
> If a response to  OPTIONS request is an 4xx message (420 or 486 etc...),
> then does the UAS wait for an ACK from Proxy ? Or not, ACK is only a
> part of the INVITE transaction?
>
> Thanks in advance,
> Hatice S. Orbay
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Session Initiation Protocol Extension for Instant 
                          Messaging
	Author(s)	: B. Campbell, J. Rosenberg
	Filename	: draft-ietf-sip-message-03.txt
	Pages		: 19
	Date		: 18-Apr-02
	
Instant Messageing (IM) refers to the transfer of messages between
users in near real-time.  These messages are usually, but not
required to be, short.  IMs are often used in a conversational mode,
that is, the transfer of messages back and forth is fast enough for
participants to maintain an interactive conversation.
The MESSAGE method is an extension to the Session Initation Protocol
(SIP) that allows the transfer of Instant Messages.  MESSAGE requests
carry the content in the form of MIME body parts.  MESSAGE requests
do not themselves initiate a SIP dialog; under normal usage each
Instant Message stands alone, much like pager messages.  MESSAGE
requests may be sent in the context of a dialog initiated by some
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Enhancements for Authenticated Identity Management in 
                          the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sip-identity-00.txt
	Pages		: 25
	Date		: 18-Apr-02
	
The existing mechanisms for expressing identity in the Session
Initiation Protocol oftentimes do not permit an administrative domain
to verify securely the identity of the originator of a request.  This
document recommends practices and conventions for authenticating end
users, and proposes a way to distribute secure authenticated
identities within SIP messages.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: A Privacy Mechanism for the Session Initiation 
                          Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sip-privacy-longterm-00.txt
	Pages		: 26
	Date		: 18-Apr-02
	
This document defines new mechanisms for the Session Initiation
Protocol (SIP) in support of privacy.  Specifically, guidelines are
provided for the creation of messages that do not divulge personal
identity information.  A new 'privacy service' logical role for
intermediaries is defined to answer some privacy requirements that
user agents cannot satisfy themselves.  Finally, means are presented
by which a user can request particular functions from a privacy
service.

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From daemon@ns.ietf.org  Fri Apr 19 10:12:16 2002
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From: "ROOPA RAO" <roopa_k_rao@hotmail.com>
To: sip@ietf.org
Date: Fri, 19 Apr 2002 09:07:28 +0530
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Subject: [Sip] Queries regd. SIP
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Hello,

    I am new to SIP and have been trying to learn more about it since
I would like to work on a project in this area.

I have some very basic queries in this. Any help would be greatly 
appreciated.

I would like to know what exactly are the applications/ uses of this SIP
protocol. I know that it can be used for Internet Telephony applications
but i am not clear as to if i build some product/software on this who are 
the likely customers/ what is the market.

Also when there are so many open source codes and applications available for 
free on the internet why should anyone build similar
sort of applications ?

Please let me know where i can get more information on such issues.

Thanks,
R.P.






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From daemon@ns.ietf.org  Fri Apr 19 10:33:44 2002
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Date: Fri, 19 Apr 2002 09:52:42 -0400
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To: Rohan Mahy <rohan@cisco.com>
CC: sip@ietf.org
Subject: Re: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Rohan Mahy wrote:

> Hi,
>
> I'd like to propose a concrete plan to move forward with a solution for
> network-asserted identity and privacy.  I'm proposing this as an
> individual (who is sick of reading about this topic on the mailing
> list).  Below I've listed some of problems that various folks were
> apparently trying to solve in this space.
>
> 1.  Provide a Network-Asserted ID
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
>
> 2.  Provide capability for Call Trace
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
>
> 3.  Provide privacy
>  a.   by the user directly withholding information from the network
>  b.   in the network for user-provided info at the user's request
>  c.   in the network for user-provided info based on network policy
>  d.   of network-provided information
>
> 4.  Allow for a user-provided "hint" for Network-Asserted ID
>
> *(with gateways in the trusted domain or federation)
>
> Most of the snags and disagreements we have had with the existing
> Remote-Party-ID draft (draft-ietf-sip-privacy-04) are related to problems
> 1c, 2c, and 4.  Even a very scaled-down version of this draft would allow
> for trusted intermediaries and user agents (including gateways) to
> exchange network asserted identity and provide the capability for call
> trace within a domain or federation of trusted domains.  The solution is
> as simple as a single header which carries the identity in clear text,
> which is added by the first proxy that authenticates the user, and
> removed before it leaves the trust boundary.

You need to be more clear. 1 and 2 talk about administrative domains and
administrative boundaries and now you are talking about trust boundaries. How
do you define an administrative domain, an administrative boundary, and a
trust boundary ?


> This would solve
> 1a, 1b, 2a, and 2b.

No. The UA is (normally) untrusted but still needs it. If you were only
talking about the case when privacy was requested, then it doesn't address the
requirement to provide call trace for anonymous calls. The user is obviously
untrusted and hence cannot be given the caller-id in clear-text. The privacy
draft solves this by handing out an encrypted Remote-Party-Id which can
subsequently be used for call trace (as well as other features for that
matter, e.g. anonymous call return).

<snip>

>
> Finally, we have problem 4.  Some folks have proposed that
> UAs should provide a hint using the same mechanism used to
> communicate network asserted identity among trusted entities.

> Others propose that authentication servers that create a

> network asserted identity accept multiple "usernames" during

> authentication and use this as the hint.  AFAIK these are the
> only two choices, and we can come up with an answer to this
> question in a separate thread.  The answer does not fundamentally
> change any of the mechanisms we select to solve the other problems.

Well it certainly impacts it. If you go with the first solution then you also
accept that Remote-Party-Id can be provided by an untrusted entity, so I don't
think we can solve these independently.

>
>
> So, in summary, I'm proposing:
>
> No additional work is required to implement 3c
>
> Three short term deliverables
> I) adopt draft-peterson-sip-privacy-longterm as a WG item to solve
>    3a and 3b
>

It's unclear that 3b is independent of which mechanism you choose to provide
the hint above.


>
> II) write an extension to I) as a WG item for requesting privacy of
> network asserted ID (however it is expressed) to support 3d.  This would
> get used by III and IV.
>
> III) pare down the mechanism in draft-ietf-sip-privacy to only address 1a,
> 1b, 2a, and 2b.  quickly get closure on which mechanism we use for 4, and
> include it in the draft.  keep the current applicability statement.
>

As mentioned above, I'm not clear on exactly what 1a, 1b, 2a and 2b are, but
if the "network asserted identity" draft would neither address privacy for
network asserted identity, nor call trace for anonymous calls, then I
certainly disagree with this.

-- Flemming


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From daemon@ns.ietf.org  Fri Apr 19 10:43:45 2002
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Subject: RE: [Sip] Queries regd. SIP
From: "ranjitka@email.masconit.com" <ranjitka@email.masconit.com>
Date: Fri, 19 Apr 2002 10:08:42 -0400
To: "roopa_k_rao@hotmail.com" <roopa_k_rao@hotmail.com>,
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Hi Roopa,

As u said SIP is mainly used in VoIP. It can be used in conjunction with other protocols like MGCP, RADIUS, COPS etc to build applications in Iternet telephony. 

U can know more about latest happenings in SIP at http://www.sipcentre.com

HTH
Regards
Ranjit

Original Message:
-----------------
From: ROOPA RAO roopa_k_rao@hotmail.com
Date: Thu, 18 Apr 2002 22:37:28 -0500
To: sip@ietf.org
Subject: [Sip] Queries regd. SIP


Hello,

    I am new to SIP and have been trying to learn more about it since
I would like to work on a project in this area.

I have some very basic queries in this. Any help would be greatly 
appreciated.

I would like to know what exactly are the applications/ uses of this SIP
protocol. I know that it can be used for Internet Telephony applications
but i am not clear as to if i build some product/software on this who
are 
the likely customers/ what is the market.

Also when there are so many open source codes and applications available
for 
free on the internet why should anyone build similar
sort of applications ?

Please let me know where i can get more information on such issues.

Thanks,
R.P.






_________________________________________________________________
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From daemon@ns.ietf.org  Fri Apr 19 11:27:24 2002
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Date: Fri, 19 Apr 2002 09:47:04 -0500
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] Queries regd. SIP
In-reply-to: <F232aRJe7l91rgAmZHh0000b092@hotmail.com>
To: "'ROOPA RAO'" <roopa_k_rao@hotmail.com>, sip@ietf.org
Message-id: <000c01c1e7b1$12ff3f70$fae023a6@hsinnreich2>
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> I would like to know what exactly are the applications/ uses 
> of this SIP protocol. 

Please see the white papers at www.sipforum.org

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of ROOPA RAO
> Sent: Thursday, April 18, 2002 10:37 PM
> To: sip@ietf.org
> Subject: [Sip] Queries regd. SIP
> 
> 
> Hello,
> 
>     I am new to SIP and have been trying to learn more about 
> it since I would like to work on a project in this area.
> 
> I have some very basic queries in this. Any help would be greatly 
> appreciated.
> 
> I would like to know what exactly are the applications/ uses 
> of this SIP protocol. I know that it can be used for Internet 
> Telephony applications but i am not clear as to if i build 
> some product/software on this who are 
> the likely customers/ what is the market.
> 
> Also when there are so many open source codes and 
> applications available for 
> free on the internet why should anyone build similar
> sort of applications ?
> 
> Please let me know where i can get more information on such issues.
> 
> Thanks,
> R.P.
> 
> 
> 
> 
> 
> 
> _________________________________________________________________
> Chat with friends online, try MSN Messenger: http://messenger.msn.com
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


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From daemon@ns.ietf.org  Fri Apr 19 12:00:35 2002
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Date: Fri, 19 Apr 2002 10:23:45 -0500
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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Subject: [Sip] SIP support for IN services
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Folks:

Enclosed is an I-D for review that describes SIP support for IN
services.  The I-D represents agreement of the IETF SIN (SIP/IN) design
team and is intended for publication as an informational RFC.  It has
been through Last Call in the IETF SIN DT and the comments of the WGs
identified in the To list of this email are now being solicited.

http://search.ietf.org/internet-drafts/draft-gurbani-sin-00.txt

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@ns.ietf.org  Fri Apr 19 12:35:42 2002
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From: "Mark Watson"<mwatson@nortelnetworks.com>
To: "'Flemming Andreasen'" <fandreas@cisco.com>,
        Rohan Mahy
	 <rohan@cisco.com>
Cc: sip@ietf.org
Subject: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Priv
	acy
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Rohan suggested a separate thread for...

> > 4.  Allow for a user-provided "hint" for Network-Asserted ID
<snip>
> >
> > Finally, we have problem 4.  Some folks have proposed that
> > UAs should provide a hint using the same mechanism used to
> > communicate network asserted identity among trusted entities.
> 
> > Others propose that authentication servers that create a
> 
> > network asserted identity accept multiple "usernames" during
> 
> > authentication and use this as the hint.  AFAIK these are the
> > only two choices, and we can come up with an answer to this
> > question in a separate thread.  The answer does not fundamentally
> > change any of the mechanisms we select to solve the other problems.
> 
> Well it certainly impacts it. If you go with the first 
> solution then you also
> accept that Remote-Party-Id can be provided by an untrusted 
> entity, so I don't
> think we can solve these independently.

I diagree. Solve the other problems first. Then we have three choices:
1) Use the same header for the 'hints'
2) Define another header for the 'hints'
3) Assume the 'hint' in in the authentication 'username'

If you choose (1), then you may need something extra in that header to
distinguish between the two use-cases.

...Mark 

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<META NAME="Generator" CONTENT="MS Exchange Server version 5.5.2655.35">
<TITLE>Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy</TITLE>
</HEAD>
<BODY>
<BR>
<BR>

<P><FONT SIZE=2>Rohan suggested a separate thread for...</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; 4.&nbsp; Allow for a user-provided &quot;hint&quot; for Network-Asserted ID</FONT>
<BR><FONT SIZE=2>&lt;snip&gt;</FONT>
<BR><FONT SIZE=2>&gt; &gt;</FONT>
<BR><FONT SIZE=2>&gt; &gt; Finally, we have problem 4.&nbsp; Some folks have proposed that</FONT>
<BR><FONT SIZE=2>&gt; &gt; UAs should provide a hint using the same mechanism used to</FONT>
<BR><FONT SIZE=2>&gt; &gt; communicate network asserted identity among trusted entities.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; Others propose that authentication servers that create a</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; network asserted identity accept multiple &quot;usernames&quot; during</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; &gt; authentication and use this as the hint.&nbsp; AFAIK these are the</FONT>
<BR><FONT SIZE=2>&gt; &gt; only two choices, and we can come up with an answer to this</FONT>
<BR><FONT SIZE=2>&gt; &gt; question in a separate thread.&nbsp; The answer does not fundamentally</FONT>
<BR><FONT SIZE=2>&gt; &gt; change any of the mechanisms we select to solve the other problems.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Well it certainly impacts it. If you go with the first </FONT>
<BR><FONT SIZE=2>&gt; solution then you also</FONT>
<BR><FONT SIZE=2>&gt; accept that Remote-Party-Id can be provided by an untrusted </FONT>
<BR><FONT SIZE=2>&gt; entity, so I don't</FONT>
<BR><FONT SIZE=2>&gt; think we can solve these independently.</FONT>
</P>

<P><FONT SIZE=2>I diagree. Solve the other problems first. Then we have three choices:</FONT>
<BR><FONT SIZE=2>1) Use the same header for the 'hints'</FONT>
<BR><FONT SIZE=2>2) Define another header for the 'hints'</FONT>
<BR><FONT SIZE=2>3) Assume the 'hint' in in the authentication 'username'</FONT>
</P>

<P><FONT SIZE=2>If you choose (1), then you may need something extra in that header to distinguish between the two use-cases.</FONT>
</P>

<P><FONT SIZE=2>...Mark </FONT>
</P>

</BODY>
</HTML>
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Juan-Carlos.Rojas@alcatel.fr
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Subject: [Sip] Re: Manyfolks using SPDng
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Hi Juan Carlos,

There is not, as far as I know. However, it would be better to get the
necessary implementation experience with manyfolks and SDP, and once we
obtain this experience, elaborate a mechanism for SDPng.

Regards,

Gonzalo

Juan-Carlos.Rojas@alcatel.fr wrote:
> 
> Hello Gonzalo,
> 
> Is there any project to create an internet-draft (or to update the existing
> one) for the preconditions mechanism currently defined in the manyfolks
> draft, but using SDPng (and so, XML based) ?
> 
> Thank you for your answer
> 
> Best regards
> Juan Carlos

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo



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Date: Fri, 19 Apr 2002 19:51:14 +0300
To: "Mark Watson"<mwatson@nortelnetworks.com>
Cc: "'Flemming Andreasen'" <fandreas@cisco.com>,
        Rohan Mahy
	 <rohan@cisco.com>, sip@ietf.org
Subject: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Priv
	acy
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Mark Watson writes:

 > I diagree. Solve the other problems first. Then we have three choices:
 > 1) Use the same header for the 'hints'
 > 2) Define another header for the 'hints'
 > 3) Assume the 'hint' in in the authentication 'username'

(3) doesn't work.  my username is jh@song.fi, but my from uri that i
want to use, can be, for example, tel+358441234567.

today i'm using the from uri as the hint, which works fine if the user
is not requesting anonymity.  if the user is requesting anonymity and
the call goes to another sip user, then either a b2bua or a new "hint"
header, such as really-from, is needed.

-- juha


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Subject: RE: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: <jh@lohi.eng.song.fi>, "Mark Watson" <mwatson@nortelnetworks.com>
Cc: "Flemming Andreasen" <fandreas@cisco.com>, "Rohan Mahy" <rohan@cisco.com>,
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>  > I diagree. Solve the other problems first. Then we have three
choices:
>  > 1) Use the same header for the 'hints'
>  > 2) Define another header for the 'hints'
>  > 3) Assume the 'hint' in in the authentication 'username'
> 
> (3) doesn't work.  my username is jh@song.fi, but my from uri that i
> want to use, can be, for example, tel+358441234567.

In fact, e-mail today uses two different fields: the "SMTP From:"
parameter defines who is posting the message, and the "From:" field
defines the user who is supposed to get the replies. (OK, there is also
a "sender" field, when someone sends a message "on behalf" of other
user(s), but we won't get there.) By this logic, we would need a "SIP
From", identifying the account sending the message, in addition to the
current "From" field, wich identifies the user.

-- Christian Huitema

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From daemon@ns.ietf.org  Fri Apr 19 15:13:33 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>,
        Robert Sparks
	 <rsparks@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Fri, 19 Apr 2002 14:37:38 -0400
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Why is such information useful if it's unauthenticated?

Are you planning to use it to implement policy? If so,
inclusion of such a header only sets a trap for naive
implementors.

If not, are you envisioning this information will be presented
to users? If that's your goal, you'll want to do something
that works with deployed phones. As a convention (and I'm not
proposing that this be standardized), you could obtain the
desired effect with something like:

INVITE sip:bob@example.com SIP/2.0
From: "Adam Roach forwarded by Robert Sparks"
<sip:adam@dynamicsoft.com>;tag=12397
...

Right?

I feel quite strongly that we should *not* include this sort of information
in the current REFER draft, and work on providing an *authenticated*
version of this draft at a later date.

/a

> -----Original Message-----
> From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> Sent: Wednesday, April 10, 2002 1:51
> To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> I agree with what Patrick Mourot says. Let's have a basic 
> refer capability
> that provides an unsecured identity of A to C, and then leave 
> it to C to
> decide whether and how to verify the identity of A. This will 
> depend on
> whether it is attended or unattended transfer - with attended 
> transfer the
> call A-C will exist and will be identified by the Replaces 
> header. This may
> be sufficient, but if not, verification can be done in the 
> context of that
> dialog A-C.
> 
>  ---------------------------------------------------------------
>  John Elwell (john.elwell@siemenscomms.co.uk)
>  Siemens Communications Limited,
>  ---------------------------------------------------------------
>  Internet communications are not secure and therefore Siemens
>  Communications Limited does not accept legal responsibility for the
>  contents of this message. Any views or opinions presented are solely
>  those of the author and do not necessarily represent those of Siemens
>  Communications Limited unless otherwise specifically stated.
>  
> 
> 
> -----Original Message-----
> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> Sent: Tuesday, April 09, 2002 5:05 PM
> To: rsparks@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> 
> Sirs,
> 
> I think there are plainty of cases where C :
> -will not need to know that the call was started by A read in 
> Referred-By
> -or will trust B (i.e: intranet cases).
> 
> Protecting Referred-By also means protecting others headers.
> 
> This means that we should avoid all solutions that are not optional
> regarding C 
> policy.
> 
> A VERIFYing approach starting at C is better suited; It 
> allows C to decide
> to 
> verify or not A identity.
> If C does NOT want to or does NOT support VERIFY the call 
> flows remains 
> consistent with current behavior.
> 
> PS: Referred-By is a means for carrying some useful Information.
> 
> That's my 2 cents.
> 
> Best Regards,
> 
> Patrick
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From daemon@ns.ietf.org  Fri Apr 19 15:14:12 2002
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	Fri, 19 Apr 2002 14:49:35 -0400 (EDT)
Subject: RE: [Sip] REFER security options - removing Referred-By
To: Adam Roach <adam@dynamicsoft.com>
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>,
        Robert Sparks <rsparks@dynamicsoft.com>, sip@ietf.org
Date: Fri, 19 Apr 2002 14:49:33 -0400
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 The information could be useful because C could authenticate the identity
outside of the SIP protocol if that is felt necesary. In many cases such
authentication is not required. There can be other means extra-protocol  of
dealing with abusive callers.






Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM

Sent by:  sip-admin@ietf.org


To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
      "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>, Robert
      Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
cc:

Subject:  RE: [Sip] REFER security options - removing Referred-By


Why is such information useful if it's unauthenticated?

Are you planning to use it to implement policy? If so,
inclusion of such a header only sets a trap for naive
implementors.

If not, are you envisioning this information will be presented
to users? If that's your goal, you'll want to do something
that works with deployed phones. As a convention (and I'm not
proposing that this be standardized), you could obtain the
desired effect with something like:

INVITE sip:bob@example.com SIP/2.0
From: "Adam Roach forwarded by Robert Sparks"
<sip:adam@dynamicsoft.com>;tag=12397
...

Right?

I feel quite strongly that we should *not* include this sort of information
in the current REFER draft, and work on providing an *authenticated*
version of this draft at a later date.

/a

> -----Original Message-----
> From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> Sent: Wednesday, April 10, 2002 1:51
> To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
> I agree with what Patrick Mourot says. Let's have a basic
> refer capability
> that provides an unsecured identity of A to C, and then leave
> it to C to
> decide whether and how to verify the identity of A. This will
> depend on
> whether it is attended or unattended transfer - with attended
> transfer the
> call A-C will exist and will be identified by the Replaces
> header. This may
> be sufficient, but if not, verification can be done in the
> context of that
> dialog A-C.
>
>  ---------------------------------------------------------------
>  John Elwell (john.elwell@siemenscomms.co.uk)
>  Siemens Communications Limited,
>  ---------------------------------------------------------------
>  Internet communications are not secure and therefore Siemens
>  Communications Limited does not accept legal responsibility for the
>  contents of this message. Any views or opinions presented are solely
>  those of the author and do not necessarily represent those of Siemens
>  Communications Limited unless otherwise specifically stated.
>
>
>
> -----Original Message-----
> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> Sent: Tuesday, April 09, 2002 5:05 PM
> To: rsparks@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
>
> Sirs,
>
> I think there are plainty of cases where C :
> -will not need to know that the call was started by A read in
> Referred-By
> -or will trust B (i.e: intranet cases).
>
> Protecting Referred-By also means protecting others headers.
>
> This means that we should avoid all solutions that are not optional
> regarding C
> policy.
>
> A VERIFYing approach starting at C is better suited; It
> allows C to decide
> to
> verify or not A identity.
> If C does NOT want to or does NOT support VERIFY the call
> flows remains
> consistent with current behavior.
>
> PS: Referred-By is a means for carrying some useful Information.
>
> That's my 2 cents.
>
> Best Regards,
>
> Patrick
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: William Marshall <wtm@research.att.com>
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To: rohan@cisco.com
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Subject: Re: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Rohan Mahy wrote:
> I'd like to propose a concrete plan to move forward with a solution for
> network-asserted identity and privacy.  I'm proposing this as an
> individual (who is sick of reading about this topic on the mailing
> list).  Below I've listed some of problems that various folks were
> apparently trying to solve in this space.
>
> 1.  Provide a Network-Asserted ID
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
>

Once a proxy has determined the value of this "Network-Asserted ID",
there are actually four possibilities to consider.
   a.   Sending it to another proxy within a trusted admin domain 
		or federation of domains
   b.   Sending it to a PSTN gateway within a trusted admin domain
		or federation of domains
   c.   Sending it to a proxy outside the trust admin domain
		or federation of domains
   d.   Sending it to a UA

Unfortunalely caller-id and call-trace are useless without choice (d).
So my first comment is that 1d and 2d are needed in your itemized list 
of problems to solve.

I think it is a major simplification to reduce these four cases to only
two:
   a.  Sending it to a network element within a trusted admin domain
		or federation of domains
   b.  Sending it to a network element outside the trusted admin domain
		or federation of domains
(and who says PSTN gateways are always trusted? Should the GW generate
a private used-once phone number for the call attempt?)


And, as much as I'd like to believe your comment that no additional work
is required to implement 3c, are we really willing to leave it that all
services will break if anonymity is requested?  Or should services
be designed with this in mind?

Bill Marshall
wtm@research.att.com

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To: dean.willis@softarmor.com
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Dean wrote:
> Bill said:
> > We have now identified TWO cases where un-proxy-like behavior will be
> > done in real networks.  The first, a few months ago, was where a
> > firewall control proxy modified the Contact header, and possibly
> > also modified the addresses and ports in the SDP.  The second is here,
> > where a privacy proxy modifies the From header (and quite probably,
> others).
> 
> Proxies in any form cannot modify bodies, and B2BUAs are likely to break
> things if they try to. SIP security relies greatly on smime-protected
> bodies.

The philosophy is nice, but not all that useful if it can't be put
into practice.  As was said previously on this subject:

Last February Jonathan Rosenberg said:
>Now, it is true that, ahem, certain prominent sip vendors have products
>called firewall control proxies which muck with the SDP. Whether these
>things are rightly proxies, based on the SIP spec, or a B2BUA, is more a
>matter of technical marketing than standards, so I would not lose too
>much sleep over it. Its far less a stretch than what people are calling
>softswitches these days ;). Our job here at IETF is to write standards
>that work and are good for the Internet.

And Rohan responded:
> I agree with Jonathan here.  We should leave the spec as is with the
> exception of *adding* Contact-Length.  Another prominent vendor that I
> know has an ALG that modifies the SDP, and I don't even want to
> *think* about what you would call it.  Both of these will break as soon as
> someone turns on the end-to-end SMIME integrity and encryption.  I would
> like to have that problem some day...

I still believe we should recognize that such things will exist, and that
services will need to cope with them, and that guidance to designers of
new services would benefit from knowing, in advance, what such elements
might do to cause them trouble.

Bill Marshall
wtm@research.att.com

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From daemon@ns.ietf.org  Fri Apr 19 16:47:14 2002
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Thread-Topic: [Sip] new drafts relating to privacy and identity concepts posted
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Hi Dean,

I have the following questions:

	1. What is the purpose of this identity draft?
	   If it is for network usage, then identity info shall not be delivered to the UAS 
	   when the Privacy is set to Anonymous.
	2. Why is using MIME type instead of new optional header field?
	   Why Call-ID and Date are required?
	   Why is only encrypted for Anonymous calls? How about non-Anonymous calls?
	3. Can we consider forwarding scenario? Such that the forwarding Identity is included.
	   I know this does not apply to IP world when 3xx is returned to the UAC. 
	   But it is very popular in the PSTN interwork scenarios.

	4. In page 14, 5.4 example of the Content-Length:147 of the Unique-Boundary-1.
	   Do you think the length shall include the length of Unique-boundary42.
	5. There are several IDs that we need to consider all together instead of 
	   separating them: 
		1) User, who actually makes the call 
		2) actual Subscriber, who subscribes the service 
	  	3) identity known by the Service Provider, who provides the service
		4) The last forwarding-by identity known by the Service Provider, 
		   who provides the service
		5) Billing Identity, who actually pays the bill
	   Not all of above are needed by the UAS, but all are needed by the Network.
	6. Every identity shall have its own privacy attribute, 
	   so we should address Identity and Privacy together.
Regards,

Mark Chiou

-----Original Message-----
From: Dean Willis [mailto:dwillis@dynamicsoft.com]
Sent: Wednesday, April 17, 2002 6:24 PM
To: sip@ietf.org
Subject: [Sip] new drafts relating to privacy and identity concepts
posted


Please review:

http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-privacy-longter
m-00.txt

http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-identity-00.txt

They've been sent into the internet-drafts repository and should appear
there shortly.

--
Dean


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Fri Apr 19 19:06:40 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: RE: [Sip] Questions on  new draft identity concepts posted
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I'd like to take a stab at these questions. Some responses below.

Jon Peterson 
NeuStar, Inc.

> -----Original Message-----
> From: Chiou, Mark [mailto:MChiou@santera.com]
> Sent: Friday, April 19, 2002 1:21 PM
> To: Dean Willis
> Cc: sip@ietf.org
> Subject: [Sip] Questions on new draft identity concepts posted
> 
> 
> Hi Dean,
> 
> I have the following questions:
> 
> 	1. What is the purpose of this identity draft?
> 	   If it is for network usage, then identity info shall 
>  not be delivered to the UAS when the Privacy is set to Anonymous.

The purpose of this identity draft is to define a general-purpose way for an
'authentication service' to provide a secure, verified identity for the
originator of a request that can be inspected by any potential recipients.
Privacy concerns are addressed in the draft - the identity will not be
decipherable by a UAS when privacy is requested.

> 	2. Why is using MIME type instead of new optional header field?

This is a good question, and unfortunately a deep one. That short story is
that it is easier to stage the required security properties (integrity via
signature with certificate, replay protection, optional confidentiality) in
a body than in a header. With a body, we can also do this through baseline
bis09 security mechanisms without having to invent anything new.

> 	   Why Call-ID and Date are required?

To provide reference integrity to the request, and a small amount of replay
protection, respectively.

> 	   Why is only encrypted for Anonymous calls? How about 
> non-Anonymous calls?

If privacy is not required, then presumably the identity can and should be
shared with any potential recipient. Seeing a network-verified identity
might make a recipient more comfortable that they know the identity of the
originator of a request. Users should only request privacy when they need it
- I often don't accept phone calls that are from a blocked ID, myself.

> 	3. Can we consider forwarding scenario? Such that the 
> forwarding Identity is included.
> 	   I know this does not apply to IP world when 3xx is 
> returned to the UAC. 
> 	   But it is very popular in the PSTN interwork scenarios.

I have some hopes that the 'auth-id' system described in this draft (or
whatever it eventually ends up being when the draft is done) will be
applicable to things like forwarding and referral. I think it may already
help out with some identity problems in forwarding.

> 
> 	4. In page 14, 5.4 example of the Content-Length:147 of 
> the Unique-Boundary-1.
> 	   Do you think the length shall include the length of 
> Unique-boundary42.

Those MIME types do get confusing  when they're embedded, eh? The
Content-Length in that example does not pretend to be accurate - maybe I'll
update it to a less inaccurate value. However, the unique-boundaries, are, I
hope, correctly aligned.

> 	5. There are several IDs that we need to consider all 
> together instead of 
> 	   separating them: 
> 		1) User, who actually makes the call 
> 		2) actual Subscriber, who subscribes the service 
> 	  	3) identity known by the Service Provider, who 
> provides the service
> 		4) The last forwarding-by identity known by the 
> Service Provider, 
> 		   who provides the service
> 		5) Billing Identity, who actually pays the bill
> 	   Not all of above are needed by the UAS, but all are 
> needed by the Network.

All interesting concepts, and as you'll note I left open the possibility
that other information could be put into the 'auth-id' token as necessary.
For the time being, the draft is only concerned, really, with (1) and (3)
above. I'm surprised you make a distinction between (2) and (5) - I though
the fact that (2) encompasses (5) is what differentiates (2) from (1). Then
again, I don't really understand (2) that well either. Anyway, I don't think
the identity draft needs to get into any of these details - this mechanism
could be extended at a later time to accomodate these concepts as necessary.
SIP needs to grapple a bit with what these other sorts of identities are
before we figure out how to stuff them into secured bodies.

> 	6. Every identity shall have its own privacy attribute, 
> 	   so we should address Identity and Privacy together.

This identity draft does depend on and extend the accompanying privacy
draft. Frankly, I think ascribing a new privacy attribute to every
conceivable form of identity is unnecessary, and possibly dangerous. A user
shouldn't have to know, when they create a message, all of the various
headers in a message for which they need privacy - merely what kind of
privacy they need. While there is definitely a relationship between identity
and privacy, I think they are separable and they benefit from separate
consideration.

> Regards,
> 
> Mark Chiou
> 
> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Wednesday, April 17, 2002 6:24 PM
> To: sip@ietf.org
> Subject: [Sip] new drafts relating to privacy and identity concepts
> posted
> 
> 
> Please review:
> 
> http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-priva
cy-longter
m-00.txt

http://www.softarmor.com/sipwg/drafts/draft-peterson-sip-identity-00.txt

They've been sent into the internet-drafts repository and should appear
there shortly.

--
Dean


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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@ns.ietf.org  Fri Apr 19 19:42:58 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Cc: "3GPP_TSG_CN_WG1" <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Date: Fri, 19 Apr 2002 18:15:16 -0500
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I've submitted draft-willis-sip-svcrtdisco-01.txt to the Internet Drafts
queue. This version includs refinements made by Bernie Hoeneisen to my
earlier draft.

The draft should be announced shortly on ietf-announce.

In the meantime, please review:

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.txt

or

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.htm
l

or

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.xml


This is a "P-header" draft addressing 3GPP's required header for
reporting a service proxy assignment in the REGISTER response.


Please review ASAP. We need to get this to IETF last call by the end of
the month if at all possible.

--
Dean


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Date: Fri, 19 Apr 2002 16:28:55 -0700
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] Proposed plan for Network-Asserted ID/Privacy
To: Flemming Andreasen <fandreas@cisco.com>
Cc: Rohan Mahy <rohan@cisco.com>, sip@ietf.org
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---- Original message ----
>Date: Fri, 19 Apr 2002 09:52:42 -0400
>From: Flemming Andreasen <fandreas@cisco.com>  
>Subject: Re: [Sip] Proposed plan for Network-Asserted 
ID/Privacy  
>To: Rohan Mahy <rohan@cisco.com>
>Cc: sip@ietf.org
>
>
>
>Rohan Mahy wrote:
>
>> Hi,
>>
>> I'd like to propose a concrete plan to move forward with a 
solution for
>> network-asserted identity and privacy.  I'm proposing this 
as an
>> individual (who is sick of reading about this topic on the 
mailing
>> list).  Below I've listed some of problems that various 
folks were
>> apparently trying to solve in this space.
>>
>> 1.  Provide a Network-Asserted ID
>>  a.   Within a trusted administrative domain or federation 
of domains
>>  b.   For interworking with PSTN mechanisms*
>>  c.   Usable across administrative boundaries
>>
>> 2.  Provide capability for Call Trace
>>  a.   Within a trusted administrative domain or federation 
of domains
>>  b.   For interworking with PSTN mechanisms*
>>  c.   Usable across administrative boundaries
>>
>> 3.  Provide privacy
>>  a.   by the user directly withholding information from 
the network
>>  b.   in the network for user-provided info at the user's 
request
>>  c.   in the network for user-provided info based on 
network policy
>>  d.   of network-provided information
>>
>> 4.  Allow for a user-provided "hint" for Network-Asserted 
ID
>>
>> *(with gateways in the trusted domain or federation)
>>
>> Most of the snags and disagreements we have had with the 
existing
>> Remote-Party-ID draft (draft-ietf-sip-privacy-04) are 
related to problems
>> 1c, 2c, and 4.  Even a very scaled-down version of this 
draft would allow
>> for trusted intermediaries and user agents (including 
gateways) to
>> exchange network asserted identity and provide the 
capability for call
>> trace within a domain or federation of trusted domains.  
The solution is
>> as simple as a single header which carries the identity in 
clear text,
>> which is added by the first proxy that authenticates the 
user, and
>> removed before it leaves the trust boundary.
>
>You need to be more clear. 1 and 2 talk about administrative 
domains and
>administrative boundaries and now you are talking about 
trust boundaries. How
>do you define an administrative domain, an administrative 
boundary, and a
>trust boundary ?

Fair enough.  Let's say for now that a trusted entity is 
trusted not to maliciously modify messages, or 
reveal "private" network asserted ID information outside 
the "trusted" federation.  So, ordinary UAs in "my" service 
provider network are not trusted.

>> This would solve
>> 1a, 1b, 2a, and 2b.
>
>No. The UA is (normally) untrusted but still needs it. If 
you were only
>talking about the case when privacy was requested, then it 
doesn't address the
>requirement to provide call trace for anonymous calls. The 
user is obviously
>untrusted and hence cannot be given the caller-id in clear-
text. The privacy
>draft solves this by handing out an encrypted Remote-Party-
Id which can
>subsequently be used for call trace (as well as other 
features for that
>matter, e.g. anonymous call return).

OK, I hope we agree that 1a, 1b, and 2b would by solved by 
the scaled down mechanism.  You could still provide call 
trace (2a) within this network by using accounting 
mechanisms.  You will probably say that insuring you have an 
accounting record for each call is a burden.  Also, you do 
lose the ability to do call return with this mechanism, but I 
do not consider this a short-term requirement.  (You might 
disagree).

><snip>
>
>>
>> Finally, we have problem 4.  Some folks have proposed that
>> UAs should provide a hint using the same mechanism used to
>> communicate network asserted identity among trusted 
entities.
>
>> Others propose that authentication servers that create a
>
>> network asserted identity accept multiple "usernames" 
during
>
>> authentication and use this as the hint.  AFAIK these are 
the
>> only two choices, and we can come up with an answer to this
>> question in a separate thread.  The answer does not 
fundamentally
>> change any of the mechanisms we select to solve the other 
problems.
>
>Well it certainly impacts it. If you go with the first 
solution then you also
>accept that Remote-Party-Id can be provided by an untrusted 
entity, so I don't
>think we can solve these independently.

i think we need to decouple them now, and can join them again 
if it makes sense.

>> So, in summary, I'm proposing:
>>
>> No additional work is required to implement 3c
>>
>> Three short term deliverables
>> I) adopt draft-peterson-sip-privacy-longterm as a WG item 
to solve
>>    3a and 3b
>>
>
>It's unclear that 3b is independent of which mechanism you 
choose to provide
>the hint above.

I don't follow you.

>> II) write an extension to I) as a WG item for requesting 
privacy of
>> network asserted ID (however it is expressed) to support 
3d.  This would
>> get used by III and IV.
>>
>> III) pare down the mechanism in draft-ietf-sip-privacy to 
only address 1a,
>> 1b, 2a, and 2b.  quickly get closure on which mechanism we 
use for 4, and
>> include it in the draft.  keep the current applicability 
statement.
>
>As mentioned above, I'm not clear on exactly what 1a, 1b, 2a 
and 2b are, but
>if the "network asserted identity" draft would neither 
address privacy for
>network asserted identity, nor call trace for anonymous 
calls, then I
>certainly disagree with this.

I think it does provide a solution here for a small scope.

thanks,
-rohan

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From daemon@ns.ietf.org  Fri Apr 19 19:52:15 2002
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Date: Fri, 19 Apr 2002 16:32:14 -0700
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
To: jh@lohi.eng.song.fi
Cc: Mark Watson <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>,
        Rohan Mahy <rohan@cisco.com>, sip@ietf.org
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Juha,

>Mark Watson writes:
> > I diagree. Solve the other problems first. Then we have 
three choices:
> > 1) Use the same header for the 'hints'
> > 2) Define another header for the 'hints'
> > 3) Assume the 'hint' in in the authentication 'username'
>
>(3) doesn't work.  my username is jh@song.fi, but my from 
uri that i
>want to use, can be, for example, tel+358441234567.

I'm suggesting that in order to solve this problem, one 
choice (3) is that we can make these proxies smart enough 
that they can accept jh@song.fi or +358441234567 in the 
username field, and understand that these two things refer to 
the same user.

thanks,
-rohan

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Date: Fri, 19 Apr 2002 16:48:37 -0700
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sip] Proposed plan for Network-Asserted ID/Privacy
To: William Marshall <wtm@research.att.com>
Cc: rohan@cisco.com, sip@ietf.org
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---- Original message ----
>Date: Fri, 19 Apr 2002 15:00:28 -0400 (EDT)
>From: William Marshall <wtm@research.att.com>  
>Subject: Re: [Sip] Proposed plan for Network-Asserted 
ID/Privacy  
>To: rohan@cisco.com
>Cc: sip@ietf.org
>
>Rohan Mahy wrote:
>> I'd like to propose a concrete plan to move forward with a 
solution for
>> network-asserted identity and privacy.  I'm proposing this 
as an
>> individual (who is sick of reading about this topic on the 
mailing
>> list).  Below I've listed some of problems that various 
folks were
>> apparently trying to solve in this space.
>>
>> 1.  Provide a Network-Asserted ID
>>  a.   Within a trusted administrative domain or federation 
of domains
>>  b.   For interworking with PSTN mechanisms*
>>  c.   Usable across administrative boundaries
>>
>
>Once a proxy has determined the value of this "Network-
Asserted ID",
>there are actually four possibilities to consider.
>   a.   Sending it to another proxy within a trusted admin 
domain 
>		or federation of domains
>   b.   Sending it to a PSTN gateway within a trusted admin 
domain
>		or federation of domains
>   c.   Sending it to a proxy outside the trust admin domain
>		or federation of domains
>   d.   Sending it to a UA

OK, I agree that this list is clearer than the one I proposed.

>Unfortunalely caller-id and call-trace are useless without 
choice (d).
>So my first comment is that 1d and 2d are needed in your 
itemized list 
>of problems to solve.

Lets assume that the Network Asserted ID is stripped as it 
leaves a trust boundary only if privacy is requested (so you 
get 1d).  Then the only problem to solve is anonymous call 
trace (2d).  I think that some providers can choose to offer 
call trace by storing accounting records for some period of 
time.  Alternatively, the provider could require a signature-
based mechanism like the one that Jon proposed.

>I think it is a major simplification to reduce these four 
cases to only
>two:
>   a.  Sending it to a network element within a trusted 
admin domain
>		or federation of domains
>   b.  Sending it to a network element outside the trusted 
admin domain
>		or federation of domains
>(and who says PSTN gateways are always trusted? Should the 
GW generate
>a private used-once phone number for the call attempt?)

To clarify, I intended for gateways which are not trusted to 
be treated like any other device in a different domain.  An 
untrusted PSTN gateway would need to be addressed using a 
signature-based mechanism.  Consequently, it would not be 
able to provide an actual phone number and just set it to 
screened.

>And, as much as I'd like to believe your comment that no 
additional work
>is required to implement 3c, are we really willing to leave 
it that all
>services will break if anonymity is requested?  Or should 
>services be designed with this in mind?

I don't think that using an anonymizer has to break features 
unless these feature explicitly need raw identity 
information.  Also, I'm not really sure what you are 
proposing instead.

thanks,
-rohan

>Bill Marshall
>wtm@research.att.com
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on 
current sip
>Use sipping@ietf.org for new developments on the application 
of sip

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From daemon@ns.ietf.org  Fri Apr 19 20:50:16 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Rohan Mahy'" <rohan@cisco.com>, sip@ietf.org
Subject: RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
Date: Fri, 19 Apr 2002 19:21:06 -0500
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I wanted to weigh in with a couple of comments on this plan, below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Thursday, April 18, 2002 4:07 PM
> To: sip@ietf.org
> Subject: [Sip] Proposed plan for Network-Asserted ID/Privacy
> 
[snip]
> 
> 1.  Provide a Network-Asserted ID
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries
> 

I assume we understand 'across administrative boundaries' here to refer to
any 'untrusted' destination?

I really have a hard time seeing why 1b (and 2b by extension) needs to be a
separate case from 1a (and 2a). Surely PSTN gateways are 'trusted', and no
element is going to do anything differently when sending a request to a PSTN
gateway than they would do when sending it to any other 'trusted' element?

I also haven't seen any great need in my own identity draft to make use of
the distinction along the lines of 1a and 1c ('trusted' and 'untrusted') - I
feel that this distinction is motivated architectures that arise from the
absence of cryptographic mechanisms. But I understand that there are some
who want to carve the problem up this way. Personally, I view 1a, 1b and 1c
as requirements that are differentiated only by pseudoproblems.

> 2.  Provide capability for Call Trace
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries

The distinction between 1 and 2 overall is a little fine to me as well.
Surely network-asserted identity is a way of implementing call trace, right?
Call trace implies revealing identity selectively, namely the privacy
requirements entailed in 3, but I'm not sure we need a separate category for
this. Is there something other than what we are already doing to implement 1
and 3 that we'd need to implement 2?

3 & 4 below are both fine.

> 
> 3.  Provide privacy
>  a.   by the user directly withholding information from the network
>  b.   in the network for user-provided info at the user's request
>  c.   in the network for user-provided info based on network policy
>  d.   of network-provided information
> 
> 4.  Allow for a user-provided "hint" for Network-Asserted ID
> 
> *(with gateways in the trusted domain or federation)
> 
[snip]
> 
> So, in summary, I'm proposing:
> 
> No additional work is required to implement 3c

Agreed. Nothing pressing, anyway.

> 
> Three short term deliverables
> I) adopt draft-peterson-sip-privacy-longterm as a WG item to solve
>    3a and 3b

Fine by me, obviously.

> 
> II) write an extension to I) as a WG item for requesting privacy of
> network asserted ID (however it is expressed) to support 3d.  This would
> get used by III and IV.
> 

So, today my identity draft already contains a privacy extension for the
style of NAI provided in that draft. It declares a Privacy header priv-value
for 'auth' for this purpose. That could be generalized, certainly, though
exactly what it would entail would undoubtedly be different for different
NAI systems. In my identity draft, it entails that one SHOULD not distribute
an NAI, and if one does, it MUST be encrypted. Personally, I don't think
that relying on downstream entities to remove the NAI when they see fit is
sufficient, and this would probably be a point of contention in any attempt
to agree on a single meaning for this 'auth' priv-value.

I'd therefore like to suggest that we think a little more about this. Maybe
there's a way to generalize the syntax but not the semantics for any
particular NAI system. However, if I were the user requesting privacy, I'd
want to know whether encryption or removal were going to be used...

> III) pare down the mechanism in draft-ietf-sip-privacy to only address 1a,
> 1b, 2a, and 2b.  quickly get closure on which mechanism we use for 4, and
> include it in the draft.  keep the current applicability statement.
> 

I don't think the mechanism you need to use for 4 should be included in this
draft - I think the manner in which you provide your identity to entities
that create NAIs has no obvious dependencies on the NAI distribution
mechanism. In so far as this may be a matter for SIP (the people that seem
to want this may have some out-of-band way of getting this data), it should
probably be in its own draft.

I'd also like to comment that I think getting the existing sip-privacy-04
mechanism into parity with a reasonable applicability, including as Flemming
sugggests robust definitions of trust, administrative boundaries, and so
forth is not necessarily a trivial task.

> One or more longer term deliverables, including
> IV) draft-peterson-sip-identity ... to address a more general way to
> solve 1 and 2 including 1c and 2c.  Note that this draft is the first
> step toward providing real cryptographic identity services.  
> I think we also need to address this topic as a charter item.
> 

Suits me. Although the mechanism my draft proposes is admittedly
heavyweight, I think it addresses the general requirements with few new
protocol mechanisms. I'm looking forward to comments on it. I have no reason
today to think that it will require a 'long term' to complete this work.

> We might also provide an informational document describing how to map
> privacy and identity between SIP and the PSTN.
> 

Juha, among others, had argued this in the past, anyway. I think we do need
a way to map SIP's privacy apparati to the appropriate ISUP presentation
restriction bits, and so on. This is just part of the SIP-T work.

> hope this helps.
> thanks,
> -rohan
> 
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Sat Apr 20 01:34:30 2002
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Date: Sat, 20 Apr 2002 07:52:14 +0300
To: Rohan Mahy <rohan@cisco.com>
Cc: Mark Watson <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>, sip@ietf.org
Subject: Re: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Rohan Mahy writes:

 > I'm suggesting that in order to solve this problem, one 
 > choice (3) is that we can make these proxies smart enough 
 > that they can accept jh@song.fi or +358441234567 in the 
 > username field, and understand that these two things refer to 
 > the same user.

the point that i wanted to make is that my "username" is totally
separate from my "from uri" and that BOTH exist in the invite that i
send.  when my sip/pstn gw forwards the invite to pstn, the network must
verify that the "from uri" really belongs to me as identified by the
username.

-- juha


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: HTTP Digest Authentication Using AKA
	Author(s)	: R. Housley, W. Polk
	Filename	: draft-ietf-sip-digest-aka-00.txt
	Pages		: 17
	Date		: 17-Apr-02
	
The Hypertext Transfer Protocol (HTTP) Authentication Framework
includes two authentication schemes: Basic and Digest.  Both schemes
employ a shared secret based mechanism for access authentication.
The Authentication and Key Agreement (AKA) mechanism performs user
authentication and session key distribution in Universal Mobile
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specifies an AKA based one-time password generation mechanism for
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From daemon@ns.ietf.org  Sat Apr 20 14:13:55 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <jh@lohi.eng.song.fi>, "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>, <sip@ietf.org>
Subject: RE: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
Date: Sat, 20 Apr 2002 11:48:25 -0500
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> the point that i wanted to make is that my "username" is 
> totally separate from my "from uri" and that BOTH exist in 
> the invite that i send.  when my sip/pstn gw forwards the 
> invite to pstn, the network must verify that the "from uri" 
> really belongs to me as identified by the username.

That's sort of a quirk in your gateway -- it maps From: to
calling-party-ID. Most gateways work that way today. They won't
necessarily do so in the future.

It COULD map "Username" from an authorization header to
calling-party-ID. It also COULD generate the challenge to get an
appropriate authorization header back. That may not be the "right" way
to do it, but that's what we're talking about here -- finding the
"right" way to go forward.

--
Dean


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From daemon@ns.ietf.org  Sat Apr 20 17:58:53 2002
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To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>, <sip@ietf.org>
Subject: RE: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Dean Willis writes:

 > It COULD map "Username" from an authorization header to
 > calling-party-ID. 

no it could not because the user may have any number of calling party
numbers belong to different area codes and even different countries.
the user thus definitely needs to be able to control the calling party
id.

 > It also COULD generate the challenge to get an
 > appropriate authorization header back. 

this is what we do today, but knowing who the user is doesn't mean that
the proxy knows which calling party id the user wants to use.

 > That may not be the "right" way
 > to do it, but that's what we're talking about here -- finding the
 > "right" way to go forward.

the proxy MUST respect the user's will regarding the calling party id
within the limits of calling party ids of that user.  the proxy has no
right to pick one on its own.

-- juha




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From daemon@optimus.ietf.org  Sun Apr 21 06:54:51 2002
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Date: Sun, 21 Apr 2002 13:07:42 +0300
From: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
Organization: OY LM Ericsson AB
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Subject: [Sip] Comment about draft-ietf-sip-dhcpv6-00.txt
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Hi:

A very simple comment regarding the DHCPv6 option for SIP servers I-D.

The draft has a duplicated reference to the "SIP: locating SIP servers" I-D, future RFC 3263. It is listed as both references 3 and 7.

Regards,

    Miguel
-- 
Miguel-Angel Garcia                     Oy LM Ericsson AB
                                        Jorvas, Finland
mailto:Miguel.A.Garcia@ericsson.com     Phone:  +358 9 299 3553
mailto:Miguel.A.Garcia@piuha.net        Mobile: +358 40 5140002

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From: "Ofir Arkin" <ofir@sys-security.com>
To: <sip@ietf.org>
Date: Sun, 21 Apr 2002 14:11:21 +0100
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Subject: [Sip] When is a 487 Request Terminated is sent?
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I would appreciate if some one would shade a light on the exact
circumstances in which a 487 Request Terminated is sent.


Thanks
Ofir Arkin [ofir@sys-security.com]
The Sys-Security Group
http://www.sys-security.com
PGP CC2C BE53 12C6 C9F2 87B1 B8C6 0DFA CF2D D360 43FA  




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: A Privacy Mechanism for the Session Initiation 
                          Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sip-privacy-longterm-00.txt
	Pages		: 26
	Date		: 18-Apr-02
	
This document defines new mechanisms for the Session Initiation
Protocol (SIP) in support of privacy.  Specifically, guidelines are
provided for the creation of messages that do not divulge personal
identity information.  A new 'privacy service' logical role for
intermediaries is defined to answer some privacy requirements that
user agents cannot satisfy themselves.  Finally, means are presented
by which a user can request particular functions from a privacy
service.

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From daemon@optimus.ietf.org  Sun Apr 21 10:13:24 2002
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Thanks; corrected and to be fixed in the next release.

"Miguel A. Garcia" wrote:
> 
> Hi:
> 
> A very simple comment regarding the DHCPv6 option for SIP servers I-D.
> 
> The draft has a duplicated reference to the "SIP: locating SIP servers" I-D, future RFC 3263. It is listed as both references 3 and 7.
> 
> Regards,
>

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From daemon@optimus.ietf.org  Sun Apr 21 14:51:34 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <jh@lohi.eng.song.fi>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>, <sip@ietf.org>
Subject: RE: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
Date: Sun, 21 Apr 2002 12:42:47 -0500
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> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi] 
> Sent: Saturday, April 20, 2002 4:21 PM
> To: Dean Willis
> Cc: 'Rohan Mahy'; 'Mark Watson'; 'Flemming Andreasen'; sip@ietf.org
> Subject: RE: Problem 4 of RE: [Sip] Proposed plan for 
> Network-Asserted ID/Privacy
> 
> 
> Dean Willis writes:
> 
>  > It COULD map "Username" from an authorization header to
>  > calling-party-ID. 
> 
> no it could not because the user may have any number of 
> calling party numbers belong to different area codes and even 
> different countries. the user thus definitely needs to be 
> able to control the calling party id.

Well, the user COULD have many different usernames to present to the
gateway, one for each public ID. They might not use the same username
for a calling-party-ID service (that is what we're talking about) as
they use for routing service on a proxy.

It's a different way to look at the problem, and I offer it only as an
existence proof for alternatives.

--
Dean


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From daemon@optimus.ietf.org  Sun Apr 21 14:53:07 2002
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To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>, <sip@ietf.org>
Subject: RE: Problem 4 of RE: [Sip] Proposed plan for Network-Asserted ID/Privacy
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Dean Willis writes:

 > Well, the user COULD have many different usernames to present to the
 > gateway, one for each public ID. They might not use the same username
 > for a calling-party-ID service (that is what we're talking about) as
 > they use for routing service on a proxy.

fine. a user may or may not have many calling party ids associated with
th same authentication username.  so the requirement is to support both
cases.

 > It's a different way to look at the problem, and I offer it only as an
 > existence proof for alternatives.

yes, there is at least the above two alternatives and the scheme
whatever it is must support both is them.

also, i would like to add one more requirement, which is important for
backwards compatibility with thousands of currently deployed UAs, which
assume that (if the calling party is not anonymous) the calling party's
call can be returned by using its from uri.

-- juha


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From daemon@optimus.ietf.org  Mon Apr 22 02:16:19 2002
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Date: Sun, 21 Apr 2002 22:30:58 -0700
From: Bobby Sardana <sardana@obsoft.com>
Organization: ObjectSoftware, Inc
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To: Ofir Arkin <ofir@sys-security.com>
CC: sip@ietf.org
Subject: Re: [Sip] When is a 487 Request Terminated is sent?
References: <000e01c1e936$0b93dcb0$0a01a8c0@joshua>
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Consider the following scenario: (A) is inviting (B):

1. (A)--------------------INVITE --------------------->(B)
2. (A)<------------------180 RIGING------------------(B)
3. (A)--------------------CANCEL------------------->(B)
4. (A)<------------------OK----------------------------(B)
5. (A)<---------487 Request Terminated-----------(B)
6. (A)--------------------ACK------------------------->(B)

OK (4) reply corresponds to the CANCEL (3) request.
487 Request Terminated (5) reply corresponds to the INVITE (1) request.

Regards,

Bobby Sardana.
sardana@obsoft.com

Ofir Arkin wrote:

> I would appreciate if some one would shade a light on the exact
> circumstances in which a 487 Request Terminated is sent.
>
> Thanks
> Ofir Arkin [ofir@sys-security.com]
> The Sys-Security Group
> http://www.sys-security.com
> PGP CC2C BE53 12C6 C9F2 87B1 B8C6 0DFA CF2D D360 43FA
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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To: Adam Roach <adam@dynamicsoft.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
Message-ID: <1019462987.3cc3c54bad458@imp.free.fr>
Date: Mon, 22 Apr 2002 10:09:47 +0200 (MEST)
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>,
        Robert Sparks <rsparks@dynamicsoft.com>, sip@ietf.org
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Adam,

I strongly support Tom position.

Referred-by is useful for Intranet/Extranet cases.

Why do you want to remove (insistenly ;-)something which is already in the draft 
and surely often used ? Could you explain the Problem ?

I would propose to let This Referred-by in the draft and to provide an 
authenticated version later.

Best Regard,

Patrick



Quoting Adam Roach <adam@dynamicsoft.com>:

> Why is such information useful if it's unauthenticated?
> 
> Are you planning to use it to implement policy? If so,
> inclusion of such a header only sets a trap for naive
> implementors.
> 
> If not, are you envisioning this information will be presented
> to users? If that's your goal, you'll want to do something
> that works with deployed phones. As a convention (and I'm not
> proposing that this be standardized), you could obtain the
> desired effect with something like:
> 
> INVITE sip:bob@example.com SIP/2.0
> From: "Adam Roach forwarded by Robert Sparks"
> <sip:adam@dynamicsoft.com>;tag=12397
> ...
> 
> Right?
> 
> I feel quite strongly that we should *not* include this sort of
> information
> in the current REFER draft, and work on providing an *authenticated*
> version of this draft at a later date.
> 
> /a
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > Sent: Wednesday, April 10, 2002 1:51
> > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com;
> sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > I agree with what Patrick Mourot says. Let's have a basic 
> > refer capability
> > that provides an unsecured identity of A to C, and then leave 
> > it to C to
> > decide whether and how to verify the identity of A. This will 
> > depend on
> > whether it is attended or unattended transfer - with attended 
> > transfer the
> > call A-C will exist and will be identified by the Replaces 
> > header. This may
> > be sufficient, but if not, verification can be done in the 
> > context of that
> > dialog A-C.
> > 
> >  ---------------------------------------------------------------
> >  John Elwell (john.elwell@siemenscomms.co.uk)
> >  Siemens Communications Limited,
> >  ---------------------------------------------------------------
> >  Internet communications are not secure and therefore Siemens
> >  Communications Limited does not accept legal responsibility for the
> >  contents of this message. Any views or opinions presented are
> solely
> >  those of the author and do not necessarily represent those of
> Siemens
> >  Communications Limited unless otherwise specifically stated.
> >  
> > 
> > 
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > Sent: Tuesday, April 09, 2002 5:05 PM
> > To: rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > 
> > Sirs,
> > 
> > I think there are plainty of cases where C :
> > -will not need to know that the call was started by A read in 
> > Referred-By
> > -or will trust B (i.e: intranet cases).
> > 
> > Protecting Referred-By also means protecting others headers.
> > 
> > This means that we should avoid all solutions that are not optional
> > regarding C 
> > policy.
> > 
> > A VERIFYing approach starting at C is better suited; It 
> > allows C to decide
> > to 
> > verify or not A identity.
> > If C does NOT want to or does NOT support VERIFY the call 
> > flows remains 
> > consistent with current behavior.
> > 
> > PS: Referred-By is a means for carrying some useful Information.
> > 
> > That's my 2 cents.
> > 
> > Best Regards,
> > 
> > Patrick
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: <Tom_Gray@Mitel.COM>, "Adam Roach" <adam@dynamicsoft.com>
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
        <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 08:48:19 -0500
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> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Tom_Gray@Mitel.COM
> Sent: Friday, April 19, 2002 1:50 PM
> To: Adam Roach
> Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
>
>  The information could be useful because C could authenticate the identity
> outside of the SIP protocol if that is felt necesary. In many cases such
> authentication is not required. There can be other means
> extra-protocol  of
> dealing with abusive callers.
>

I'm not sure I see the connection between the last two sentences? How would
one make use of an unauthenticated referred-by in dealing with abusive
callers? I would tend to expect that abusive callers would be very likely
spoof refered-by headers if it is allowed.

I can accept the idea of referred-by being useful _if_ the endpoint has
another channel of authentication. I am skeptical about any real use of it
without authentication.

Considering that Robert's original question was not, "is there a use for an
unauthenticated refered-by?" but was instead "Is there use for refer if it
is published without referred-by?" with the understanding that a solid
referred-by header would be the subject of future work. From that
perspective, I have to agree with Adam's belief that referred-by should be
removed from the current draft.


>
>
>
>
>
> Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
>
> Sent by:  sip-admin@ietf.org
>
>
> To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
>       "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>, Robert
>       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> cc:
>
> Subject:  RE: [Sip] REFER security options - removing Referred-By
>
>
> Why is such information useful if it's unauthenticated?
>
> Are you planning to use it to implement policy? If so,
> inclusion of such a header only sets a trap for naive
> implementors.
>
> If not, are you envisioning this information will be presented
> to users? If that's your goal, you'll want to do something
> that works with deployed phones. As a convention (and I'm not
> proposing that this be standardized), you could obtain the
> desired effect with something like:
>
> INVITE sip:bob@example.com SIP/2.0
> From: "Adam Roach forwarded by Robert Sparks"
> <sip:adam@dynamicsoft.com>;tag=12397
> ...
>
> Right?
>
> I feel quite strongly that we should *not* include this sort of
> information
> in the current REFER draft, and work on providing an *authenticated*
> version of this draft at a later date.
>
> /a
>
> > -----Original Message-----
> > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > Sent: Wednesday, April 10, 2002 1:51
> > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > I agree with what Patrick Mourot says. Let's have a basic
> > refer capability
> > that provides an unsecured identity of A to C, and then leave
> > it to C to
> > decide whether and how to verify the identity of A. This will
> > depend on
> > whether it is attended or unattended transfer - with attended
> > transfer the
> > call A-C will exist and will be identified by the Replaces
> > header. This may
> > be sufficient, but if not, verification can be done in the
> > context of that
> > dialog A-C.
> >
> >  ---------------------------------------------------------------
> >  John Elwell (john.elwell@siemenscomms.co.uk)
> >  Siemens Communications Limited,
> >  ---------------------------------------------------------------
> >  Internet communications are not secure and therefore Siemens
> >  Communications Limited does not accept legal responsibility for the
> >  contents of this message. Any views or opinions presented are solely
> >  those of the author and do not necessarily represent those of Siemens
> >  Communications Limited unless otherwise specifically stated.
> >
> >
> >
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > Sent: Tuesday, April 09, 2002 5:05 PM
> > To: rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> >
> > Sirs,
> >
> > I think there are plainty of cases where C :
> > -will not need to know that the call was started by A read in
> > Referred-By
> > -or will trust B (i.e: intranet cases).
> >
> > Protecting Referred-By also means protecting others headers.
> >
> > This means that we should avoid all solutions that are not optional
> > regarding C
> > policy.
> >
> > A VERIFYing approach starting at C is better suited; It
> > allows C to decide
> > to
> > verify or not A identity.
> > If C does NOT want to or does NOT support VERIFY the call
> > flows remains
> > consistent with current behavior.
> >
> > PS: Referred-By is a means for carrying some useful Information.
> >
> > That's my 2 cents.
> >
> > Best Regards,
> >
> > Patrick
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: sip@ietf.org
Subject: [Sip] Re: Proposed plan for Network-Asserted ID/Privacy
Date: Mon, 22 Apr 2002 15:00:27 +0200
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additional NAI application inline..
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Friday, April 19, 2002 1:07 AM
> 
> [..] Below I've listed some of problems that various folks were
> apparently trying to solve in this space.
> 
> 1.  Provide a Network-Asserted ID
>  a.   Within a trusted administrative domain or federation of domains
>  b.   For interworking with PSTN mechanisms*
>  c.   Usable across administrative boundaries

   d.   Usable for billing purposes; callee sends an invoice to network
        asserted caller, inbound proxies work as inline trusted third party to
        ensure non-repudiation.

> [..]

--
Wolfgang Beck
T-Systems Nova GmbH 




 


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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 11:16:51 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 10:20:56 -0400
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Y'know, sometimes I (personally, not as chair) think we take
the security argument just a little too far.

Sure, I can finagle the From, but Referred-By is the header
I want to use in my GUI.  I'll attach no more credence to it
then I get from FROM unless it's authenticated, but it's damn
useful to display to a user.

We have it implemented, we like it, and I'm keeping it.

Let's leave it in and define it, like FROM, as informational
only.

Or shall we remove FROM on the same basis?

Brian

> -----Original Message-----
> From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> Sent: Monday, April 22, 2002 9:48 AM
> To: Tom_Gray@mitel.com; Adam Roach
> Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> 
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Tom_Gray@Mitel.COM
> > Sent: Friday, April 19, 2002 1:50 PM
> > To: Adam Roach
> > Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> > sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> >
> >  The information could be useful because C could 
> authenticate the identity
> > outside of the SIP protocol if that is felt necesary. In 
> many cases such
> > authentication is not required. There can be other means
> > extra-protocol  of
> > dealing with abusive callers.
> >
> 
> I'm not sure I see the connection between the last two 
> sentences? How would
> one make use of an unauthenticated referred-by in dealing with abusive
> callers? I would tend to expect that abusive callers would be 
> very likely
> spoof refered-by headers if it is allowed.
> 
> I can accept the idea of referred-by being useful _if_ the 
> endpoint has
> another channel of authentication. I am skeptical about any 
> real use of it
> without authentication.
> 
> Considering that Robert's original question was not, "is 
> there a use for an
> unauthenticated refered-by?" but was instead "Is there use 
> for refer if it
> is published without referred-by?" with the understanding that a solid
> referred-by header would be the subject of future work. From that
> perspective, I have to agree with Adam's belief that 
> referred-by should be
> removed from the current draft.
> 
> 
> >
> >
> >
> >
> >
> > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
> >
> > Sent by:  sip-admin@ietf.org
> >
> >
> > To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
> >       "'patrick.mourot@online.fr'" 
> <patrick.mourot@online.fr>, Robert
> >       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> > cc:
> >
> > Subject:  RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > Why is such information useful if it's unauthenticated?
> >
> > Are you planning to use it to implement policy? If so,
> > inclusion of such a header only sets a trap for naive
> > implementors.
> >
> > If not, are you envisioning this information will be presented
> > to users? If that's your goal, you'll want to do something
> > that works with deployed phones. As a convention (and I'm not
> > proposing that this be standardized), you could obtain the
> > desired effect with something like:
> >
> > INVITE sip:bob@example.com SIP/2.0
> > From: "Adam Roach forwarded by Robert Sparks"
> > <sip:adam@dynamicsoft.com>;tag=12397
> > ...
> >
> > Right?
> >
> > I feel quite strongly that we should *not* include this sort of
> > information
> > in the current REFER draft, and work on providing an *authenticated*
> > version of this draft at a later date.
> >
> > /a
> >
> > > -----Original Message-----
> > > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > > Sent: Wednesday, April 10, 2002 1:51
> > > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; 
> sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > I agree with what Patrick Mourot says. Let's have a basic
> > > refer capability
> > > that provides an unsecured identity of A to C, and then leave
> > > it to C to
> > > decide whether and how to verify the identity of A. This will
> > > depend on
> > > whether it is attended or unattended transfer - with attended
> > > transfer the
> > > call A-C will exist and will be identified by the Replaces
> > > header. This may
> > > be sufficient, but if not, verification can be done in the
> > > context of that
> > > dialog A-C.
> > >
> > >  ---------------------------------------------------------------
> > >  John Elwell (john.elwell@siemenscomms.co.uk)
> > >  Siemens Communications Limited,
> > >  ---------------------------------------------------------------
> > >  Internet communications are not secure and therefore Siemens
> > >  Communications Limited does not accept legal 
> responsibility for the
> > >  contents of this message. Any views or opinions 
> presented are solely
> > >  those of the author and do not necessarily represent 
> those of Siemens
> > >  Communications Limited unless otherwise specifically stated.
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > Sent: Tuesday, April 09, 2002 5:05 PM
> > > To: rsparks@dynamicsoft.com; sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > >
> > > Sirs,
> > >
> > > I think there are plainty of cases where C :
> > > -will not need to know that the call was started by A read in
> > > Referred-By
> > > -or will trust B (i.e: intranet cases).
> > >
> > > Protecting Referred-By also means protecting others headers.
> > >
> > > This means that we should avoid all solutions that are 
> not optional
> > > regarding C
> > > policy.
> > >
> > > A VERIFYing approach starting at C is better suited; It
> > > allows C to decide
> > > to
> > > verify or not A identity.
> > > If C does NOT want to or does NOT support VERIFY the call
> > > flows remains
> > > consistent with current behavior.
> > >
> > > PS: Referred-By is a means for carrying some useful Information.
> > >
> > > That's my 2 cents.
> > >
> > > Best Regards,
> > >
> > > Patrick
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 11:28:56 2002
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From: Tom_Gray@Mitel.COM
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Subject: RE: [Sip] REFER security options - removing Referred-By
To: "Ben Campbell" <bcampbell@dynamicsoft.com>
Cc: <Tom_Gray@Mitel.COM>, "Adam Roach" <adam@dynamicsoft.com>,
        "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
        <sip@ietf.org>
Date: Mon, 22 Apr 2002 11:00:58 -0400
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In response to the question of when would authenticated referred-by's not
be required, a valid example would be in an enterprise setting where
extra-protocol means of dealing with abuse can easily be used.

Taking the original scenario that led to the discussion of the need for
authentication of Refered-By's (at least as I remember it).

a) Alice is a CEO who only accepts calls screened by her secretary Bob.

b) Carole is an employee who discovers she can call Alice directly  by
creating a fraudulent Refered-By

c) Carole calls Alice and annoys her with some petty grievances.

d) Alice calls Bob and tells him to not allow calls from Carole any more

e) Bob tells Alice that he did not put the call through and that Carole
must have did it herself with an illicit use of the telephone system

f) Alice does not, as in the originally proposed scenario, order the
immedate removal of the PBX. Instead she tells Bob to take care of the
matter
 .
g) Bob calls Carole's manager Dave and they inform her that further
incidents of this sort will lead to unpleasant consequences.

The scenario could be extended with modifications to public network
situations.





"Ben Campbell" <bcampbell@dynamicsoft.com> on 04/22/2002 09:48:19 AM

To:   <Tom_Gray@Mitel.COM>, "Adam Roach" <adam@dynamicsoft.com>
cc:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
      <patrick.mourot@online.fr>, "Robert Sparks"
      <rsparks@dynamicsoft.com>, <sip@ietf.org>

Subject:  RE: [Sip] REFER security options - removing Referred-By




> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Tom_Gray@Mitel.COM
> Sent: Friday, April 19, 2002 1:50 PM
> To: Adam Roach
> Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
>
>  The information could be useful because C could authenticate the
identity
> outside of the SIP protocol if that is felt necesary. In many cases such
> authentication is not required. There can be other means
> extra-protocol  of
> dealing with abusive callers.
>

I'm not sure I see the connection between the last two sentences? How would
one make use of an unauthenticated referred-by in dealing with abusive
callers? I would tend to expect that abusive callers would be very likely
spoof refered-by headers if it is allowed.

I can accept the idea of referred-by being useful _if_ the endpoint has
another channel of authentication. I am skeptical about any real use of it
without authentication.

Considering that Robert's original question was not, "is there a use for an
unauthenticated refered-by?" but was instead "Is there use for refer if it
is published without referred-by?" with the understanding that a solid
referred-by header would be the subject of future work. From that
perspective, I have to agree with Adam's belief that referred-by should be
removed from the current draft.


>
>
>
>
>
> Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
>
> Sent by:  sip-admin@ietf.org
>
>
> To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
>       "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>, Robert
>       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> cc:
>
> Subject:  RE: [Sip] REFER security options - removing Referred-By
>
>
> Why is such information useful if it's unauthenticated?
>
> Are you planning to use it to implement policy? If so,
> inclusion of such a header only sets a trap for naive
> implementors.
>
> If not, are you envisioning this information will be presented
> to users? If that's your goal, you'll want to do something
> that works with deployed phones. As a convention (and I'm not
> proposing that this be standardized), you could obtain the
> desired effect with something like:
>
> INVITE sip:bob@example.com SIP/2.0
> From: "Adam Roach forwarded by Robert Sparks"
> <sip:adam@dynamicsoft.com>;tag=12397
> ...
>
> Right?
>
> I feel quite strongly that we should *not* include this sort of
> information
> in the current REFER draft, and work on providing an *authenticated*
> version of this draft at a later date.
>
> /a
>
> > -----Original Message-----
> > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > Sent: Wednesday, April 10, 2002 1:51
> > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > I agree with what Patrick Mourot says. Let's have a basic
> > refer capability
> > that provides an unsecured identity of A to C, and then leave
> > it to C to
> > decide whether and how to verify the identity of A. This will
> > depend on
> > whether it is attended or unattended transfer - with attended
> > transfer the
> > call A-C will exist and will be identified by the Replaces
> > header. This may
> > be sufficient, but if not, verification can be done in the
> > context of that
> > dialog A-C.
> >
> >  ---------------------------------------------------------------
> >  John Elwell (john.elwell@siemenscomms.co.uk)
> >  Siemens Communications Limited,
> >  ---------------------------------------------------------------
> >  Internet communications are not secure and therefore Siemens
> >  Communications Limited does not accept legal responsibility for the
> >  contents of this message. Any views or opinions presented are solely
> >  those of the author and do not necessarily represent those of Siemens
> >  Communications Limited unless otherwise specifically stated.
> >
> >
> >
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > Sent: Tuesday, April 09, 2002 5:05 PM
> > To: rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> >
> > Sirs,
> >
> > I think there are plainty of cases where C :
> > -will not need to know that the call was started by A read in
> > Referred-By
> > -or will trust B (i.e: intranet cases).
> >
> > Protecting Referred-By also means protecting others headers.
> >
> > This means that we should avoid all solutions that are not optional
> > regarding C
> > policy.
> >
> > A VERIFYing approach starting at C is better suited; It
> > allows C to decide
> > to
> > verify or not A identity.
> > If C does NOT want to or does NOT support VERIFY the call
> > flows remains
> > consistent with current behavior.
> >
> > PS: Referred-By is a means for carrying some useful Information.
> >
> > That's my 2 cents.
> >
> > Best Regards,
> >
> > Patrick
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip





_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 12:07:30 2002
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From: Michael Thomas <mat@cisco.com>
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Date: Mon, 22 Apr 2002 08:31:24 -0700 (PDT)
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>,
        "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
In-Reply-To: <313680C9A886D511A06000204840E1CF030B51E4@whq-msgusr-02.pit.comms.marconi.com>
References: <313680C9A886D511A06000204840E1CF030B51E4@whq-msgusr-02.pit.comms.marconi.com>
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Rosen, Brian writes:
 > Y'know, sometimes I (personally, not as chair) think we take
 > the security argument just a little too far.
 > 
 > Sure, I can finagle the From, but Referred-By is the header
 > I want to use in my GUI.  I'll attach no more credence to it
 > then I get from FROM unless it's authenticated, but it's damn
 > useful to display to a user.

   So you're saying that it's OK for me to be able
   to just put Kofi Anann's admin's name in the
   Referred-by header, so that I can get through
   his screening so we can chat about my thoughts
   on San Francisco annexing Canada?

   Groovy.

		    Mike

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Sriram Parameswar"<sriramp@nortelnetworks.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Ben Campbell'"
	 <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 10:26:08 -0500
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I have already pointed out that its only a SHOULD in the REFER spec "If the
URL is a SIP URL, the Referred-By header in the REFER request should be
copied into the request sent to the referred-to resource."

Thus removal is not required if you dont like it dont send it. Otherwise I
simply echo Brian's sentiments.

Thanks,

Sriram

-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
Sent: Monday, April 22, 2002 9:21 AM
To: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach
Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By


Y'know, sometimes I (personally, not as chair) think we take
the security argument just a little too far.

Sure, I can finagle the From, but Referred-By is the header
I want to use in my GUI.  I'll attach no more credence to it
then I get from FROM unless it's authenticated, but it's damn
useful to display to a user.

We have it implemented, we like it, and I'm keeping it.

Let's leave it in and define it, like FROM, as informational
only.

Or shall we remove FROM on the same basis?

Brian

> -----Original Message-----
> From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> Sent: Monday, April 22, 2002 9:48 AM
> To: Tom_Gray@mitel.com; Adam Roach
> Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> 
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > Tom_Gray@Mitel.COM
> > Sent: Friday, April 19, 2002 1:50 PM
> > To: Adam Roach
> > Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> > sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> >
> >  The information could be useful because C could 
> authenticate the identity
> > outside of the SIP protocol if that is felt necesary. In 
> many cases such
> > authentication is not required. There can be other means
> > extra-protocol  of
> > dealing with abusive callers.
> >
> 
> I'm not sure I see the connection between the last two 
> sentences? How would
> one make use of an unauthenticated referred-by in dealing with abusive
> callers? I would tend to expect that abusive callers would be 
> very likely
> spoof refered-by headers if it is allowed.
> 
> I can accept the idea of referred-by being useful _if_ the 
> endpoint has
> another channel of authentication. I am skeptical about any 
> real use of it
> without authentication.
> 
> Considering that Robert's original question was not, "is 
> there a use for an
> unauthenticated refered-by?" but was instead "Is there use 
> for refer if it
> is published without referred-by?" with the understanding that a solid
> referred-by header would be the subject of future work. From that
> perspective, I have to agree with Adam's belief that 
> referred-by should be
> removed from the current draft.
> 
> 
> >
> >
> >
> >
> >
> > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
> >
> > Sent by:  sip-admin@ietf.org
> >
> >
> > To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
> >       "'patrick.mourot@online.fr'" 
> <patrick.mourot@online.fr>, Robert
> >       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> > cc:
> >
> > Subject:  RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > Why is such information useful if it's unauthenticated?
> >
> > Are you planning to use it to implement policy? If so,
> > inclusion of such a header only sets a trap for naive
> > implementors.
> >
> > If not, are you envisioning this information will be presented
> > to users? If that's your goal, you'll want to do something
> > that works with deployed phones. As a convention (and I'm not
> > proposing that this be standardized), you could obtain the
> > desired effect with something like:
> >
> > INVITE sip:bob@example.com SIP/2.0
> > From: "Adam Roach forwarded by Robert Sparks"
> > <sip:adam@dynamicsoft.com>;tag=12397
> > ...
> >
> > Right?
> >
> > I feel quite strongly that we should *not* include this sort of
> > information
> > in the current REFER draft, and work on providing an *authenticated*
> > version of this draft at a later date.
> >
> > /a
> >
> > > -----Original Message-----
> > > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > > Sent: Wednesday, April 10, 2002 1:51
> > > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; 
> sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > I agree with what Patrick Mourot says. Let's have a basic
> > > refer capability
> > > that provides an unsecured identity of A to C, and then leave
> > > it to C to
> > > decide whether and how to verify the identity of A. This will
> > > depend on
> > > whether it is attended or unattended transfer - with attended
> > > transfer the
> > > call A-C will exist and will be identified by the Replaces
> > > header. This may
> > > be sufficient, but if not, verification can be done in the
> > > context of that
> > > dialog A-C.
> > >
> > >  ---------------------------------------------------------------
> > >  John Elwell (john.elwell@siemenscomms.co.uk)
> > >  Siemens Communications Limited,
> > >  ---------------------------------------------------------------
> > >  Internet communications are not secure and therefore Siemens
> > >  Communications Limited does not accept legal 
> responsibility for the
> > >  contents of this message. Any views or opinions 
> presented are solely
> > >  those of the author and do not necessarily represent 
> those of Siemens
> > >  Communications Limited unless otherwise specifically stated.
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > Sent: Tuesday, April 09, 2002 5:05 PM
> > > To: rsparks@dynamicsoft.com; sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > >
> > > Sirs,
> > >
> > > I think there are plainty of cases where C :
> > > -will not need to know that the call was started by A read in
> > > Referred-By
> > > -or will trust B (i.e: intranet cases).
> > >
> > > Protecting Referred-By also means protecting others headers.
> > >
> > > This means that we should avoid all solutions that are 
> not optional
> > > regarding C
> > > policy.
> > >
> > > A VERIFYing approach starting at C is better suited; It
> > > allows C to decide
> > > to
> > > verify or not A identity.
> > > If C does NOT want to or does NOT support VERIFY the call
> > > flows remains
> > > consistent with current behavior.
> > >
> > > PS: Referred-By is a means for carrying some useful Information.
> > >
> > > That's my 2 cents.
> > >
> > > Best Regards,
> > >
> > > Patrick
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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<TITLE>RE: [Sip] REFER security options - removing Referred-By</TITLE>
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<P><FONT SIZE=3D2>I have already pointed out that its only a SHOULD in =
the REFER spec &quot;If the URL is a SIP URL, the Referred-By header in =
the REFER request should be copied into the request sent to the =
referred-to resource.&quot;</FONT></P>

<P><FONT SIZE=3D2>Thus removal is not required if you dont like it dont =
send it. Otherwise I simply echo Brian's sentiments.</FONT>
</P>

<P><FONT SIZE=3D2>Thanks,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Rosen, Brian [<A =
HREF=3D"mailto:Brian.Rosen@marconi.com">mailto:Brian.Rosen@marconi.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, April 22, 2002 9:21 AM</FONT>
<BR><FONT SIZE=3D2>To: 'Ben Campbell'; Tom_Gray@mitel.com; Adam =
Roach</FONT>
<BR><FONT SIZE=3D2>Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert =
Sparks;</FONT>
<BR><FONT SIZE=3D2>sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] REFER security options - removing =
Referred-By</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Y'know, sometimes I (personally, not as chair) think =
we take</FONT>
<BR><FONT SIZE=3D2>the security argument just a little too far.</FONT>
</P>

<P><FONT SIZE=3D2>Sure, I can finagle the From, but Referred-By is the =
header</FONT>
<BR><FONT SIZE=3D2>I want to use in my GUI.&nbsp; I'll attach no more =
credence to it</FONT>
<BR><FONT SIZE=3D2>then I get from FROM unless it's authenticated, but =
it's damn</FONT>
<BR><FONT SIZE=3D2>useful to display to a user.</FONT>
</P>

<P><FONT SIZE=3D2>We have it implemented, we like it, and I'm keeping =
it.</FONT>
</P>

<P><FONT SIZE=3D2>Let's leave it in and define it, like FROM, as =
informational</FONT>
<BR><FONT SIZE=3D2>only.</FONT>
</P>

<P><FONT SIZE=3D2>Or shall we remove FROM on the same basis?</FONT>
</P>

<P><FONT SIZE=3D2>Brian</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Ben Campbell [<A =
HREF=3D"mailto:bcampbell@dynamicsoft.com">mailto:bcampbell@dynamicsoft.c=
om</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Monday, April 22, 2002 9:48 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Tom_Gray@mitel.com; Adam Roach</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Elwell, John'; patrick.mourot@online.fr; =
Robert Sparks;</FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] REFER security options - =
removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: sip-admin@ietf.org [<A =
HREF=3D"mailto:sip-admin@ietf.org">mailto:sip-admin@ietf.org</A>]On =
Behalf Of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Tom_Gray@Mitel.COM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: Friday, April 19, 2002 1:50 =
PM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To: Adam Roach</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cc: 'Elwell, John'; =
'patrick.mourot@online.fr'; Robert Sparks;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: RE: [Sip] REFER security options =
- removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp; The information could be useful =
because C could </FONT>
<BR><FONT SIZE=3D2>&gt; authenticate the identity</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; outside of the SIP protocol if that is =
felt necesary. In </FONT>
<BR><FONT SIZE=3D2>&gt; many cases such</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; authentication is not required. There can =
be other means</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; extra-protocol&nbsp; of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; dealing with abusive callers.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I'm not sure I see the connection between the =
last two </FONT>
<BR><FONT SIZE=3D2>&gt; sentences? How would</FONT>
<BR><FONT SIZE=3D2>&gt; one make use of an unauthenticated referred-by =
in dealing with abusive</FONT>
<BR><FONT SIZE=3D2>&gt; callers? I would tend to expect that abusive =
callers would be </FONT>
<BR><FONT SIZE=3D2>&gt; very likely</FONT>
<BR><FONT SIZE=3D2>&gt; spoof refered-by headers if it is =
allowed.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I can accept the idea of referred-by being =
useful _if_ the </FONT>
<BR><FONT SIZE=3D2>&gt; endpoint has</FONT>
<BR><FONT SIZE=3D2>&gt; another channel of authentication. I am =
skeptical about any </FONT>
<BR><FONT SIZE=3D2>&gt; real use of it</FONT>
<BR><FONT SIZE=3D2>&gt; without authentication.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Considering that Robert's original question was =
not, &quot;is </FONT>
<BR><FONT SIZE=3D2>&gt; there a use for an</FONT>
<BR><FONT SIZE=3D2>&gt; unauthenticated refered-by?&quot; but was =
instead &quot;Is there use </FONT>
<BR><FONT SIZE=3D2>&gt; for refer if it</FONT>
<BR><FONT SIZE=3D2>&gt; is published without referred-by?&quot; with =
the understanding that a solid</FONT>
<BR><FONT SIZE=3D2>&gt; referred-by header would be the subject of =
future work. From that</FONT>
<BR><FONT SIZE=3D2>&gt; perspective, I have to agree with Adam's belief =
that </FONT>
<BR><FONT SIZE=3D2>&gt; referred-by should be</FONT>
<BR><FONT SIZE=3D2>&gt; removed from the current draft.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Adam Roach =
&lt;adam@dynamicsoft.com&gt;@ietf.org on 04/19/2002 02:37:38 PM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent by:&nbsp; sip-admin@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To:&nbsp;&nbsp; &quot;'Elwell, John'&quot; =
&lt;John.Elwell@siemenscomms.co.uk&gt;,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&quot;'patrick.mourot@online.fr'&quot; </FONT>
<BR><FONT SIZE=3D2>&gt; &lt;patrick.mourot@online.fr&gt;, Robert</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Sparks&nbsp;&nbsp;&nbsp; &lt;rsparks@dynamicsoft.com&gt;, =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; cc:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject:&nbsp; RE: [Sip] REFER security =
options - removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Why is such information useful if it's =
unauthenticated?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Are you planning to use it to implement =
policy? If so,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; inclusion of such a header only sets a =
trap for naive</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; implementors.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; If not, are you envisioning this =
information will be presented</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to users? If that's your goal, you'll want =
to do something</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; that works with deployed phones. As a =
convention (and I'm not</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; proposing that this be standardized), you =
could obtain the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; desired effect with something like:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; INVITE sip:bob@example.com SIP/2.0</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: &quot;Adam Roach forwarded by Robert =
Sparks&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
&lt;sip:adam@dynamicsoft.com&gt;;tag=3D12397</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; ...</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Right?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I feel quite strongly that we should *not* =
include this sort of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; information</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; in the current REFER draft, and work on =
providing an *authenticated*</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; version of this draft at a later =
date.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; /a</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From: Elwell, John [<A =
HREF=3D"mailto:John.Elwell@siemenscomms.co.uk">mailto:John.Elwell@siemen=
scomms.co.uk</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sent: Wednesday, April 10, 2002 =
1:51</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; To: 'patrick.mourot@online.fr'; =
rsparks@dynamicsoft.com; </FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Subject: RE: [Sip] REFER security =
options - removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I agree with what Patrick Mourot =
says. Let's have a basic</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; refer capability</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; that provides an unsecured identity =
of A to C, and then leave</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; it to C to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; decide whether and how to verify the =
identity of A. This will</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; depend on</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; whether it is attended or unattended =
transfer - with attended</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; transfer the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; call A-C will exist and will be =
identified by the Replaces</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; header. This may</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; be sufficient, but if not, =
verification can be done in the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; context of that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; dialog A-C.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; =
---------------------------------------------------------------</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; John Elwell =
(john.elwell@siemenscomms.co.uk)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; Siemens Communications =
Limited,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; =
---------------------------------------------------------------</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; Internet communications are not =
secure and therefore Siemens</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; Communications Limited does not =
accept legal </FONT>
<BR><FONT SIZE=3D2>&gt; responsibility for the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; contents of this message. Any =
views or opinions </FONT>
<BR><FONT SIZE=3D2>&gt; presented are solely</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; those of the author and do not =
necessarily represent </FONT>
<BR><FONT SIZE=3D2>&gt; those of Siemens</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp; Communications Limited unless =
otherwise specifically stated.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From: patrick.mourot@online.fr [<A =
HREF=3D"mailto:patrick.mourot@online.fr">mailto:patrick.mourot@online.fr=
</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sent: Tuesday, April 09, 2002 5:05 =
PM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; To: rsparks@dynamicsoft.com; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Subject: RE: [Sip] REFER security =
options - removing Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sirs,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I think there are plainty of cases =
where C :</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -will not need to know that the call =
was started by A read in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Referred-By</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -or will trust B (i.e: intranet =
cases).</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Protecting Referred-By also means =
protecting others headers.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This means that we should avoid all =
solutions that are </FONT>
<BR><FONT SIZE=3D2>&gt; not optional</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; regarding C</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; policy.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; A VERIFYing approach starting at C is =
better suited; It</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; allows C to decide</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; verify or not A identity.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; If C does NOT want to or does NOT =
support VERIFY the call</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; flows remains</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; consistent with current =
behavior.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; PS: Referred-By is a means for =
carrying some useful Information.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; That's my 2 cents.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Best Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Patrick</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This list is for NEW development of =
the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sipping@ietf.org for new =
developments on the </FONT>
<BR><FONT SIZE=3D2>&gt; application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sip mailing list&nbsp; <A =
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<BR><FONT SIZE=3D2>&gt; &gt; &gt; This list is for NEW development of =
the core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu =
for questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Use sipping@ietf.org for new =
developments on the </FONT>
<BR><FONT SIZE=3D2>&gt; application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@ns.ietf.org  Mon Apr 22 12:21:23 2002
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Date: Mon, 22 Apr 2002 10:59:18 -0500
From: Alan Johnston <alan.johnston@wcom.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
In-reply-to: 
 <313680C9A886D511A06000204840E1CF030B51E4@whq-msgusr-02.pit.comms.marconi.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        "'Adam Roach'" <adam@dynamicsoft.com>
Cc: "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, "'Robert Sparks'" <rsparks@dynamicsoft.com>,
        sip@ietf.org
Message-id: <000d01c1ea16$aa14e240$07212aa6@ajohnston>
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I agree completely with Brian.  Many vendors have already implemented
this optional part of REFER - lets just add a sentence in the security
section then begin work on the authenticated Referred-By.

Any sort of call screening must use some other authentication scheme.
For example, in the Service Examples I-D, a proxy digest provides this
authentication - it is NEVER based on From, and should not be based on
Referred-By, either.  

Thanks,
Alan Johnston
WorldCom
sip:alan@siptest.wcom.com

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Rosen, Brian
> Sent: Monday, April 22, 2002 9:21 AM
> To: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach
> Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks; 
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> Y'know, sometimes I (personally, not as chair) think we take 
> the security argument just a little too far.
> 
> Sure, I can finagle the From, but Referred-By is the header
> I want to use in my GUI.  I'll attach no more credence to it 
> then I get from FROM unless it's authenticated, but it's damn 
> useful to display to a user.
> 
> We have it implemented, we like it, and I'm keeping it.
> 
> Let's leave it in and define it, like FROM, as informational only.
> 
> Or shall we remove FROM on the same basis?
> 
> Brian
> 
> > -----Original Message-----
> > From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> > Sent: Monday, April 22, 2002 9:48 AM
> > To: Tom_Gray@mitel.com; Adam Roach
> > Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks; 
> > sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of 
> > > Tom_Gray@Mitel.COM
> > > Sent: Friday, April 19, 2002 1:50 PM
> > > To: Adam Roach
> > > Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks; 
> > > sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > >
> > >  The information could be useful because C could
> > authenticate the identity
> > > outside of the SIP protocol if that is felt necesary. In
> > many cases such
> > > authentication is not required. There can be other means 
> > > extra-protocol  of dealing with abusive callers.
> > >
> > 
> > I'm not sure I see the connection between the last two 
> > sentences? How would
> > one make use of an unauthenticated referred-by in dealing 
> with abusive
> > callers? I would tend to expect that abusive callers would be 
> > very likely
> > spoof refered-by headers if it is allowed.
> > 
> > I can accept the idea of referred-by being useful _if_ the 
> > endpoint has
> > another channel of authentication. I am skeptical about any 
> > real use of it
> > without authentication.
> > 
> > Considering that Robert's original question was not, "is 
> > there a use for an
> > unauthenticated refered-by?" but was instead "Is there use 
> > for refer if it
> > is published without referred-by?" with the understanding 
> that a solid
> > referred-by header would be the subject of future work. From that
> > perspective, I have to agree with Adam's belief that 
> > referred-by should be
> > removed from the current draft.
> > 
> > 
> > >
> > >
> > >
> > >
> > >
> > > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 
> 02:37:38 PM
> > >
> > > Sent by:  sip-admin@ietf.org
> > >
> > >
> > > To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
> > >       "'patrick.mourot@online.fr'" 
> > <patrick.mourot@online.fr>, Robert
> > >       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> > > cc:
> > >
> > > Subject:  RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > Why is such information useful if it's unauthenticated?
> > >
> > > Are you planning to use it to implement policy? If so,
> > > inclusion of such a header only sets a trap for naive
> > > implementors.
> > >
> > > If not, are you envisioning this information will be presented
> > > to users? If that's your goal, you'll want to do something
> > > that works with deployed phones. As a convention (and I'm not
> > > proposing that this be standardized), you could obtain the
> > > desired effect with something like:
> > >
> > > INVITE sip:bob@example.com SIP/2.0
> > > From: "Adam Roach forwarded by Robert Sparks"
> > > <sip:adam@dynamicsoft.com>;tag=12397
> > > ...
> > >
> > > Right?
> > >
> > > I feel quite strongly that we should *not* include this sort of
> > > information
> > > in the current REFER draft, and work on providing an 
> *authenticated*
> > > version of this draft at a later date.
> > >
> > > /a
> > >
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > > > Sent: Wednesday, April 10, 2002 1:51
> > > > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; 
> > sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > > I agree with what Patrick Mourot says. Let's have a basic
> > > > refer capability
> > > > that provides an unsecured identity of A to C, and then leave
> > > > it to C to
> > > > decide whether and how to verify the identity of A. This will
> > > > depend on
> > > > whether it is attended or unattended transfer - with attended
> > > > transfer the
> > > > call A-C will exist and will be identified by the Replaces
> > > > header. This may
> > > > be sufficient, but if not, verification can be done in the
> > > > context of that
> > > > dialog A-C.
> > > >
> > > >  ---------------------------------------------------------------
> > > >  John Elwell (john.elwell@siemenscomms.co.uk)
> > > >  Siemens Communications Limited,
> > > >  ---------------------------------------------------------------
> > > >  Internet communications are not secure and therefore Siemens
> > > >  Communications Limited does not accept legal 
> > responsibility for the
> > > >  contents of this message. Any views or opinions 
> > presented are solely
> > > >  those of the author and do not necessarily represent 
> > those of Siemens
> > > >  Communications Limited unless otherwise specifically stated.
> > > >
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > > Sent: Tuesday, April 09, 2002 5:05 PM
> > > > To: rsparks@dynamicsoft.com; sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > >
> > > > Sirs,
> > > >
> > > > I think there are plainty of cases where C :
> > > > -will not need to know that the call was started by A read in
> > > > Referred-By
> > > > -or will trust B (i.e: intranet cases).
> > > >
> > > > Protecting Referred-By also means protecting others headers.
> > > >
> > > > This means that we should avoid all solutions that are 
> > not optional
> > > > regarding C
> > > > policy.
> > > >
> > > > A VERIFYing approach starting at C is better suited; It
> > > > allows C to decide
> > > > to
> > > > verify or not A identity.
> > > > If C does NOT want to or does NOT support VERIFY the call
> > > > flows remains
> > > > consistent with current behavior.
> > > >
> > > > PS: Referred-By is a means for carrying some useful Information.
> > > >
> > > > That's my 2 cents.
> > > >
> > > > Best Regards,
> > > >
> > > > Patrick
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'" <mat@cisco.com>
Cc: <Tom_Gray@mitel.com>, "Adam Roach" <adam@dynamicsoft.com>,
        "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
        <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 10:56:55 -0500
Message-ID: <HNEOJECGFHIABDLENMMCEELICGAA.bcampbell@dynamicsoft.com>
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But of course there are other standard mechanisms by which you can
authenticate the caller. I was under the impression that is not currently
the case for the referror. If we _had_ a strong, agreed upon approach for
doing that, then referred-by would not be a big deal.

Robert did _not_ suggest making referred-by go away forever, just that we
link it to the effort of figuring out how to authenticate referrors, and
letting the refer draft move forward without blocking on that effort. Robert
had proposed some potential mechansism in Minneapolis, but I was under the
impression that it would take more work to complete them to the satisfaction
of the security guys.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Monday, April 22, 2002 10:43 AM
> To: 'Michael Thomas'
> Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
> Well, since it's okay to put George Bush's name in the FROM header,
> it's got to be okay to put Kofi Anann's admin's name in the Referred-By
> header.  If Mr. Annan accepts the forged FROM, he will accept the
> forged referral.
>
> These headers are (presently) un-authenticated.  We don't DO anything
> with them (presently) other than show them in a GUI.  It's useful to
> do that.  I'll accept an authentication mechanism some day, but
> an un-authenticated field is useful.
>
> Brian
>
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Monday, April 22, 2002 11:31 AM
> > To: Rosen, Brian
> > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > Rosen, Brian writes:
> >  > Y'know, sometimes I (personally, not as chair) think we take
> >  > the security argument just a little too far.
> >  >
> >  > Sure, I can finagle the From, but Referred-By is the header
> >  > I want to use in my GUI.  I'll attach no more credence to it
> >  > then I get from FROM unless it's authenticated, but it's damn
> >  > useful to display to a user.
> >
> >    So you're saying that it's OK for me to be able
> >    to just put Kofi Anann's admin's name in the
> >    Referred-by header, so that I can get through
> >    his screening so we can chat about my thoughts
> >    on San Francisco annexing Canada?
> >
> >    Groovy.
> >
> > 		    Mike
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


_______________________________________________
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From daemon@ns.ietf.org  Mon Apr 22 12:32:49 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Michael Thomas'" <mat@cisco.com>
Cc: "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>,
        "'Elwell, John'"
	 <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 11:42:47 -0400
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Well, since it's okay to put George Bush's name in the FROM header,
it's got to be okay to put Kofi Anann's admin's name in the Referred-By
header.  If Mr. Annan accepts the forged FROM, he will accept the
forged referral.  

These headers are (presently) un-authenticated.  We don't DO anything
with them (presently) other than show them in a GUI.  It's useful to
do that.  I'll accept an authentication mechanism some day, but
an un-authenticated field is useful.

Brian

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Monday, April 22, 2002 11:31 AM
> To: Rosen, Brian
> Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> Rosen, Brian writes:
>  > Y'know, sometimes I (personally, not as chair) think we take
>  > the security argument just a little too far.
>  > 
>  > Sure, I can finagle the From, but Referred-By is the header
>  > I want to use in my GUI.  I'll attach no more credence to it
>  > then I get from FROM unless it's authenticated, but it's damn
>  > useful to display to a user.
> 
>    So you're saying that it's OK for me to be able
>    to just put Kofi Anann's admin's name in the
>    Referred-by header, so that I can get through
>    his screening so we can chat about my thoughts
>    on San Francisco annexing Canada?
> 
>    Groovy.
> 
> 		    Mike
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Mon Apr 22 12:58:47 2002
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Date: Mon, 22 Apr 2002 19:37:16 +0300
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Michael Thomas'" <mat@cisco.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>,
        "'Elwell, John'"
	 <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Rosen, Brian writes:

 > These headers are (presently) un-authenticated.  

not in my service.  as i have said many times, we do check that the from
header matches with ones that my user is supposed to be using.  i don't
think there is any statement in the specs that says that we can't do so.
this is the same things as checking the source ip address, which we also
do.  internet would be much more user friendly if everybody would do so.

-- juha


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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Michael Thomas'" <mat@cisco.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>,
        "'Elwell, John'"
	 <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Rosen, Brian writes:
 > Well, since it's okay to put George Bush's name in the FROM header,
 > it's got to be okay to put Kofi Anann's admin's name in the Referred-By
 > header.  If Mr. Annan accepts the forged FROM, he will accept the
 > forged referral.  
 > 
 > These headers are (presently) un-authenticated.  We don't DO anything
 > with them (presently) other than show them in a GUI.  It's useful to
 > do that.  I'll accept an authentication mechanism some day, but
 > an un-authenticated field is useful.

   A trip back to first principles might be a good
   idea here. Back in the 50's when my secretary
   Gladys would get me my coffee, she would also buzz
   me in my spacious office when I had an incoming call
   that I should take. Since Gladys was the source of
   my morning caffeine, her voice imprinted at very
   deep level. Thus, when I had an incoming call I
   could be quite sure that an imposter wasn't
   taking away valuable time from my helm of the
   Department of Loafing.

   Fast forward 50 years. Gladys is by and large
   history, and most assuredly the morning coffee
   she used get me is no more. Along with it went
   the deep imprinting, and the spacious office. Now I
   have to rely on a little LCD display that
   doesn't bring me coffee and now you're telling
   me that unimportant people will be able make
   their way through my screening and reach the
   inner sanctum of Loafing decision making with
   complete impunity?

   Progress. Gotta love it.

		   Mike

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From daemon@ns.ietf.org  Mon Apr 22 13:12:35 2002
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Date: Mon, 22 Apr 2002 09:48:43 -0700
To: "Dean Willis" <dwillis@dynamicsoft.com>, <sip@ietf.org>
From: AC Mahendran <mahendra@qualcomm.com>
Subject: Re: [Sip] Revised Service Route Discovery Draft
Cc: "3GPP_TSG_CN_WG1" <3GPP_TSG_CN_WG1@list.etsi.fr>
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Dean:

The Record-Route headers in section 5.6 (in transactions F3 & F5) are in 
the wrong order.

Also, doesn't the "Path" header provide the same functionality as the 
"P-Service-Route" ?

thanks,
AC


At 06:15 PM 4/19/2002 -0500, Dean Willis wrote:

>I've submitted draft-willis-sip-svcrtdisco-01.txt to the Internet Drafts
>queue. This version includs refinements made by Bernie Hoeneisen to my
>earlier draft.
>
>The draft should be announced shortly on ietf-announce.
>
>In the meantime, please review:
>
>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.txt
>
>or
>
>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.htm
>l
>
>or
>
>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.xml
>
>
>This is a "P-header" draft addressing 3GPP's required header for
>reporting a service proxy assignment in the REGISTER response.
>
>
>Please review ASAP. We need to get this to IETF last call by the end of
>the month if at all possible.
>
>--
>Dean
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 10:55:34 -0600
Message-ID: <EF4C65F18BE6464B8E9DF3C212B6B2930168E7A1@cof110avexu1.global.avaya.com>
Thread-Topic: [Sip] REFER security options - removing Referred-By
Thread-Index: AcHqFCDyRgVuqZo9S7ij0r7XcMUGogACQOEg
From: "Zmolek, Andrew (Andrew)" <zmolek@avaya.com>
To: "Michael Thomas" <mat@cisco.com>, "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "Ben Campbell" <bcampbell@dynamicsoft.com>, <Tom_Gray@mitel.com>,
        "Adam Roach" <adam@dynamicsoft.com>,
        "Elwell, John" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
        <sip@ietf.org>
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Michael Thomas wrote:

>    So you're saying that it's OK for me to be able
>    to just put Kofi Anann's admin's name in the
>    Referred-by header, so that I can get through
>    his screening so we can chat about my thoughts
>    on San Francisco annexing Canada?

As long as you believe it's OK for email, it's OK for SIP. How's that for some moral relativity?

In fact, it's possible to claim another persons identity without even using an IETF protocol. It's a real shame there isn't a secure protocol to eliminate social engineering. 

I think it's clear that a secure Referred-By should happen. I'm not going to suggest that we should be satisfied with unauthenticated Referred-by information.

But let's not throw the baby out with the bathwater, guys.

--Andy Zmolek 
    Technology & Standards Engineer 
      CTO Standards 
        Avaya Inc. 

            zmolek@avaya.com 
              +1 720 444 4001 



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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Zmolek, Andrew \(Andrew\)" <zmolek@avaya.com>,
        "Michael Thomas" <mat@cisco.com>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: <Tom_Gray@mitel.com>, "Adam Roach" <adam@dynamicsoft.com>,
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Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 12:26:48 -0500
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> -----Original Message-----
> From: Zmolek, Andrew (Andrew) [mailto:zmolek@avaya.com]
> Sent: Monday, April 22, 2002 11:56 AM
> To: Michael Thomas; Rosen, Brian
> Cc: Ben Campbell; Tom_Gray@mitel.com; Adam Roach; Elwell, John;
> patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
> Michael Thomas wrote:
>
> >    So you're saying that it's OK for me to be able
> >    to just put Kofi Anann's admin's name in the
> >    Referred-by header, so that I can get through
> >    his screening so we can chat about my thoughts
> >    on San Francisco annexing Canada?
>
> As long as you believe it's OK for email, it's OK for SIP. How's
> that for some moral relativity?

No. It's not. Email does not demand my immediate attention. (OK, so it does,
but I'm an email addict, and assume that not _everyone_ on this list shares
my affliction.) Email allows me to compose my thoughts, read carefully, and
respond at my leisure. The phone on the other hand rips my brain from one
context to another. It demands me to immediately shift gears to the callers
subject. I respond with a well reasoned and eloquent "huh?" as I'm trying to
remember the subject the caller wants to talk to me about. Usually this
memory recall happens some minutes after we hang up, leaving me wondering
how stupid I really did sound to the caller.

OK, I exaggerate--but the point is, interruptive, synchronous, intrusive
communication should be held to a higher authentication and authorization
standard than asynchronous communication that I can respond to when I get
good and ready.

>
> In fact, it's possible to claim another persons identity without
> even using an IETF protocol. It's a real shame there isn't a
> secure protocol to eliminate social engineering.

Heh. There are technologies that _could_ be used for this, if anyone wanted
to go to the trouble. The fact that social engineering works so well is a
testament to what end users will do with unauthenticated identity info if
you let them.

>
> I think it's clear that a secure Referred-By should happen. I'm
> not going to suggest that we should be satisfied with
> unauthenticated Referred-by information.
>
> But let's not throw the baby out with the bathwater, guys.

No one suggested that. They only suggested not delivering the baby
prematurely, and not making the rest of refer depend on its presence.





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From daemon@ns.ietf.org  Mon Apr 22 13:56:40 2002
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Subject: RE: [Sip] REFER security options - removing Referred-By
To: Michael Thomas <mat@cisco.com>
Cc: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'" <mat@cisco.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>, Tom_Gray@Mitel.COM,
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Why would there be impunity. This can be handled as it is handled now
outside of any protocol. VIPs can use their workplace authority to prevent
unpleasnt calls as they useit to prevent unpleasant visitors. They have all
had experience with people who want to attach themsleves to certain types
of  meetings.





Michael Thomas <mat@cisco.com>@ietf.org on 04/22/2002 12:33:54 PM

Sent by:  sip-admin@ietf.org


To:   "Rosen, Brian" <Brian.Rosen@marconi.com>
cc:   "'Michael Thomas'" <mat@cisco.com>, "'Ben Campbell'"
      <bcampbell@dynamicsoft.com>, Tom_Gray@Mitel.COM, Adam Roach
      <adam@dynamicsoft.com>, "'Elwell, John'"
      <John.Elwell@siemenscomms.co.uk>, patrick.mourot@online.fr, Robert
      Sparks <rsparks@dynamicsoft.com>, sip@ietf.org

Subject:  RE: [Sip] REFER security options - removing Referred-By


Rosen, Brian writes:
 > Well, since it's okay to put George Bush's name in the FROM header,
 > it's got to be okay to put Kofi Anann's admin's name in the Referred-By
 > header.  If Mr. Annan accepts the forged FROM, he will accept the
 > forged referral.
 >
 > These headers are (presently) un-authenticated.  We don't DO anything
 > with them (presently) other than show them in a GUI.  It's useful to
 > do that.  I'll accept an authentication mechanism some day, but
 > an un-authenticated field is useful.

   A trip back to first principles might be a good
   idea here. Back in the 50's when my secretary
   Gladys would get me my coffee, she would also buzz
   me in my spacious office when I had an incoming call
   that I should take. Since Gladys was the source of
   my morning caffeine, her voice imprinted at very
   deep level. Thus, when I had an incoming call I
   could be quite sure that an imposter wasn't
   taking away valuable time from my helm of the
   Department of Loafing.

   Fast forward 50 years. Gladys is by and large
   history, and most assuredly the morning coffee
   she used get me is no more. Along with it went
   the deep imprinting, and the spacious office. Now I
   have to rely on a little LCD display that
   doesn't bring me coffee and now you're telling
   me that unimportant people will be able make
   their way through my screening and reach the
   inner sanctum of Loafing decision making with
   complete impunity?

   Progress. Gotta love it.

             Mike

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Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 11:35:29 -0600
Message-ID: <EF4C65F18BE6464B8E9DF3C212B6B2930168E7C1@cof110avexu1.global.avaya.com>
Thread-Topic: [Sip] REFER security options - removing Referred-By
Thread-Index: AcHqIxKW41vsOyrqS6qENGqAM4BpjQAAJcbg
From: "Zmolek, Andrew (Andrew)" <zmolek@avaya.com>
To: "Ben Campbell" <bcampbell@dynamicsoft.com>,
        "Michael Thomas" <mat@cisco.com>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: <Tom_Gray@mitel.com>, "Adam Roach" <adam@dynamicsoft.com>,
        "Elwell, John" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
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> > But let's not throw the baby out with the bathwater, guys.
> 
> No one suggested that. They only suggested not delivering the baby
> prematurely, and not making the rest of refer depend on its presence.

Since we already have a number of implementations of non-secure Referred-By, would it be fair to call this an attempt at a late-term abortion?

--Andy

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To: Michael Thomas <mat@cisco.com>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
X-Mailer: Lotus Notes Release 5.0.5  September 22, 2000
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Mike,

none of the headers can be trusted. If we move in a direction in 
which nothing is considered valid and useable unless authenticated, 
SIP will have a long long way to go. 


Frank




Michael Thomas <mat@cisco.com>
Sent by: sip-admin@ietf.org
22-04-2002 05:31 PM

 
        To:     "Rosen, Brian" <Brian.Rosen@marconi.com>
        cc:     "'Ben Campbell'" <bcampbell@dynamicsoft.com>
Tom_Gray@mitel.com
Adam Roach <adam@dynamicsoft.com>
"'Elwell, John'" <John.Elwell@siemenscomms.co.uk>
patrick.mourot@online.fr
Robert Sparks <rsparks@dynamicsoft.com>
sip@ietf.org
(bcc: Frank Derks/HVS/BE/PHILIPS)
        Subject:        RE: [Sip] REFER security options - removing Referred-By
        Classification: 



Rosen, Brian writes:
 > Y'know, sometimes I (personally, not as chair) think we take
 > the security argument just a little too far.
 > 
 > Sure, I can finagle the From, but Referred-By is the header
 > I want to use in my GUI.  I'll attach no more credence to it
 > then I get from FROM unless it's authenticated, but it's damn
 > useful to display to a user.

   So you're saying that it's OK for me to be able
   to just put Kofi Anann's admin's name in the
   Referred-by header, so that I can get through
   his screening so we can chat about my thoughts
   on San Francisco annexing Canada?

   Groovy.

                                     Mike

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--=_alternative 006A3357C1256BA3_=
Content-Type: text/html; charset="us-ascii"


<br><font size=2 face="Courier">Mike,</font>
<br>
<br><font size=2 face="Courier">none of the headers can be trusted. If we move in a direction in </font>
<br><font size=2 face="Courier">which nothing is considered valid and useable unless authenticated, </font>
<br><font size=2 face="Courier">SIP will have a long long way to go. <br>
<br>
</font>
<br><font size=2 face="Courier">Frank</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>Michael Thomas &lt;mat@cisco.com&gt;</b></font>
<br><font size=1 face="sans-serif">Sent by: sip-admin@ietf.org</font>
<p><font size=1 face="sans-serif">22-04-2002 05:31 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&quot;Rosen, Brian&quot; &lt;Brian.Rosen@marconi.com&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;&quot;'Ben Campbell'&quot; &lt;bcampbell@dynamicsoft.com&gt;<br>
Tom_Gray@mitel.com<br>
Adam Roach &lt;adam@dynamicsoft.com&gt;<br>
&quot;'Elwell, John'&quot; &lt;John.Elwell@siemenscomms.co.uk&gt;<br>
patrick.mourot@online.fr<br>
Robert Sparks &lt;rsparks@dynamicsoft.com&gt;<br>
sip@ietf.org<br>
(bcc: Frank Derks/HVS/BE/PHILIPS)</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sip] REFER security options - removing Referred-By</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">Rosen, Brian writes:<br>
 &gt; Y'know, sometimes I (personally, not as chair) think we take<br>
 &gt; the security argument just a little too far.<br>
 &gt; <br>
 &gt; Sure, I can finagle the From, but Referred-By is the header<br>
 &gt; I want to use in my GUI. &nbsp;I'll attach no more credence to it<br>
 &gt; then I get from FROM unless it's authenticated, but it's damn<br>
 &gt; useful to display to a user.<br>
<br>
 &nbsp; So you're saying that it's OK for me to be able<br>
 &nbsp; to just put Kofi Anann's admin's name in the<br>
 &nbsp; Referred-by header, so that I can get through<br>
 &nbsp; his screening so we can chat about my thoughts<br>
 &nbsp; on San Francisco annexing Canada?<br>
<br>
 &nbsp; Groovy.<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Mike<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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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 sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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To: Tom_Gray@Mitel.COM
Cc: sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
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Tom,

the same would apply to fraudulent From headers, if similar services are 
built on it. Obviously, we can then again get into the issue that the From
header can not be trusted either, but it simply happens to be what's there
today. 

Authentication may certainly be an issue, but it doesn't help to assume 
that it is always an issue. If we start doing this, then nothing in the
SIP message can be trusted and consequently we cannot build any services
(including a simple "call") with SIP as it stands today.

It is not impossible that within certain domains (e.g. a company) the 
SIP telephones in use can be configured, by design or by the system 
administrator, so the problems we are all so afraid of _can_ not occur. 
Furthermore, if someone then tries to abuse the system with a softphone 
that can get around such limitations, there are ways to "detect" that 
this phone is not a "device" that is allowed to use the system.

Headers and the message body are there to convey information, if we state
that none of this information can be trusted and that we can therefore not 

make use of them unless they are authenticated, SIP will have a long way 
to 
go before widespread adoption. 


Regards,

Frank




Tom_Gray@Mitel.COM
Sent by: sip-admin@ietf.org
22-04-2002 05:00 PM

 
        To:     "Ben Campbell" <bcampbell@dynamicsoft.com>
        cc:     <Tom_Gray@Mitel.COM>
"Adam Roach" <adam@dynamicsoft.com>
"'Elwell, John'" <John.Elwell@siemenscomms.co.uk>
<patrick.mourot@online.fr>
"Robert Sparks" <rsparks@dynamicsoft.com>
<sip@ietf.org>
(bcc: Frank Derks/HVS/BE/PHILIPS)
        Subject:        RE: [Sip] REFER security options - removing Referred-By
        Classification: 






In response to the question of when would authenticated referred-by's not
be required, a valid example would be in an enterprise setting where
extra-protocol means of dealing with abuse can easily be used.

Taking the original scenario that led to the discussion of the need for
authentication of Refered-By's (at least as I remember it).

a) Alice is a CEO who only accepts calls screened by her secretary Bob.

b) Carole is an employee who discovers she can call Alice directly  by
creating a fraudulent Refered-By

c) Carole calls Alice and annoys her with some petty grievances.

d) Alice calls Bob and tells him to not allow calls from Carole any more

e) Bob tells Alice that he did not put the call through and that Carole
must have did it herself with an illicit use of the telephone system

f) Alice does not, as in the originally proposed scenario, order the
immedate removal of the PBX. Instead she tells Bob to take care of the
matter
 .
g) Bob calls Carole's manager Dave and they inform her that further
incidents of this sort will lead to unpleasant consequences.

The scenario could be extended with modifications to public network
situations.





"Ben Campbell" <bcampbell@dynamicsoft.com> on 04/22/2002 09:48:19 AM

To:   <Tom_Gray@Mitel.COM>, "Adam Roach" <adam@dynamicsoft.com>
cc:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
      <patrick.mourot@online.fr>, "Robert Sparks"
      <rsparks@dynamicsoft.com>, <sip@ietf.org>

Subject:  RE: [Sip] REFER security options - removing Referred-By




> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Tom_Gray@Mitel.COM
> Sent: Friday, April 19, 2002 1:50 PM
> To: Adam Roach
> Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
>
>  The information could be useful because C could authenticate the
identity
> outside of the SIP protocol if that is felt necesary. In many cases such
> authentication is not required. There can be other means
> extra-protocol  of
> dealing with abusive callers.
>

I'm not sure I see the connection between the last two sentences? How 
would
one make use of an unauthenticated referred-by in dealing with abusive
callers? I would tend to expect that abusive callers would be very likely
spoof refered-by headers if it is allowed.

I can accept the idea of referred-by being useful _if_ the endpoint has
another channel of authentication. I am skeptical about any real use of it
without authentication.

Considering that Robert's original question was not, "is there a use for 
an
unauthenticated refered-by?" but was instead "Is there use for refer if it
is published without referred-by?" with the understanding that a solid
referred-by header would be the subject of future work. From that
perspective, I have to agree with Adam's belief that referred-by should be
removed from the current draft.


>
>
>
>
>
> Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
>
> Sent by:  sip-admin@ietf.org
>
>
> To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
>       "'patrick.mourot@online.fr'" <patrick.mourot@online.fr>, Robert
>       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> cc:
>
> Subject:  RE: [Sip] REFER security options - removing Referred-By
>
>
> Why is such information useful if it's unauthenticated?
>
> Are you planning to use it to implement policy? If so,
> inclusion of such a header only sets a trap for naive
> implementors.
>
> If not, are you envisioning this information will be presented
> to users? If that's your goal, you'll want to do something
> that works with deployed phones. As a convention (and I'm not
> proposing that this be standardized), you could obtain the
> desired effect with something like:
>
> INVITE sip:bob@example.com SIP/2.0
> From: "Adam Roach forwarded by Robert Sparks"
> <sip:adam@dynamicsoft.com>;tag=12397
> ...
>
> Right?
>
> I feel quite strongly that we should *not* include this sort of
> information
> in the current REFER draft, and work on providing an *authenticated*
> version of this draft at a later date.
>
> /a
>
> > -----Original Message-----
> > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > Sent: Wednesday, April 10, 2002 1:51
> > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > I agree with what Patrick Mourot says. Let's have a basic
> > refer capability
> > that provides an unsecured identity of A to C, and then leave
> > it to C to
> > decide whether and how to verify the identity of A. This will
> > depend on
> > whether it is attended or unattended transfer - with attended
> > transfer the
> > call A-C will exist and will be identified by the Replaces
> > header. This may
> > be sufficient, but if not, verification can be done in the
> > context of that
> > dialog A-C.
> >
> >  ---------------------------------------------------------------
> >  John Elwell (john.elwell@siemenscomms.co.uk)
> >  Siemens Communications Limited,
> >  ---------------------------------------------------------------
> >  Internet communications are not secure and therefore Siemens
> >  Communications Limited does not accept legal responsibility for the
> >  contents of this message. Any views or opinions presented are solely
> >  those of the author and do not necessarily represent those of Siemens
> >  Communications Limited unless otherwise specifically stated.
> >
> >
> >
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > Sent: Tuesday, April 09, 2002 5:05 PM
> > To: rsparks@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> >
> > Sirs,
> >
> > I think there are plainty of cases where C :
> > -will not need to know that the call was started by A read in
> > Referred-By
> > -or will trust B (i.e: intranet cases).
> >
> > Protecting Referred-By also means protecting others headers.
> >
> > This means that we should avoid all solutions that are not optional
> > regarding C
> > policy.
> >
> > A VERIFYing approach starting at C is better suited; It
> > allows C to decide
> > to
> > verify or not A identity.
> > If C does NOT want to or does NOT support VERIFY the call
> > flows remains
> > consistent with current behavior.
> >
> > PS: Referred-By is a means for carrying some useful Information.
> >
> > That's my 2 cents.
> >
> > Best Regards,
> >
> > Patrick
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip





_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



--=_alternative 006A223FC1256BA3_=
Content-Type: text/html; charset="us-ascii"


<br><font size=2 face="Courier"><br>
Tom,</font>
<br>
<br><font size=2 face="Courier">the same would apply to fraudulent From headers, if similar services are </font>
<br><font size=2 face="Courier">built on it. Obviously, we can then again get into the issue that the From</font>
<br><font size=2 face="Courier">header can not be trusted either, but it simply happens to be what's there</font>
<br><font size=2 face="Courier">today. </font>
<br>
<br><font size=2 face="Courier">Authentication may certainly be an issue, but it doesn't help to assume </font>
<br><font size=2 face="Courier">that it is always an issue. If we start doing this, then nothing in the</font>
<br><font size=2 face="Courier">SIP message can be trusted and consequently we cannot build any services</font>
<br><font size=2 face="Courier">(including a simple &quot;call&quot;) with SIP as it stands today.</font>
<br>
<br><font size=2 face="Courier">It is not impossible that within certain domains (e.g. a company) the </font>
<br><font size=2 face="Courier">SIP telephones in use can be configured, by design or by the system </font>
<br><font size=2 face="Courier">administrator, so the problems we are all so afraid of _can_ not occur. </font>
<br><font size=2 face="Courier">Furthermore, if someone then tries to abuse the system with a softphone </font>
<br><font size=2 face="Courier">that can get around such limitations, there are ways to &quot;detect&quot; that </font>
<br><font size=2 face="Courier">this phone is not a &quot;device&quot; that is allowed to use the system.</font>
<br>
<br><font size=2 face="Courier">Headers and the message body are there to convey information, if we state</font>
<br><font size=2 face="Courier">that none of this information can be trusted and that we can therefore not </font>
<br><font size=2 face="Courier">make use of them unless they are authenticated, SIP will have a long way to </font>
<br><font size=2 face="Courier">go before widespread adoption. </font>
<br>
<br>
<br><font size=2 face="Courier">Regards,</font>
<br>
<br><font size=2 face="Courier">Frank</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="Courier"><b>Tom_Gray@Mitel.COM</b></font>
<br><font size=1 face="Courier">Sent by: sip-admin@ietf.org</font>
<p><font size=1 face="Courier">22-04-2002 05:00 PM</font>
<br>
<td><font size=1 face="Courier">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="Courier">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&quot;Ben Campbell&quot; &lt;bcampbell@dynamicsoft.com&gt;</font>
<br><font size=1 face="Courier">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;&lt;Tom_Gray@Mitel.COM&gt;<br>
&quot;Adam Roach&quot; &lt;adam@dynamicsoft.com&gt;<br>
&quot;'Elwell, John'&quot; &lt;John.Elwell@siemenscomms.co.uk&gt;<br>
&lt;patrick.mourot@online.fr&gt;<br>
&quot;Robert Sparks&quot; &lt;rsparks@dynamicsoft.com&gt;<br>
&lt;sip@ietf.org&gt;<br>
(bcc: Frank Derks/HVS/BE/PHILIPS)</font>
<br><font size=1 face="Courier">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sip] REFER security options - removing Referred-By</font>
<p><font size=1 face="Courier">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier"><br>
<br>
<br>
In response to the question of when would authenticated referred-by's not<br>
be required, a valid example would be in an enterprise setting where<br>
extra-protocol means of dealing with abuse can easily be used.<br>
<br>
Taking the original scenario that led to the discussion of the need for<br>
authentication of Refered-By's (at least as I remember it).<br>
<br>
a) Alice is a CEO who only accepts calls screened by her secretary Bob.<br>
<br>
b) Carole is an employee who discovers she can call Alice directly &nbsp;by<br>
creating a fraudulent Refered-By<br>
<br>
c) Carole calls Alice and annoys her with some petty grievances.<br>
<br>
d) Alice calls Bob and tells him to not allow calls from Carole any more<br>
<br>
e) Bob tells Alice that he did not put the call through and that Carole<br>
must have did it herself with an illicit use of the telephone system<br>
<br>
f) Alice does not, as in the originally proposed scenario, order the<br>
immedate removal of the PBX. Instead she tells Bob to take care of the<br>
matter<br>
 .<br>
g) Bob calls Carole's manager Dave and they inform her that further<br>
incidents of this sort will lead to unpleasant consequences.<br>
<br>
The scenario could be extended with modifications to public network<br>
situations.<br>
<br>
<br>
<br>
<br>
<br>
&quot;Ben Campbell&quot; &lt;bcampbell@dynamicsoft.com&gt; on 04/22/2002 09:48:19 AM<br>
<br>
To: &nbsp; &lt;Tom_Gray@Mitel.COM&gt;, &quot;Adam Roach&quot; &lt;adam@dynamicsoft.com&gt;<br>
cc: &nbsp; &quot;'Elwell, John'&quot; &lt;John.Elwell@siemenscomms.co.uk&gt;,<br>
 &nbsp; &nbsp; &nbsp;&lt;patrick.mourot@online.fr&gt;, &quot;Robert Sparks&quot;<br>
 &nbsp; &nbsp; &nbsp;&lt;rsparks@dynamicsoft.com&gt;, &lt;sip@ietf.org&gt;<br>
<br>
Subject: &nbsp;RE: [Sip] REFER security options - removing Referred-By<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of<br>
&gt; Tom_Gray@Mitel.COM<br>
&gt; Sent: Friday, April 19, 2002 1:50 PM<br>
&gt; To: Adam Roach<br>
&gt; Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;<br>
&gt; sip@ietf.org<br>
&gt; Subject: RE: [Sip] REFER security options - removing Referred-By<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; &nbsp;The information could be useful because C could authenticate the<br>
identity<br>
&gt; outside of the SIP protocol if that is felt necesary. In many cases such<br>
&gt; authentication is not required. There can be other means<br>
&gt; extra-protocol &nbsp;of<br>
&gt; dealing with abusive callers.<br>
&gt;<br>
<br>
I'm not sure I see the connection between the last two sentences? How would<br>
one make use of an unauthenticated referred-by in dealing with abusive<br>
callers? I would tend to expect that abusive callers would be very likely<br>
spoof refered-by headers if it is allowed.<br>
<br>
I can accept the idea of referred-by being useful _if_ the endpoint has<br>
another channel of authentication. I am skeptical about any real use of it<br>
without authentication.<br>
<br>
Considering that Robert's original question was not, &quot;is there a use for an<br>
unauthenticated refered-by?&quot; but was instead &quot;Is there use for refer if it<br>
is published without referred-by?&quot; with the understanding that a solid<br>
referred-by header would be the subject of future work. From that<br>
perspective, I have to agree with Adam's belief that referred-by should be<br>
removed from the current draft.<br>
<br>
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; Adam Roach &lt;adam@dynamicsoft.com&gt;@ietf.org on 04/19/2002 02:37:38 PM<br>
&gt;<br>
&gt; Sent by: &nbsp;sip-admin@ietf.org<br>
&gt;<br>
&gt;<br>
&gt; To: &nbsp; &quot;'Elwell, John'&quot; &lt;John.Elwell@siemenscomms.co.uk&gt;,<br>
&gt; &nbsp; &nbsp; &nbsp; &quot;'patrick.mourot@online.fr'&quot; &lt;patrick.mourot@online.fr&gt;, Robert<br>
&gt; &nbsp; &nbsp; &nbsp; Sparks &nbsp; &nbsp;&lt;rsparks@dynamicsoft.com&gt;, sip@ietf.org<br>
&gt; cc:<br>
&gt;<br>
&gt; Subject: &nbsp;RE: [Sip] REFER security options - removing Referred-By<br>
&gt;<br>
&gt;<br>
&gt; Why is such information useful if it's unauthenticated?<br>
&gt;<br>
&gt; Are you planning to use it to implement policy? If so,<br>
&gt; inclusion of such a header only sets a trap for naive<br>
&gt; implementors.<br>
&gt;<br>
&gt; If not, are you envisioning this information will be presented<br>
&gt; to users? If that's your goal, you'll want to do something<br>
&gt; that works with deployed phones. As a convention (and I'm not<br>
&gt; proposing that this be standardized), you could obtain the<br>
&gt; desired effect with something like:<br>
&gt;</font>
<br><font size=2 face="Courier">&gt; INVITE sip:bob@example.com SIP/2.0<br>
&gt; From: &quot;Adam Roach forwarded by Robert Sparks&quot;<br>
&gt; &lt;sip:adam@dynamicsoft.com&gt;;tag=12397<br>
&gt; ...<br>
&gt;<br>
&gt; Right?<br>
&gt;<br>
&gt; I feel quite strongly that we should *not* include this sort of<br>
&gt; information<br>
&gt; in the current REFER draft, and work on providing an *authenticated*<br>
&gt; version of this draft at a later date.<br>
&gt;<br>
&gt; /a<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]<br>
&gt; &gt; Sent: Wednesday, April 10, 2002 1:51<br>
&gt; &gt; To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; sip@ietf.org<br>
&gt; &gt; Subject: RE: [Sip] REFER security options - removing Referred-By<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; I agree with what Patrick Mourot says. Let's have a basic<br>
&gt; &gt; refer capability<br>
&gt; &gt; that provides an unsecured identity of A to C, and then leave<br>
&gt; &gt; it to C to<br>
&gt; &gt; decide whether and how to verify the identity of A. This will<br>
&gt; &gt; depend on<br>
&gt; &gt; whether it is attended or unattended transfer - with attended<br>
&gt; &gt; transfer the<br>
&gt; &gt; call A-C will exist and will be identified by the Replaces<br>
&gt; &gt; header. This may<br>
&gt; &gt; be sufficient, but if not, verification can be done in the<br>
&gt; &gt; context of that<br>
&gt; &gt; dialog A-C.<br>
&gt; &gt;<br>
&gt; &gt; &nbsp;---------------------------------------------------------------<br>
&gt; &gt; &nbsp;John Elwell (john.elwell@siemenscomms.co.uk)<br>
&gt; &gt; &nbsp;Siemens Communications Limited,<br>
&gt; &gt; &nbsp;---------------------------------------------------------------<br>
&gt; &gt; &nbsp;Internet communications are not secure and therefore Siemens<br>
&gt; &gt; &nbsp;Communications Limited does not accept legal responsibility for the<br>
&gt; &gt; &nbsp;contents of this message. Any views or opinions presented are solely<br>
&gt; &gt; &nbsp;those of the author and do not necessarily represent those of Siemens<br>
&gt; &gt; &nbsp;Communications Limited unless otherwise specifically stated.<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]<br>
&gt; &gt; Sent: Tuesday, April 09, 2002 5:05 PM<br>
&gt; &gt; To: rsparks@dynamicsoft.com; sip@ietf.org<br>
&gt; &gt; Subject: RE: [Sip] REFER security options - removing Referred-By<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Sirs,<br>
&gt; &gt;<br>
&gt; &gt; I think there are plainty of cases where C :<br>
&gt; &gt; -will not need to know that the call was started by A read in<br>
&gt; &gt; Referred-By<br>
&gt; &gt; -or will trust B (i.e: intranet cases).<br>
&gt; &gt;<br>
&gt; &gt; Protecting Referred-By also means protecting others headers.<br>
&gt; &gt;<br>
&gt; &gt; This means that we should avoid all solutions that are not optional<br>
&gt; &gt; regarding C<br>
&gt; &gt; policy.<br>
&gt; &gt;<br>
&gt; &gt; A VERIFYing approach starting at C is better suited; It<br>
&gt; &gt; allows C to decide<br>
&gt; &gt; to<br>
&gt; &gt; verify or not A identity.<br>
&gt; &gt; If C does NOT want to or does NOT support VERIFY the call<br>
&gt; &gt; flows remains<br>
&gt; &gt; consistent with current behavior.<br>
&gt; &gt;<br>
&gt; &gt; PS: Referred-By is a means for carrying some useful Information.<br>
&gt; &gt;<br>
&gt; &gt; That's my 2 cents.<br>
&gt; &gt;<br>
&gt; &gt; Best Regards,<br>
&gt; &gt;<br>
&gt; &gt; Patrick<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; &gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; &gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt; &gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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To: Michael Thomas <mat@cisco.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Ben Campbell'" <bcampbell@dynamicsoft.com>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>,
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Subject: RE: [Sip] REFER security options - removing Referred-By
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Mike,

none of the headers can be trusted. If we move in a direction in 
which nothing is considered valid and useable unless authenticated, 
SIP will have a long long way to go. 


Frank




Michael Thomas <mat@cisco.com>
Sent by: sip-admin@ietf.org
22-04-2002 05:31 PM

 
        To:     "Rosen, Brian" <Brian.Rosen@marconi.com>
        cc:     "'Ben Campbell'" <bcampbell@dynamicsoft.com>
Tom_Gray@mitel.com
Adam Roach <adam@dynamicsoft.com>
"'Elwell, John'" <John.Elwell@siemenscomms.co.uk>
patrick.mourot@online.fr
Robert Sparks <rsparks@dynamicsoft.com>
sip@ietf.org
(bcc: Frank Derks/HVS/BE/PHILIPS)
        Subject:        RE: [Sip] REFER security options - removing Referred-By
        Classification: 



Rosen, Brian writes:
 > Y'know, sometimes I (personally, not as chair) think we take
 > the security argument just a little too far.
 > 
 > Sure, I can finagle the From, but Referred-By is the header
 > I want to use in my GUI.  I'll attach no more credence to it
 > then I get from FROM unless it's authenticated, but it's damn
 > useful to display to a user.

   So you're saying that it's OK for me to be able
   to just put Kofi Anann's admin's name in the
   Referred-by header, so that I can get through
   his screening so we can chat about my thoughts
   on San Francisco annexing Canada?

   Groovy.

                                     Mike

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<br><font size=2 face="Courier">Mike,</font>
<br>
<br><font size=2 face="Courier">none of the headers can be trusted. If we move in a direction in </font>
<br><font size=2 face="Courier">which nothing is considered valid and useable unless authenticated, </font>
<br><font size=2 face="Courier">SIP will have a long long way to go. <br>
<br>
</font>
<br><font size=2 face="Courier">Frank</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>Michael Thomas &lt;mat@cisco.com&gt;</b></font>
<br><font size=1 face="sans-serif">Sent by: sip-admin@ietf.org</font>
<p><font size=1 face="sans-serif">22-04-2002 05:31 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&quot;Rosen, Brian&quot; &lt;Brian.Rosen@marconi.com&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;&quot;'Ben Campbell'&quot; &lt;bcampbell@dynamicsoft.com&gt;<br>
Tom_Gray@mitel.com<br>
Adam Roach &lt;adam@dynamicsoft.com&gt;<br>
&quot;'Elwell, John'&quot; &lt;John.Elwell@siemenscomms.co.uk&gt;<br>
patrick.mourot@online.fr<br>
Robert Sparks &lt;rsparks@dynamicsoft.com&gt;<br>
sip@ietf.org<br>
(bcc: Frank Derks/HVS/BE/PHILIPS)</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sip] REFER security options - removing Referred-By</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">Rosen, Brian writes:<br>
 &gt; Y'know, sometimes I (personally, not as chair) think we take<br>
 &gt; the security argument just a little too far.<br>
 &gt; <br>
 &gt; Sure, I can finagle the From, but Referred-By is the header<br>
 &gt; I want to use in my GUI. &nbsp;I'll attach no more credence to it<br>
 &gt; then I get from FROM unless it's authenticated, but it's damn<br>
 &gt; useful to display to a user.<br>
<br>
 &nbsp; So you're saying that it's OK for me to be able<br>
 &nbsp; to just put Kofi Anann's admin's name in the<br>
 &nbsp; Referred-by header, so that I can get through<br>
 &nbsp; his screening so we can chat about my thoughts<br>
 &nbsp; on San Francisco annexing Canada?<br>
<br>
 &nbsp; Groovy.<br>
<br>
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Mike<br>
<br>
_______________________________________________<br>
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This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
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</font>
<br>
<br>
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From: "Cullen Jennings" <fluffy@cisco.com>
To: <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 12:35:04 -0700
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I agree with Brian on this, hopefully my phone will interoperate with his. 

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Rosen,
> Brian
> Sent: Monday, April 22, 2002 7:21 AM
> To: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach
> Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> Y'know, sometimes I (personally, not as chair) think we take
> the security argument just a little too far.
> 
> Sure, I can finagle the From, but Referred-By is the header
> I want to use in my GUI.  I'll attach no more credence to it
> then I get from FROM unless it's authenticated, but it's damn
> useful to display to a user.
> 
> We have it implemented, we like it, and I'm keeping it.
> 
> Let's leave it in and define it, like FROM, as informational
> only.
> 
> Or shall we remove FROM on the same basis?
> 
> Brian
> 
> > -----Original Message-----
> > From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> > Sent: Monday, April 22, 2002 9:48 AM
> > To: Tom_Gray@mitel.com; Adam Roach
> > Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> > sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > > Tom_Gray@Mitel.COM
> > > Sent: Friday, April 19, 2002 1:50 PM
> > > To: Adam Roach
> > > Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> > > sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > >
> > >  The information could be useful because C could 
> > authenticate the identity
> > > outside of the SIP protocol if that is felt necesary. In 
> > many cases such
> > > authentication is not required. There can be other means
> > > extra-protocol  of
> > > dealing with abusive callers.
> > >
> > 
> > I'm not sure I see the connection between the last two 
> > sentences? How would
> > one make use of an unauthenticated referred-by in dealing with abusive
> > callers? I would tend to expect that abusive callers would be 
> > very likely
> > spoof refered-by headers if it is allowed.
> > 
> > I can accept the idea of referred-by being useful _if_ the 
> > endpoint has
> > another channel of authentication. I am skeptical about any 
> > real use of it
> > without authentication.
> > 
> > Considering that Robert's original question was not, "is 
> > there a use for an
> > unauthenticated refered-by?" but was instead "Is there use 
> > for refer if it
> > is published without referred-by?" with the understanding that a solid
> > referred-by header would be the subject of future work. From that
> > perspective, I have to agree with Adam's belief that 
> > referred-by should be
> > removed from the current draft.
> > 
> > 
> > >
> > >
> > >
> > >
> > >
> > > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
> > >
> > > Sent by:  sip-admin@ietf.org
> > >
> > >
> > > To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
> > >       "'patrick.mourot@online.fr'" 
> > <patrick.mourot@online.fr>, Robert
> > >       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> > > cc:
> > >
> > > Subject:  RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > Why is such information useful if it's unauthenticated?
> > >
> > > Are you planning to use it to implement policy? If so,
> > > inclusion of such a header only sets a trap for naive
> > > implementors.
> > >
> > > If not, are you envisioning this information will be presented
> > > to users? If that's your goal, you'll want to do something
> > > that works with deployed phones. As a convention (and I'm not
> > > proposing that this be standardized), you could obtain the
> > > desired effect with something like:
> > >
> > > INVITE sip:bob@example.com SIP/2.0
> > > From: "Adam Roach forwarded by Robert Sparks"
> > > <sip:adam@dynamicsoft.com>;tag=12397
> > > ...
> > >
> > > Right?
> > >
> > > I feel quite strongly that we should *not* include this sort of
> > > information
> > > in the current REFER draft, and work on providing an *authenticated*
> > > version of this draft at a later date.
> > >
> > > /a
> > >
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > > > Sent: Wednesday, April 10, 2002 1:51
> > > > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com; 
> > sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > > I agree with what Patrick Mourot says. Let's have a basic
> > > > refer capability
> > > > that provides an unsecured identity of A to C, and then leave
> > > > it to C to
> > > > decide whether and how to verify the identity of A. This will
> > > > depend on
> > > > whether it is attended or unattended transfer - with attended
> > > > transfer the
> > > > call A-C will exist and will be identified by the Replaces
> > > > header. This may
> > > > be sufficient, but if not, verification can be done in the
> > > > context of that
> > > > dialog A-C.
> > > >
> > > >  ---------------------------------------------------------------
> > > >  John Elwell (john.elwell@siemenscomms.co.uk)
> > > >  Siemens Communications Limited,
> > > >  ---------------------------------------------------------------
> > > >  Internet communications are not secure and therefore Siemens
> > > >  Communications Limited does not accept legal 
> > responsibility for the
> > > >  contents of this message. Any views or opinions 
> > presented are solely
> > > >  those of the author and do not necessarily represent 
> > those of Siemens
> > > >  Communications Limited unless otherwise specifically stated.
> > > >
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > > Sent: Tuesday, April 09, 2002 5:05 PM
> > > > To: rsparks@dynamicsoft.com; sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > >
> > > > Sirs,
> > > >
> > > > I think there are plainty of cases where C :
> > > > -will not need to know that the call was started by A read in
> > > > Referred-By
> > > > -or will trust B (i.e: intranet cases).
> > > >
> > > > Protecting Referred-By also means protecting others headers.
> > > >
> > > > This means that we should avoid all solutions that are 
> > not optional
> > > > regarding C
> > > > policy.
> > > >
> > > > A VERIFYing approach starting at C is better suited; It
> > > > allows C to decide
> > > > to
> > > > verify or not A identity.
> > > > If C does NOT want to or does NOT support VERIFY the call
> > > > flows remains
> > > > consistent with current behavior.
> > > >
> > > > PS: Referred-By is a means for carrying some useful Information.
> > > >
> > > > That's my 2 cents.
> > > >
> > > > Best Regards,
> > > >
> > > > Patrick
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 17:59:37 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'"
	 <mat@cisco.com>
Cc: Ben Campbell <bcampbell@dynamicsoft.com>, Tom_Gray@mitel.com,
        Adam Roach <adam@dynamicsoft.com>,
        "'Elwell, John'"
	 <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 17:26:03 -0400
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There's a major difference, though.

In the protocol, there is currently a clear way that you can respond
to me, saying "you claim to be George Bush. Prove it." (c.f. 401 and
407 responses).

There is absolutely no way that we've been able to define where you
could equivalently say, "you claim you were referred by Kofi Annan's
secretary. Prove it."

This is a fundimental problem.

The "From" field has a defined way of be challenged, but the "Referred-By"
does not. Any arguments that these are the same problem are misguided.

/a

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Monday, April 22, 2002 10:43
> To: 'Michael Thomas'
> Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
> 
> 
> Well, since it's okay to put George Bush's name in the FROM header,
> it's got to be okay to put Kofi Anann's admin's name in the 
> Referred-By
> header.  If Mr. Annan accepts the forged FROM, he will accept the
> forged referral.  
> 
> These headers are (presently) un-authenticated.  We don't DO anything
> with them (presently) other than show them in a GUI.  It's useful to
> do that.  I'll accept an authentication mechanism some day, but
> an un-authenticated field is useful.
> 
> Brian
> 
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Monday, April 22, 2002 11:31 AM
> > To: Rosen, Brian
> > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> > 
> > 
> > Rosen, Brian writes:
> >  > Y'know, sometimes I (personally, not as chair) think we take
> >  > the security argument just a little too far.
> >  > 
> >  > Sure, I can finagle the From, but Referred-By is the header
> >  > I want to use in my GUI.  I'll attach no more credence to it
> >  > then I get from FROM unless it's authenticated, but it's damn
> >  > useful to display to a user.
> > 
> >    So you're saying that it's OK for me to be able
> >    to just put Kofi Anann's admin's name in the
> >    Referred-by header, so that I can get through
> >    his screening so we can chat about my thoughts
> >    on San Francisco annexing Canada?
> > 
> >    Groovy.
> > 
> > 		    Mike
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 18:09:09 2002
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From: "Ben Campbell" <bcampbell@dynamicsoft.com>
To: "Adam Roach" <adam@dynamicsoft.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'" <mat@cisco.com>
Cc: <Tom_Gray@mitel.com>, "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        <patrick.mourot@online.fr>, "Robert Sparks" <rsparks@dynamicsoft.com>,
        <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 16:36:35 -0500
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Now as I see it, we have a couple of camps--one is a group of people who
believe referred-by without authentication is still useful, and presumably
not _too_ harmful. A second, perhaps smaller but pretty vocal, group says
that referred-by without some authentication mechanism is dangerous.

So, to return to Robert's root question: We obviously have a lot of
discussion left on referred-by. So, can we separate the problem, move
forward the refer draft _without_ referred-by, and continue working on a
referred-by mechansism we can all agree on? I have not seen anyone argue to
scrap referred-by altogether (although that seems to be a common strawman.)

Or in otherwords, is refer so useless without referred-by that we should
block the draft until we can close this discussion?

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Monday, April 22, 2002 4:26 PM
> To: 'Rosen, Brian'; 'Michael Thomas'
> Cc: Ben Campbell; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
> There's a major difference, though.
>
> In the protocol, there is currently a clear way that you can respond
> to me, saying "you claim to be George Bush. Prove it." (c.f. 401 and
> 407 responses).
>
> There is absolutely no way that we've been able to define where you
> could equivalently say, "you claim you were referred by Kofi Annan's
> secretary. Prove it."
>
> This is a fundimental problem.
>
> The "From" field has a defined way of be challenged, but the
> "Referred-By"
> does not. Any arguments that these are the same problem are misguided.
>
> /a
>
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Monday, April 22, 2002 10:43
> > To: 'Michael Thomas'
> > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > Well, since it's okay to put George Bush's name in the FROM header,
> > it's got to be okay to put Kofi Anann's admin's name in the
> > Referred-By
> > header.  If Mr. Annan accepts the forged FROM, he will accept the
> > forged referral.
> >
> > These headers are (presently) un-authenticated.  We don't DO anything
> > with them (presently) other than show them in a GUI.  It's useful to
> > do that.  I'll accept an authentication mechanism some day, but
> > an un-authenticated field is useful.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Michael Thomas [mailto:mat@cisco.com]
> > > Sent: Monday, April 22, 2002 11:31 AM
> > > To: Rosen, Brian
> > > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> > > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > Rosen, Brian writes:
> > >  > Y'know, sometimes I (personally, not as chair) think we take
> > >  > the security argument just a little too far.
> > >  >
> > >  > Sure, I can finagle the From, but Referred-By is the header
> > >  > I want to use in my GUI.  I'll attach no more credence to it
> > >  > then I get from FROM unless it's authenticated, but it's damn
> > >  > useful to display to a user.
> > >
> > >    So you're saying that it's OK for me to be able
> > >    to just put Kofi Anann's admin's name in the
> > >    Referred-by header, so that I can get through
> > >    his screening so we can chat about my thoughts
> > >    on San Francisco annexing Canada?
> > >
> > >    Groovy.
> > >
> > > 		    Mike
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> >


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 18:09:49 2002
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Cullen Jennings" <fluffy@cisco.com>, <sip@ietf.org>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 14:43:59 -0700
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I seem to have contradicted myself from my first transaction in this thread
to my most recent. Clearly I must be insane.


So lets separate a few issues that got me crossed up.

1) Do we need something like Referred-By at all?

2) What level of authentication security do we needed on it?

3) Do we have to solve the above two items in the REFER draft or can we work
them in a separate draft that comes after REFER.

I think RJS was originally trying to ask about 3 but I got confused and
thought we have moved to discussing 1.



> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Cullen
> Jennings
> Sent: Monday, April 22, 2002 12:35 PM
> To: sip@ietf.org
> Subject: RE: [Sip] REFER security options - removing Referred-By
>
>
>
> I agree with Brian on this, hopefully my phone will interoperate
> with his.
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Rosen,
> > Brian
> > Sent: Monday, April 22, 2002 7:21 AM
> > To: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach
> > Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> > sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > Y'know, sometimes I (personally, not as chair) think we take
> > the security argument just a little too far.
> >
> > Sure, I can finagle the From, but Referred-By is the header
> > I want to use in my GUI.  I'll attach no more credence to it
> > then I get from FROM unless it's authenticated, but it's damn
> > useful to display to a user.
> >
> > We have it implemented, we like it, and I'm keeping it.
> >
> > Let's leave it in and define it, like FROM, as informational
> > only.
> >
> > Or shall we remove FROM on the same basis?
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> > > Sent: Monday, April 22, 2002 9:48 AM
> > > To: Tom_Gray@mitel.com; Adam Roach
> > > Cc: 'Elwell, John'; patrick.mourot@online.fr; Robert Sparks;
> > > sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> > > > Tom_Gray@Mitel.COM
> > > > Sent: Friday, April 19, 2002 1:50 PM
> > > > To: Adam Roach
> > > > Cc: 'Elwell, John'; 'patrick.mourot@online.fr'; Robert Sparks;
> > > > sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > >
> > > >  The information could be useful because C could
> > > authenticate the identity
> > > > outside of the SIP protocol if that is felt necesary. In
> > > many cases such
> > > > authentication is not required. There can be other means
> > > > extra-protocol  of
> > > > dealing with abusive callers.
> > > >
> > >
> > > I'm not sure I see the connection between the last two
> > > sentences? How would
> > > one make use of an unauthenticated referred-by in dealing with abusive
> > > callers? I would tend to expect that abusive callers would be
> > > very likely
> > > spoof refered-by headers if it is allowed.
> > >
> > > I can accept the idea of referred-by being useful _if_ the
> > > endpoint has
> > > another channel of authentication. I am skeptical about any
> > > real use of it
> > > without authentication.
> > >
> > > Considering that Robert's original question was not, "is
> > > there a use for an
> > > unauthenticated refered-by?" but was instead "Is there use
> > > for refer if it
> > > is published without referred-by?" with the understanding that a solid
> > > referred-by header would be the subject of future work. From that
> > > perspective, I have to agree with Adam's belief that
> > > referred-by should be
> > > removed from the current draft.
> > >
> > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Adam Roach <adam@dynamicsoft.com>@ietf.org on 04/19/2002 02:37:38 PM
> > > >
> > > > Sent by:  sip-admin@ietf.org
> > > >
> > > >
> > > > To:   "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
> > > >       "'patrick.mourot@online.fr'"
> > > <patrick.mourot@online.fr>, Robert
> > > >       Sparks    <rsparks@dynamicsoft.com>, sip@ietf.org
> > > > cc:
> > > >
> > > > Subject:  RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > > Why is such information useful if it's unauthenticated?
> > > >
> > > > Are you planning to use it to implement policy? If so,
> > > > inclusion of such a header only sets a trap for naive
> > > > implementors.
> > > >
> > > > If not, are you envisioning this information will be presented
> > > > to users? If that's your goal, you'll want to do something
> > > > that works with deployed phones. As a convention (and I'm not
> > > > proposing that this be standardized), you could obtain the
> > > > desired effect with something like:
> > > >
> > > > INVITE sip:bob@example.com SIP/2.0
> > > > From: "Adam Roach forwarded by Robert Sparks"
> > > > <sip:adam@dynamicsoft.com>;tag=12397
> > > > ...
> > > >
> > > > Right?
> > > >
> > > > I feel quite strongly that we should *not* include this sort of
> > > > information
> > > > in the current REFER draft, and work on providing an *authenticated*
> > > > version of this draft at a later date.
> > > >
> > > > /a
> > > >
> > > > > -----Original Message-----
> > > > > From: Elwell, John [mailto:John.Elwell@siemenscomms.co.uk]
> > > > > Sent: Wednesday, April 10, 2002 1:51
> > > > > To: 'patrick.mourot@online.fr'; rsparks@dynamicsoft.com;
> > > sip@ietf.org
> > > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > > >
> > > > >
> > > > > I agree with what Patrick Mourot says. Let's have a basic
> > > > > refer capability
> > > > > that provides an unsecured identity of A to C, and then leave
> > > > > it to C to
> > > > > decide whether and how to verify the identity of A. This will
> > > > > depend on
> > > > > whether it is attended or unattended transfer - with attended
> > > > > transfer the
> > > > > call A-C will exist and will be identified by the Replaces
> > > > > header. This may
> > > > > be sufficient, but if not, verification can be done in the
> > > > > context of that
> > > > > dialog A-C.
> > > > >
> > > > >  ---------------------------------------------------------------
> > > > >  John Elwell (john.elwell@siemenscomms.co.uk)
> > > > >  Siemens Communications Limited,
> > > > >  ---------------------------------------------------------------
> > > > >  Internet communications are not secure and therefore Siemens
> > > > >  Communications Limited does not accept legal
> > > responsibility for the
> > > > >  contents of this message. Any views or opinions
> > > presented are solely
> > > > >  those of the author and do not necessarily represent
> > > those of Siemens
> > > > >  Communications Limited unless otherwise specifically stated.
> > > > >
> > > > >
> > > > >
> > > > > -----Original Message-----
> > > > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > > > Sent: Tuesday, April 09, 2002 5:05 PM
> > > > > To: rsparks@dynamicsoft.com; sip@ietf.org
> > > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > > >
> > > > >
> > > > >
> > > > > Sirs,
> > > > >
> > > > > I think there are plainty of cases where C :
> > > > > -will not need to know that the call was started by A read in
> > > > > Referred-By
> > > > > -or will trust B (i.e: intranet cases).
> > > > >
> > > > > Protecting Referred-By also means protecting others headers.
> > > > >
> > > > > This means that we should avoid all solutions that are
> > > not optional
> > > > > regarding C
> > > > > policy.
> > > > >
> > > > > A VERIFYing approach starting at C is better suited; It
> > > > > allows C to decide
> > > > > to
> > > > > verify or not A identity.
> > > > > If C does NOT want to or does NOT support VERIFY the call
> > > > > flows remains
> > > > > consistent with current behavior.
> > > > >
> > > > > PS: Referred-By is a means for carrying some useful Information.
> > > > >
> > > > > That's my 2 cents.
> > > > >
> > > > > Best Regards,
> > > > >
> > > > > Patrick
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > > Use sipping@ietf.org for new developments on the
> > > application of sip
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > > Use sipping@ietf.org for new developments on the
> > > application of sip
> > > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the application of sip
> > > >
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the application of sip
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From: Sean Olson <seancolson@yahoo.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
To: Ben Campbell <bcampbell@dynamicsoft.com>,
        Adam Roach <adam@dynamicsoft.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'" <mat@cisco.com>
Cc: Tom_Gray@mitel.com, "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
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Is it conceivable that a future Referred-By:
mechanism would deviate significantly from
what is currently in the draft? What will be
the impact for those who have implemented the
Referred-By: mechanism already?

/sean

=====
Sean Olson <seancolson@yahoo.com>

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Rohan Mahy wrote:

> ---- Original message ----
> >Date: Fri, 19 Apr 2002 09:52:42 -0400
> >From: Flemming Andreasen <fandreas@cisco.com>
> >Subject: Re: [Sip] Proposed plan for Network-Asserted
> ID/Privacy
> >To: Rohan Mahy <rohan@cisco.com>
> >Cc: sip@ietf.org
> >
> >
> >
> >Rohan Mahy wrote:
> >
> >> Hi,
> >>
> >> I'd like to propose a concrete plan to move forward with a
> solution for
> >> network-asserted identity and privacy.  I'm proposing this
> as an
> >> individual (who is sick of reading about this topic on the
> mailing
> >> list).  Below I've listed some of problems that various
> folks were
> >> apparently trying to solve in this space.
> >>
> >> 1.  Provide a Network-Asserted ID
> >>  a.   Within a trusted administrative domain or federation
> of domains
> >>  b.   For interworking with PSTN mechanisms*
> >>  c.   Usable across administrative boundaries
> >>
> >> 2.  Provide capability for Call Trace
> >>  a.   Within a trusted administrative domain or federation
> of domains
> >>  b.   For interworking with PSTN mechanisms*
> >>  c.   Usable across administrative boundaries
> >>
> >> 3.  Provide privacy
> >>  a.   by the user directly withholding information from
> the network
> >>  b.   in the network for user-provided info at the user's
> request
> >>  c.   in the network for user-provided info based on
> network policy
> >>  d.   of network-provided information
> >>
> >> 4.  Allow for a user-provided "hint" for Network-Asserted
> ID
> >>
> >> *(with gateways in the trusted domain or federation)
> >>
> >> Most of the snags and disagreements we have had with the
> existing
> >> Remote-Party-ID draft (draft-ietf-sip-privacy-04) are
> related to problems
> >> 1c, 2c, and 4.  Even a very scaled-down version of this
> draft would allow
> >> for trusted intermediaries and user agents (including
> gateways) to
> >> exchange network asserted identity and provide the
> capability for call
> >> trace within a domain or federation of trusted domains.
> The solution is
> >> as simple as a single header which carries the identity in
> clear text,
> >> which is added by the first proxy that authenticates the
> user, and
> >> removed before it leaves the trust boundary.
> >
> >You need to be more clear. 1 and 2 talk about administrative
> domains and
> >administrative boundaries and now you are talking about
> trust boundaries. How
> >do you define an administrative domain, an administrative
> boundary, and a
> >trust boundary ?
>
> Fair enough.  Let's say for now that a trusted entity is
> trusted not to maliciously modify messages, or
> reveal "private" network asserted ID information outside
> the "trusted" federation.  So, ordinary UAs in "my" service
> provider network are not trusted.
>
> >> This would solve
> >> 1a, 1b, 2a, and 2b.
> >
> >No. The UA is (normally) untrusted but still needs it. If
> you were only
> >talking about the case when privacy was requested, then it
> doesn't address the
> >requirement to provide call trace for anonymous calls. The
> user is obviously
> >untrusted and hence cannot be given the caller-id in clear-
> text. The privacy
> >draft solves this by handing out an encrypted Remote-Party-
> Id which can
> >subsequently be used for call trace (as well as other
> features for that
> >matter, e.g. anonymous call return).
>
> OK, I hope we agree that 1a, 1b, and 2b would by solved by
> the scaled down mechanism.

Sure.

> You could still provide call
> trace (2a) within this network by using accounting
> mechanisms. You will probably say that insuring you have an

> accounting record for each call is a burden.  Also, you do
> lose the ability to do call return with this mechanism, but I
> do not consider this a short-term requirement.  (You might
> disagree).

Indeed I do disagree. From the very first version of this
"forbidden-name" draft, the motivation section stated the need for
providing CLIP/CLIR services while supporting call trace. SIP itself has
adopted the idea of pushing state to the endpoints for later retrieval
and we even have a more general work item building on this in the form
of state/cookies. Being able to send down an opague URI that can be used
as the target for a subsequent call trace still seems like a pretty good
idea to me. What is the problem with this and what alternative do you
suggest for the UA to invoke the trace ?


>
>
> ><snip>
> >
> >>
> >> Finally, we have problem 4.  Some folks have proposed that
> >> UAs should provide a hint using the same mechanism used to
> >> communicate network asserted identity among trusted
> entities.
> >
> >> Others propose that authentication servers that create a
> >
> >> network asserted identity accept multiple "usernames"
> during
> >
> >> authentication and use this as the hint.  AFAIK these are
> the
> >> only two choices, and we can come up with an answer to this
> >> question in a separate thread.  The answer does not
> fundamentally
> >> change any of the mechanisms we select to solve the other
> problems.
> >
> >Well it certainly impacts it. If you go with the first
> solution then you also
> >accept that Remote-Party-Id can be provided by an untrusted
> entity, so I don't
> >think we can solve these independently.
>
> i think we need to decouple them now, and can join them again
> if it makes sense.
>

All right - I suppose we can always define a duplicate header to make
the purists feel good.


>
> >> So, in summary, I'm proposing:
> >>
> >> No additional work is required to implement 3c
> >>
> >> Three short term deliverables
> >> I) adopt draft-peterson-sip-privacy-longterm as a WG item
> to solve
> >>    3a and 3b
> >>
> >
> >It's unclear that 3b is independent of which mechanism you
> choose to provide
> >the hint above.
>
> I don't follow you.

If you were to use Remote-Party-Id for the user-provided info, and
Remote-Party-Id for network-asserted identity has a way of indicating
privacy, then it would be rather silly to now require a completely
different mechanism to provide privacy for this user-provided
Remote-Party-Id.

>
> >> II) write an extension to I) as a WG item for requesting
> privacy of
> >> network asserted ID (however it is expressed) to support
> 3d.  This would
> >> get used by III and IV.
> >>

Again, I don't think it is desirable to require a multitude of
extensions to solve a single problem. The "forbidden-name" draft as
defined today already solves this for network-asserted identity and I
don't see why I can't continue doing that.


>
> >> III) pare down the mechanism in draft-ietf-sip-privacy to
> only address 1a,
> >> 1b, 2a, and 2b.  quickly get closure on which mechanism we
> use for 4, and
> >> include it in the draft.  keep the current applicability
> statement.
> >
> >As mentioned above, I'm not clear on exactly what 1a, 1b, 2a
> and 2b are, but
> >if the "network asserted identity" draft would neither
> address privacy for
> >network asserted identity, nor call trace for anonymous
> calls, then I
> >certainly disagree with this.
>
> I think it does provide a solution here for a small scope.

Too small IMHO. It's basically useless without additional extensions.
Paraphrasing Jon here, it would seem that too much has now been taken
away.

-- Flemming



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From daemon@ns.ietf.org  Mon Apr 22 19:12:55 2002
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From: "Victor Paulsamy" <victor@zapex.com>
To: <sip@ietf.org>
Cc: "Victor Paulsamy" <victor@zapex.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 15:47:13 -0700
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Take an example of full-mesh conferencing. In that, A and B are talking.
Now, C calls B. B decides to confer with C & A, hence sends REFER to, say, C
with "Refer-To:A". To C, "Referred-By:B" is useful because, it signals
"conference" mode.

Regards,

--victor

Victor Paulsamy               E-mail: victor@zapex.com
Senior Software Engineer      Phone : 650.930.1339 (Direct)
Zapex Technologies, Inc.      Phone : 650.930.1300 (Main)
Mountain View, CA 94043       Fax   : 650.930.1399


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To: "Victor Paulsamy" <victor@zapex.com>, <sip@ietf.org>
From: AC Mahendran <mahendra@qualcomm.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
Cc: "Victor Paulsamy" <victor@zapex.com>
In-Reply-To: <NEBBIPNANPGLCDPGFJGPIEDOCEAA.victor@zapex.com>
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B  can add C into the conference without REFERing it to A.

Anyway, I support the idea of leaving the specification as is (i.e without 
removing the Referred-By header). However, we should definitely work 
(later) on providing extensions for authentication of the Referred-By UA.

thanks,
-AC

At 03:47 PM 4/22/2002 -0700, Victor Paulsamy wrote:
>Take an example of full-mesh conferencing. In that, A and B are talking.
>Now, C calls B. B decides to confer with C & A, hence sends REFER to, say, C
>with "Refer-To:A". To C, "Referred-By:B" is useful because, it signals
>"conference" mode.
>
>Regards,
>
>--victor
>
>Victor Paulsamy               E-mail: victor@zapex.com
>Senior Software Engineer      Phone : 650.930.1339 (Direct)
>Zapex Technologies, Inc.      Phone : 650.930.1300 (Main)
>Mountain View, CA 94043       Fax   : 650.930.1399
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 22 19:47:08 2002
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From: "Victor Paulsamy" <victor@zapex.com>
To: "AC Mahendran" <mahendra@qualcomm.com>, <sip@ietf.org>
Cc: "Victor Paulsamy" <victor@zapex.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
Date: Mon, 22 Apr 2002 16:21:54 -0700
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> B  can add C into the conference without REFERing it to A.
>
True. However, I'm talking about full-mesh conference, where each entity is
maintaining both sip and rtp relations with each other.

--victor

> Anyway, I support the idea of leaving the specification as is
> (i.e without
> removing the Referred-By header). However, we should definitely work
> (later) on providing extensions for authentication of the Referred-By UA.
>
> thanks,
> -AC
>
> At 03:47 PM 4/22/2002 -0700, Victor Paulsamy wrote:
> >Take an example of full-mesh conferencing. In that, A and B are talking.
> >Now, C calls B. B decides to confer with C & A, hence sends
> REFER to, say, C
> >with "Refer-To:A". To C, "Referred-By:B" is useful because, it signals
> >"conference" mode.
> >
> >Regards,
> >
> >--victor
> >
> >Victor Paulsamy               E-mail: victor@zapex.com
> >Senior Software Engineer      Phone : 650.930.1339 (Direct)
> >Zapex Technologies, Inc.      Phone : 650.930.1300 (Main)
> >Mountain View, CA 94043       Fax   : 650.930.1399
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
>
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 23 02:50:14 2002
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Subject: Conferencing semantics : was RE: [Sip] REFER security options -
	removing Referred-By
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Victor Paulsamy <victor@zapex.com>
Cc: AC Mahendran <mahendra@qualcomm.com>, sip@ietf.org
In-Reply-To: <NEBBIPNANPGLCDPGFJGPKEDPCEAA.victor@zapex.com>
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Referred-By should not be used to make that association.
Look at the discussion of Join in draft-sipping-cc-framework-00.

RjS

On Tue, 2002-04-23 at 01:21, Victor Paulsamy wrote:
> > B  can add C into the conference without REFERing it to A.
> >
> True. However, I'm talking about full-mesh conference, where each entity is
> maintaining both sip and rtp relations with each other.
> 
> --victor
> 
> > Anyway, I support the idea of leaving the specification as is
> > (i.e without
> > removing the Referred-By header). However, we should definitely work
> > (later) on providing extensions for authentication of the Referred-By UA.
> >
> > thanks,
> > -AC
> >
> > At 03:47 PM 4/22/2002 -0700, Victor Paulsamy wrote:
> > >Take an example of full-mesh conferencing. In that, A and B are talking.
> > >Now, C calls B. B decides to confer with C & A, hence sends
> > REFER to, say, C
> > >with "Refer-To:A". To C, "Referred-By:B" is useful because, it signals
> > >"conference" mode.
> > >
> > >Regards,
> > >
> > >--victor
> > >
> > >Victor Paulsamy               E-mail: victor@zapex.com
> > >Senior Software Engineer      Phone : 650.930.1339 (Direct)
> > >Zapex Technologies, Inc.      Phone : 650.930.1300 (Main)
> > >Mountain View, CA 94043       Fax   : 650.930.1399
> > >
> > >
> > >_______________________________________________
> > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >This list is for NEW development of the core SIP Protocol
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sipping@ietf.org for new developments on the application of sip
> >
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 23 03:13:17 2002
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To: Ben Campbell <bcampbell@dynamicsoft.com>
Subject: RE: [Sip] REFER security options - removing Referred-By
Message-ID: <1019544391.3cc50347ba177@imp.free.fr>
Date: Tue, 23 Apr 2002 08:46:31 +0200 (MEST)
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Michael Thomas'" <mat@cisco.com>, Tom_Gray@mitel.com,
        "'Elwell, John'" <John.Elwell@siemenscomms.co.uk>,
        patrick.mourot@online.fr, Robert Sparks <rsparks@dynamicsoft.com>,
        sip@ietf.org
References: <HNEOJECGFHIABDLENMMCEEMBCGAA.bcampbell@dynamicsoft.com>
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Ben,

In order to take into account :
-existing implementations,
-situations where authentication is not a problem,
-"real-life" constraints : CPU & Memory consumption and infrastructure needs
-rules applied for other SIP headers/fields (FROM, ...)
-opinions expressed in that thread,
-...

I think it is fair to let Referred-by in the draft and propose _LATER_ an  
OPTIONAL true authentication mechanism.

I hope this view will satisfy every body over there.

Best Regards,

Patrick


Quoting Ben Campbell <bcampbell@dynamicsoft.com>:

> Now as I see it, we have a couple of camps--one is a group of people
> who
> believe referred-by without authentication is still useful, and
> presumably
> not _too_ harmful. A second, perhaps smaller but pretty vocal, group
> says
> that referred-by without some authentication mechanism is dangerous.
> 
> So, to return to Robert's root question: We obviously have a lot of
> discussion left on referred-by. So, can we separate the problem, move
> forward the refer draft _without_ referred-by, and continue working on
> a
> referred-by mechansism we can all agree on? I have not seen anyone argue
> to
> scrap referred-by altogether (although that seems to be a common
> strawman.)
> 
> Or in otherwords, is refer so useless without referred-by that we
> should
> block the draft until we can close this discussion?
> 
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > Sent: Monday, April 22, 2002 4:26 PM
> > To: 'Rosen, Brian'; 'Michael Thomas'
> > Cc: Ben Campbell; Tom_Gray@mitel.com; Adam Roach; 'Elwell, John';
> > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > Subject: RE: [Sip] REFER security options - removing Referred-By
> >
> >
> > There's a major difference, though.
> >
> > In the protocol, there is currently a clear way that you can respond
> > to me, saying "you claim to be George Bush. Prove it." (c.f. 401 and
> > 407 responses).
> >
> > There is absolutely no way that we've been able to define where you
> > could equivalently say, "you claim you were referred by Kofi Annan's
> > secretary. Prove it."
> >
> > This is a fundimental problem.
> >
> > The "From" field has a defined way of be challenged, but the
> > "Referred-By"
> > does not. Any arguments that these are the same problem are
> misguided.
> >
> > /a
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: Monday, April 22, 2002 10:43
> > > To: 'Michael Thomas'
> > > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell,
> John';
> > > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > >
> > >
> > > Well, since it's okay to put George Bush's name in the FROM
> header,
> > > it's got to be okay to put Kofi Anann's admin's name in the
> > > Referred-By
> > > header.  If Mr. Annan accepts the forged FROM, he will accept the
> > > forged referral.
> > >
> > > These headers are (presently) un-authenticated.  We don't DO
> anything
> > > with them (presently) other than show them in a GUI.  It's useful
> to
> > > do that.  I'll accept an authentication mechanism some day, but
> > > an un-authenticated field is useful.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Michael Thomas [mailto:mat@cisco.com]
> > > > Sent: Monday, April 22, 2002 11:31 AM
> > > > To: Rosen, Brian
> > > > Cc: 'Ben Campbell'; Tom_Gray@mitel.com; Adam Roach; 'Elwell,
> John';
> > > > patrick.mourot@online.fr; Robert Sparks; sip@ietf.org
> > > > Subject: RE: [Sip] REFER security options - removing Referred-By
> > > >
> > > >
> > > > Rosen, Brian writes:
> > > >  > Y'know, sometimes I (personally, not as chair) think we take
> > > >  > the security argument just a little too far.
> > > >  >
> > > >  > Sure, I can finagle the From, but Referred-By is the header
> > > >  > I want to use in my GUI.  I'll attach no more credence to it
> > > >  > then I get from FROM unless it's authenticated, but it's damn
> > > >  > useful to display to a user.
> > > >
> > > >    So you're saying that it's OK for me to be able
> > > >    to just put Kofi Anann's admin's name in the
> > > >    Referred-by header, so that I can get through
> > > >    his screening so we can chat about my thoughts
> > > >    on San Francisco annexing Canada?
> > > >
> > > >    Groovy.
> > > >
> > > > 		    Mike
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current
> sip
> > > > Use sipping@ietf.org for new developments on the application of
> sip
> > > >
> > >
> 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Tue, 23 Apr 2002 10:28:37 +0300
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Thread-Topic: [Sip] Revised Service Route Discovery Draft
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To: <sip@ietf.org>, <3GPP_TSG_CN_WG1@list.etsi.fr>
Cc: <dwillis@dynamicsoft.com>
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Hi AC!

> -----Original Message-----
> From: ext AC Mahendran [mailto:mahendra@qualcomm.com]
> Sent: 22 April, 2002 19:49
> To: Dean Willis; sip@ietf.org
> Cc: 3GPP_TSG_CN_WG1
> Subject: Re: [Sip] Revised Service Route Discovery Draft
> 
> Dean:
> 
> The Record-Route headers in section 5.6 (in transactions F3 & 
> F5) are in the wrong order.

Well spotted! And the same thing happened to all the Via headers...
Will be corrected in the next update. Thanks for pointing this out.

> Also, doesn't the "Path" header provide the same functionality as the 
> "P-Service-Route" ?

In IETF SIP WG is was impossible to reach consensus on using
Path for both - originated and terminated - session cases.
Thus, the Path header will be used for terminated session
cases only, whereas the P-Service-Route header only affects
the originated session cases.

cheers,
 Bernie 


> thanks,
> AC
> 
> 
> At 06:15 PM 4/19/2002 -0500, Dean Willis wrote:
> 
> >I've submitted draft-willis-sip-svcrtdisco-01.txt to the 
> Internet Drafts
> >queue. This version includs refinements made by Bernie 
> Hoeneisen to my
> >earlier draft.
> >
> >The draft should be announced shortly on ietf-announce.
> >
> >In the meantime, please review:
> >
> >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtd
isco-01.txt
>
>or
>
>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.htm
>l
>
>or
>
>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.xml
>
>
>This is a "P-header" draft addressing 3GPP's required header for
>reporting a service proxy assignment in the REGISTER response.
>
>
>Please review ASAP. We need to get this to IETF last call by the end of
>the month if at all possible.
>
>--
>Dean
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Tue Apr 23 05:15:08 2002
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From: "Vesa Torvinen (LMF)" <Vesa.Torvinen@lmf.ericsson.se>
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Subject: [Sip] new version: Security Mechanism Agreement for SIP Sessions
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Hi, 

We have submitted a new version of SIP Security Agreement draft
(ex draft-arkko-sip-sec-agree-01, now draft-ietf-sip-sec-agree-00). 

We have changed quite many details in the draft (including the 
syntax, usage of option tags, etc), and added more text on behaviors 
expected from SIP entities using the extension. We hope this version 
will now better fulfill the needs of both IETF and 3GPP. 

If you want to see the draft before it is officially available, go to: 

http://standards.ericsson.net/sip/drafts/draft-ietf-sip-sec-agree-00.txt

The abstract of the draft: 

SIP has a number of security mechanisms for hop-by-hop and end-to-end
protection. Some of the security mechanisms have been built in to the
SIP protocol, such as HTTP authentication or secure attachments. In
these mechanisms there are even alternative algorithms and parameters.
Currently it isn't possible to select which security mechanisms to use
over a connection. In particular, even if some mechanisms such as
OPTIONS were used to make this selection, the selection would be vul
nerable against the Bidding-Down attack.  This document defines a
header for negotiating the security mechanisms within SIP. A SIP
entity applying this mechanism must always require some minimum secu
rity (i.e. integrity protection) from all communicating parties in
order to secure the negotiation, but the negotiation can agree on
which specific minimum security is used.

All comments are most welcome! 

Vesa 


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From daemon@optimus.ietf.org  Tue Apr 23 06:43:15 2002
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Date: Tue, 23 Apr 2002 13:10:43 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: sip <sip@ietf.org>, Dave Oran <oran@cisco.com>,
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Subject: [Sip] Reason draft
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Hi,

as previously suggested in the mailing list, I have removed the reason
specific codes from the Reason draft. Until it appears in the archives,
you can fetch it from:

http://www.cs.columbia.edu/~gonzalo/draft-ietf-sip-reason-00.txt

I would appreciate comments on the new format.

Thanks,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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Subject: [Sip] I-D ACTION:draft-willis-sip-scvrtdisco-01.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension Header for Service Route Discovery in 
                          Private Networks
	Author(s)	: D. Willis, B. Hoeneisen
	Filename	: draft-willis-sip-scvrtdisco-01.txt
	Pages		: 13
	Date		: 22-Apr-02
	
This document proposes a private SIP extension header used in
conjunction with responses to REGISTER messages to provide a
mechanism by which a registrar may inform a registering UA of a
service route that the UA may use to request outbound services from
the registrar's domain.

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From daemon@ns.ietf.org  Tue Apr 23 09:50:53 2002
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From: "Konstantinos Katrinis" <katrinis@tik.ee.ethz.ch>
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Subject: [Sip] Comments on the draft-wu-sipping-floor-control-00.txt
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	Dear all,

referring to the draft "Use of SIP and SOAP for conference floor control", I
've been asking myself whether it is handy to use SOAP as the language to
exchange floor control related information(commands/events).My alternative
would be to define additional SIP method(s) and use some text-based
representation language in the payload for the floor control
information(extended SDP).
	My argument is that a ready made (XML) parser would be too much for the
user agent (regarding performance).Please express your against arguments, if
I am missing any.

Thanks in advance,
Kostas Katrinis
ETH Zuerich




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From daemon@ns.ietf.org  Tue Apr 23 10:28:20 2002
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Date: Tue, 23 Apr 2002 09:47:36 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: katrinis@tik.ee.ethz.ch
CC: sip@ietf.org
Subject: Re: [Sip] Comments on the draft-wu-sipping-floor-control-00.txt
References: <KHEOKFKAADGKAFJIDEOFEELFCBAA.katrinis@tik.ee.ethz.ch>
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User agents that support presence will already need to support XML.
Maybe in the future, they'll need it for SDPng. There are a number of
light-weight XML parsers that do not consume the resources that, say,
Xerces does. Generation of XML is even easier, since you can do that via
sprintf, with a locally-stored template. I doubt that having half a
dozen different text formats for different types of data is particularly
helpful in terms of overall complexity and correctness.

Konstantinos Katrinis wrote:
> 
>         Dear all,
> 
> referring to the draft "Use of SIP and SOAP for conference floor control", I
> 've been asking myself whether it is handy to use SOAP as the language to
> exchange floor control related information(commands/events).My alternative
> would be to define additional SIP method(s) and use some text-based
> representation language in the payload for the floor control
> information(extended SDP).
>         My argument is that a ready made (XML) parser would be too much for the
> user agent (regarding performance).Please express your against arguments, if
> I am missing any.
> 
> Thanks in advance,
> Kostas Katrinis
> ETH Zuerich
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From daemon@ns.ietf.org  Tue Apr 23 17:18:53 2002
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Date: Tue, 23 Apr 2002 13:39:43 -0700
To: bernhard.honeisen@nokia.com
From: AC Mahendran <mahendra@qualcomm.com>
Subject: RE: [Sip] Revised Service Route Discovery Draft
Cc: sip@ietf.org
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Bernhard:

Thanks for the clarification. I have a few more questions regarding the draft.

1) You have mentioned that P-Service-Route header is optional in REGISTER 
requests; How will the header, if present, be interpreted by the Registrar ?
2) How should the UAC interpret the order of proxies in the P-Service-Route 
header ? For eg: The UAC inverts the order of proxies in the Record-Route 
header to form a route set.
3) Also, you have mentioned that the UAC should add the P-Service-Route 
header information to the existing route set to form a new Route header 
value, which needs to be included in all origination requests. Should this 
header information be added to the beginning or the end of the route set ?
4) Should the UAC over write its existing P-Service-Route value if if gets 
a new value for P-Service-Route in 2XX response ?

It would be great if you can clarify all this information in the draft too.

thanks,
AC

At 10:28 AM 4/23/2002 +0300, you wrote:
>Hi AC!
>
> > -----Original Message-----
> > From: ext AC Mahendran [mailto:mahendra@qualcomm.com]
> > Sent: 22 April, 2002 19:49
> > To: Dean Willis; sip@ietf.org
> > Cc: 3GPP_TSG_CN_WG1
> > Subject: Re: [Sip] Revised Service Route Discovery Draft
> >
> > Dean:
> >
> > The Record-Route headers in section 5.6 (in transactions F3 &
> > F5) are in the wrong order.
>
>Well spotted! And the same thing happened to all the Via headers...
>Will be corrected in the next update. Thanks for pointing this out.
>
> > Also, doesn't the "Path" header provide the same functionality as the
> > "P-Service-Route" ?
>
>In IETF SIP WG is was impossible to reach consensus on using
>Path for both - originated and terminated - session cases.
>Thus, the Path header will be used for terminated session
>cases only, whereas the P-Service-Route header only affects
>the originated session cases.
>
>cheers,
>  Bernie
>
>
> > thanks,
> > AC
> >
> >
> > At 06:15 PM 4/19/2002 -0500, Dean Willis wrote:
> >
> > >I've submitted draft-willis-sip-svcrtdisco-01.txt to the
> > Internet Drafts
> > >queue. This version includs refinements made by Bernie
> > Hoeneisen to my
> > >earlier draft.
> > >
> > >The draft should be announced shortly on ietf-announce.
> > >
> > >In the meantime, please review:
> > >
> > >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtd
>isco-01.txt
> >
> >or
> >
> >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.htm
> >l
> >
> >or
> >
> >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.xml
> >
> >
> >This is a "P-header" draft addressing 3GPP's required header for
> >reporting a service proxy assignment in the REGISTER response.
> >
> >
> >Please review ASAP. We need to get this to IETF last call by the end of
> >the month if at all possible.
> >
> >--
> >Dean
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
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From daemon@ns.ietf.org  Tue Apr 23 17:43:13 2002
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Date: Tue, 23 Apr 2002 14:10:02 -0700
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
From: AC Mahendran <mahendra@qualcomm.com>
Subject: Re: [Sip] Reason draft
Cc: sip@ietf.org
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Gonzalo:

Section 2 of the draft states the following:
"A SIP message MAY contain more than one Reason values (i.e., multiple 
Reason lines), but all of them MUST have different protocol values (e.g., 
one SIP and another Q.850)"

Why is the restriction that the Reason values MUST have multiple protocol 
values ? For example, What if I want to provide multiple reasons (using the 
same protocol) as to why I am CANCELing a particular request ?

I don't see any need for having the restriction.

thanks,
AC

At 01:10 PM 4/23/2002 +0300, Gonzalo Camarillo wrote:
>Hi,
>
>as previously suggested in the mailing list, I have removed the reason
>specific codes from the Reason draft. Until it appears in the archives,
>you can fetch it from:
>
>http://www.cs.columbia.edu/~gonzalo/draft-ietf-sip-reason-00.txt
>
>I would appreciate comments on the new format.
>
>Thanks,
>
>Gonzalo
>--
>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>Telecom R&D               Fax   :  +358  9 299 30 52
>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>Finland                   http://www.hut.fi/~gonzalo
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From daemon@ns.ietf.org  Tue Apr 23 18:02:19 2002
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From: "Hussam Jarada" <hussam.jarada@ulticomdal.com>
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Hi,

In draft-ietf-sipping-call-flows-00.txt, there's many sections like section
2.1.1 and section 2.1.2 were the To header field is missing tag parameter,
is this a miss typed error ?, I thought per bis-0.9 SIP spec that the tag
parameter must be added to final responses as well provisional responses
except 100 (which may).

	Thanks,

Hussam Jarada
Ulticom Inc.





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Subject: [Sip] I-D ACTION:draft-willis-sip-path-03.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension for Registering Non-Adjacent Contacts
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-path-03.txt
	Pages		: 11
	Date		: 15-Apr-02
	
The REGISTER function is used in a SIP system primarily to associate
a temporary contact address with an address-of-record.  This contact
is generally in the form of a URI, such as Contact:
&ltsip:alice@pc33.atlanta.com> and is generally dynamic and
associated with the IP address or hostname of the SIP UA.  The
problem is that network topology may be that there are one or more
SIP proxies between the UA and the registrar, such that any message
from the user's home network to the registered UA must traverse these
proxies.  The REGISTER method itself does not give us a mechanism to
discover and record this sequence of proxies in the registrar for
future use.  This document defines an extension header, 'Path' which
provides such a mechanism.

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From daemon@optimus.ietf.org  Wed Apr 24 01:50:27 2002
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Date: Wed, 24 Apr 2002 08:19:34 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: AC Mahendran <mahendra@qualcomm.com>
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Subject: Re: [Sip] Reason draft
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Hello,

First of all, I need to fix the BNF in order to allow multiple Reason
lines. It will be fixed in the next release.

About your comment, which scenario do you have in mind for needing two
Reason codes using the same protocol? Could you provide us with a
particular example?

Thanks,

Gonzalo

AC Mahendran wrote:
> 
> Gonzalo:
> 
> Section 2 of the draft states the following:
> "A SIP message MAY contain more than one Reason values (i.e., multiple
> Reason lines), but all of them MUST have different protocol values (e.g.,
> one SIP and another Q.850)"
> 
> Why is the restriction that the Reason values MUST have multiple protocol
> values ? For example, What if I want to provide multiple reasons (using the
> same protocol) as to why I am CANCELing a particular request ?
> 
> I don't see any need for having the restriction.
> 
> thanks,
> AC
> 
> At 01:10 PM 4/23/2002 +0300, Gonzalo Camarillo wrote:
> >Hi,
> >
> >as previously suggested in the mailing list, I have removed the reason
> >specific codes from the Reason draft. Until it appears in the archives,
> >you can fetch it from:
> >
> >http://www.cs.columbia.edu/~gonzalo/draft-ietf-sip-reason-00.txt
> >
> >I would appreciate comments on the new format.
> >
> >Thanks,
> >
> >Gonzalo
> >--
> >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> >Telecom R&D               Fax   :  +358  9 299 30 52
> >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> >Finland                   http://www.hut.fi/~gonzalo
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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From daemon@optimus.ietf.org  Wed Apr 24 04:40:37 2002
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Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Wed, 24 Apr 2002 11:17:06 +0300
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Hi AC!

Thanks for your feedback, and pointing out these issues!
We certainly will consider those for the next version.

More comments inline.

cheers,
 Bernie

> -----Original Message-----
> From: ext AC Mahendran [mailto:mahendra@qualcomm.com]
> Sent: 23 April, 2002 23:40
> To: Honeisen Bernhard (NET/Helsinki)
> Cc: sip@ietf.org
> Subject: RE: [Sip] Revised Service Route Discovery Draft
> 
> 
> 
> Bernhard:
> 
> Thanks for the clarification. I have a few more questions 
> regarding the draft.
> 
> 1) You have mentioned that P-Service-Route header is optional 
> in REGISTER requests; How will the header, if present,
> be interpreted by the Registrar ?

This is wrong in the draft (a typical copy-paste-modify
mistake...;-| ). Will be fixed in the next version.

> 2) How should the UAC interpret the order of proxies in the 
> P-Service-Route  header ? For eg: The UAC inverts the order
> of proxies in the Record-Route header to form a route set.

I guess you mean here the case where more than one
P-Service-Route header entry is present, and how the order
will be in the preloaded Route.

This issue is already in the process of beeing fixed.

> 3) Also, you have mentioned that the UAC should add the
> P-Service-Route header information to the existing route
> set to form a new Route header value, which needs to be
> included in all origination requests. Should this 
> header information be added to the beginning or the end of 
> the route set ?

In section 5.1. (Procedures at the UA) it reads:
   "The UA MAY choose to exercise a service route for future messages
   associated with a given address-of-record for which a service route
   is known.  If so, it appends the given service route to any local
   required Route headers, and uses the result as a preloaded Route
   header in outgoing messages."!

The order for constructing the route-set in case there
are local Routes AND P-Service-Route should be more clarified.

> 4) Should the UAC over write its existing P-Service-Route 
> value if if gets a new value for P-Service-Route in 2XX response ?

I guess you are talking about re-registration.
The idea is, that the P-Service-Route of latest 200 OK
(for REGISTER) will be used. Will be clarified in the next version.

> It would be great if you can clarify all this information in 
> the draft too.

I agree...

> thanks,
> AC
> 
> At 10:28 AM 4/23/2002 +0300, you wrote:
> >Hi AC!
> >
> > > -----Original Message-----
> > > From: ext AC Mahendran [mailto:mahendra@qualcomm.com]
> > > Sent: 22 April, 2002 19:49
> > > To: Dean Willis; sip@ietf.org
> > > Cc: 3GPP_TSG_CN_WG1
> > > Subject: Re: [Sip] Revised Service Route Discovery Draft
> > >
> > > Dean:
> > >
> > > The Record-Route headers in section 5.6 (in transactions F3 &
> > > F5) are in the wrong order.
> >
> >Well spotted! And the same thing happened to all the Via headers...
> >Will be corrected in the next update. Thanks for pointing this out.
> >
> > > Also, doesn't the "Path" header provide the same 
> functionality as the
> > > "P-Service-Route" ?
> >
> >In IETF SIP WG is was impossible to reach consensus on using
> >Path for both - originated and terminated - session cases.
> >Thus, the Path header will be used for terminated session
> >cases only, whereas the P-Service-Route header only affects
> >the originated session cases.
> >
> >cheers,
> >  Bernie
> >
> >
> > > thanks,
> > > AC
> > >
> > >
> > > At 06:15 PM 4/19/2002 -0500, Dean Willis wrote:
> > >
> > > >I've submitted draft-willis-sip-svcrtdisco-01.txt to the
> > > Internet Drafts
> > > >queue. This version includs refinements made by Bernie
> > > Hoeneisen to my
> > > >earlier draft.
> > > >
> > > >The draft should be announced shortly on ietf-announce.
> > > >
> > > >In the meantime, please review:
> > > >
> > > >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtd
> >isco-01.txt
> > >
> > >or
> > >
> > 
> >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtd
isco-01.htm
> >l
> >
> >or
> >
> >http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-01.xml
> >
> >
> >This is a "P-header" draft addressing 3GPP's required header for
> >reporting a service proxy assignment in the REGISTER response.
> >
> >
> >Please review ASAP. We need to get this to IETF last call by the end of
> >the month if at all possible.
> >
> >--
> >Dean
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Wed Apr 24 07:02:59 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: "'AC Mahendran'" <mahendra@qualcomm.com>, <sip@ietf.org>
Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Wed, 24 Apr 2002 05:33:17 -0500
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Well, the return code of the Path MAY be similar to the value of
P-Service-Route. It might not. The Path is an exact subset of the
traversal path of the REGISTER request. The P-Service-Route header could
return something completely disjoint -- it just requires that the
registrar have knowledge of some serving proxy that it is "recommending"
to the

WRT the RR headers -- I think the spec is backwards. But that's just me
. . .

--
Dean

AC wrote:
> The Record-Route headers in section 5.6 (in transactions F3 &
> F5) are in 
> the wrong order.
> 
> Also, doesn't the "Path" header provide the same functionality as the
> "P-Service-Route" ?
> 


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From daemon@optimus.ietf.org  Wed Apr 24 08:17:49 2002
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From: "Steffen Fries" <steffen.fries@mchp.siemens.de>
Organization: Siemens AG
To: sip@ietf.org
Date: Wed, 24 Apr 2002 13:51:05 +0200
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Hi,

I've got a question to the provision of message integrity, 
which might be scenario specific.

SIP itself does not provide message integrity by itself. 
Instead message body integrity may be achieved by using HTTP 
Digest Authentication. This is not sufficient for many 
scenarios. Especially the security for the last hop should have 
an integrity protection that covers the complete message.

For the 3GPP networks there was a decision to have integrity 
for the last hop by using IPSec in favour of the SIP Digest 
Authentication. But for wireline networks the usage of an 
additional security protocol might not always be the best 
choice.

During IETF53 I've got the impression that people rather tend 
to go with TLS to achieve message integrity than using an 
extension to SIP. A possible extension to be used would be the 
SIP Digest Authentication (draft-undery-sip-auth-00.txt), which 
was also discussed elaborately during the last meeting. 

Did I get a wrong impression or is the TLS way really 
sufficient for most of the cases? 

Regards
	Steffen

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Steffen Fries writes:
 > Did I get a wrong impression or is the TLS way really 
 > sufficient for most of the cases? 

   If you run SIP over UDP, TLS is useless and IPsec
   is a better choice. Also: the digest schemes don't
   address confidentiality.

	Mike

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From: "Steffen Fries" <steffen.fries@mchp.siemens.de>
Organization: Siemens AG
To: Michael Thomas <mat@cisco.com>
Date: Wed, 24 Apr 2002 14:24:13 +0200
MIME-Version: 1.0
Subject: Re: [Sip] SIP message integrity
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>    If you run SIP over UDP, TLS is useless and IPsec
>    is a better choice. Also: the digest schemes don't
>    address confidentiality.
That is true, but is confidentiallity especially for the first 
hop necessary? Imagine a scenario, where you have a SIP 
telephony service provider and a small size company uses this 
service. The company will probably have a firewall deployed. If 
you use IPSec or TLS to protect the link between the user in 
the company and the telephony service provider, the user will 
not be able to make a phone call, since the port information of 
the RTP communication is encrypted and the firewall does not 
possess the appropriate key.

Steffen
___________________________________________________________
Steffen Fries,     Siemens AG, CT IC 3	
Otto-Hahn-Ring 6,  D-81730 Munich, Germany 
Phone:  (+49) 89 / 636-53403, Fax  :  (+49) 89 / 636-48000
Email:  Steffen.Fries@mchp.siemens.de
___________________________________________________________
    This email was written on 100% recycled bytes!



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Cc: Michael Thomas <mat@cisco.com>, sip@ietf.org
Subject: Re: [Sip] SIP message integrity
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Steffen Fries writes:
 > >    If you run SIP over UDP, TLS is useless and IPsec
 > >    is a better choice. Also: the digest schemes don't
 > >    address confidentiality.
 > That is true, but is confidentiallity especially for the first 
 > hop necessary? Imagine a scenario [...]

   It's equally easy for me to imagine scenarios like
   cable MSO's not turing on DOCSIS BPI (which I've heard
   is quite common), wireless AP's with braindead
   L2 encryption, and DSL providers not doing
   anything at all. When you need it, you need it.

   As for the middle boxen you posit, this is the
   price of non-transparency. Your middle box will 
   have an equal amount of trouble adultering a
   message covered by a keyed hash if it doesn't
   posess a key. Some of us think this is a feature.

		   Mike

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From: "Steffen Fries" <steffen.fries@mchp.siemens.de>
Organization: Siemens AG
To: Michael Thomas <mat@cisco.com>
Date: Wed, 24 Apr 2002 16:36:32 +0200
MIME-Version: 1.0
Subject: Re: [Sip] SIP message integrity
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>    As for the middle boxen you posit, this is the
>    price of non-transparency. Your middle box will 
>    have an equal amount of trouble adultering a
>    message covered by a keyed hash if it doesn't
>    posess a key. Some of us think this is a feature.

You are certainly right, but when no NAT/PAT is involved there 
is no need from the middle-box to alter the message. It would 
just need to trace the messages to open/close the appropriate 
ports on the firewall.

Anyway, I was interested if there are more scenarios evident 
were security in terms of message integrity protection is 
necessary and additional security protocols are not favoured to 
provide this service.

Steffen________________________________________________________
___
Steffen Fries,     Siemens AG, CT IC 3	
Otto-Hahn-Ring 6,  D-81730 Munich, Germany 
Phone:  (+49) 89 / 636-53403, Fax  :  (+49) 89 / 636-48000
Email:  Steffen.Fries@mchp.siemens.de
___________________________________________________________
    This email was written on 100% recycled bytes!



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From: "Finnie, Donald" <donald.finnie@intel.com>
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Subject: [Sip] 2543bis-09 Transport protocol mis-match
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Line 3585 in PS version:  "This is only to provide backwards compatibility
with  RFC 2543 compliant implementations that do not support UDP."

Should this be "...TCP."?

Regards

Donald Finnie

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From daemon@optimus.ietf.org  Wed Apr 24 12:36:18 2002
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Date: Wed, 24 Apr 2002 09:17:02 -0700
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
From: AC Mahendran <mahendra@qualcomm.com>
Subject: Re: [Sip] Reason draft
Cc: sip@ietf.org
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Gonzalo:

The 155 (Update Requested) response can definitely have multiple reasons.

For eg:  If authentication is required and at the same time the UAS doesn't 
support a particular extension in the request, the UAS can respond with two 
Reason codes:

Reason: SIP ;cause=401 ;text="Authentication needed"
Reason: SIP;cause=420  ;text="Bad Extension"

The corresponding headers, WWW-Authenticate & Unsupported, will provide 
more information needed in the UPDATE.

I really don't see the need in having this restriction in the specification.

thanks,
AC


At 08:19 AM 4/24/2002 +0300, Gonzalo Camarillo wrote:
>Hello,
>
>First of all, I need to fix the BNF in order to allow multiple Reason
>lines. It will be fixed in the next release.
>
>About your comment, which scenario do you have in mind for needing two
>Reason codes using the same protocol? Could you provide us with a
>particular example?
>
>Thanks,
>
>Gonzalo
>
>AC Mahendran wrote:
> >
> > Gonzalo:
> >
> > Section 2 of the draft states the following:
> > "A SIP message MAY contain more than one Reason values (i.e., multiple
> > Reason lines), but all of them MUST have different protocol values (e.g.,
> > one SIP and another Q.850)"
> >
> > Why is the restriction that the Reason values MUST have multiple protocol
> > values ? For example, What if I want to provide multiple reasons (using the
> > same protocol) as to why I am CANCELing a particular request ?
> >
> > I don't see any need for having the restriction.
> >
> > thanks,
> > AC
> >
> > At 01:10 PM 4/23/2002 +0300, Gonzalo Camarillo wrote:
> > >Hi,
> > >
> > >as previously suggested in the mailing list, I have removed the reason
> > >specific codes from the Reason draft. Until it appears in the archives,
> > >you can fetch it from:
> > >
> > >http://www.cs.columbia.edu/~gonzalo/draft-ietf-sip-reason-00.txt
> > >
> > >I would appreciate comments on the new format.
> > >
> > >Thanks,
> > >
> > >Gonzalo
> > >--
> > >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > >Telecom R&D               Fax   :  +358  9 299 30 52
> > >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > >Finland                   http://www.hut.fi/~gonzalo
> > >
> > >_______________________________________________
> > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >This list is for NEW development of the core SIP Protocol
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sipping@ietf.org for new developments on the application of sip
>
>--
>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>Telecom R&D               Fax   :  +358  9 299 30 52
>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>Finland                   http://www.hut.fi/~gonzalo
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



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From daemon@ns.ietf.org  Wed Apr 24 13:50:25 2002
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"Finnie, Donald" wrote:
> 
> Line 3585 in PS version:  "This is only to provide backwards
> compatibility
> with  RFC 2543 compliant implementations that do not support UDP."
> 
> Should this be "...TCP."?

Yes. We'll fix that during the authors 48 hours.

Thanks,
Jonathan R.

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

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From daemon@ns.ietf.org  Wed Apr 24 14:06:53 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Robert Sparks'" <rsparks@dynamicsoft.com>, <brian.rosen@cisco.com>,
        <jo@ipdialog.com>
Date: Wed, 24 Apr 2002 12:32:27 -0500
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Subject: [Sip] Seperating REFER and Referred By:
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Ok, here's how this cascade started, I think.

I asked Robert to extract Referred-By from the REFER draft and publish
it as a separate draft. This does not imply that Referred-By is being
abandoned, just that it will be pursued as a separate effort. 

Why?

Because Referred-by is a 450 message (count 'em) rat-hole, and we need
REFER for other stuff like NOW.

This is an application of the time-honored technique of "divide and
conquer", and it's time to do some conquering.

Thank you!

--
Dean


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From daemon@ns.ietf.org  Wed Apr 24 14:11:49 2002
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Date: Wed, 24 Apr 2002 13:42:47 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Sreerekha Madhava Shenoy <sreerekha.shenoy@wipro.com>
CC: sip@ietf.org
Subject: Re: [Sip] 'maddr' in Contact Header
References: <059901c1e755$351835a0$cbbca8c0@sreerekha>
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maddr will take precedence, see draft-ietf-sip-srv. However, we are
trying to move away from the use of maddr to specify hops. Loose routing
in bis eliminates the need for it. There is really no need at all to
register with an maddr in the contact in the registration.

-Jonathan R.


Sreerekha Madhava Shenoy wrote:
> 
> hi
>   We had a doubt about the significance of "maddr" parameter in the
> Contact
> header.
>   When the proxy gets the next hop information from the Location
> Service,
> should it honour the "maddr" (when present) as the next hop address ? Or
> should it take the   "host" part of the Contact header ?
> 
> Thanks
> regards
> rekha.
> 
>   ------------------------------------------------------------------------
> 
>    Wipro_Disclaimer.txtName: Wipro_Disclaimer.txt
>                        Type: Plain Text (text/plain)

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

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From daemon@ns.ietf.org  Wed Apr 24 14:18:22 2002
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Date: Wed, 24 Apr 2002 13:46:22 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Diether De Praetere <diether.de_praetere@alcatel.be>
CC: sip@ietf.org
Subject: Re: [Sip] Retry-After header
References: <3CBD760A.B3C15FB9@alcatel.be>
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Diether De Praetere wrote:
> 
> "14.2 UAS Behaviour"
> mentions that a 500 (Server Internal Error) needs to include a
> Retry-After header field ico. the 2nd invite is received before the
> final response of the 1st INVITE is not sent yet.
> 
> "20.33 Retry-After"
> this header field can be used with 503, 404, 413, 480, 486, 600 or 603.
> 
> I assume that this list should include the 500 too?

Yes. We will try to fix that during authors 48 hours.

Thanks,
Jonathan R.

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

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From daemon@ns.ietf.org  Wed Apr 24 14:54:02 2002
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Date: Wed, 24 Apr 2002 13:51:31 -0400
From: Vijaya Venkatachalam <vijaya@ulticom.com>
Organization: Ulticom, Inc.
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This is a multi-part message in MIME format.
--------------5DB090691C5FC8CEEA1708CD
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Hi,

Why are two drafts for ISUP to SIP mapping?
There is one in SIP WG (draft-ietf-sip-isup-03.txt)
and the other in SIPPING WG (draft-ietf-sipping-isup-01.txt).

Thanks,
Vijaya

--------------5DB090691C5FC8CEEA1708CD
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begin:vcard 
n:Venkatachalam;Vijaya
tel;fax:+1-856-866-2033
tel;work:+1-856-787-2853
x-mozilla-html:FALSE
url:www.ulticom.com
org:Ulticom, Inc.;Advanced Research 
adr:;;1020 Briggs Rd;Mt. Laurel;NJ;08054;USA
version:2.1
email;internet:vijaya@ulticom.com
fn:Vijaya Venkatachalam
end:vcard

--------------5DB090691C5FC8CEEA1708CD--



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From daemon@ns.ietf.org  Wed Apr 24 15:48:43 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Wed, 24 Apr 2002 15:14:16 -0400
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Subject: [Sip] To-tag Corner Case
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Section 8.2.6.2 of 2543bis says a To tag MUST be added to any provisional
response (except perhaps 100 Trying).  It goes on to say:

   The same tag MUST be used for all responses to that request, both 
   final and provisional (again excepting the 100 (Trying)). 

Section 12.1 says:

   Dialogs are created through the generation of non-failure responses
   to requests with specific methods. Within this specification, only
   2xx and 101-199 responses with a To tag to INVITE establish a dialog.

and of course, creation of a dialog involves the UAS adding a To-tag.

What I get out of this is:

(a) If the first response to an INVITE is a failure response, a To-tag MUST
NOT be added.

(b) If the first response to an INVITE is a 1xx (greater than 100), then a
To-tag MUST be present in that response and in ALL succeeding responses,
even if the final response is a failure response.

Some unexpected tags in the call flows document stimulated this observation,
which I present for your confirmation.

Tom Taylor
taylor@nortelnetworks.com
Ph. +1 613 736 0961 (ESN 396 1490)
 

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From daemon@ns.ietf.org  Wed Apr 24 16:30:14 2002
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	Wed, 24 Apr 2002 15:01:46 -0500
From: "Hussam Jarada" <hussam.jarada@ulticomdal.com>
To: "'Tom-PT Taylor'" <taylor@nortelnetworks.com>, <sip@ietf.org>
Subject: RE: [Sip] To-tag Corner Case
Date: Wed, 24 Apr 2002 15:01:44 -0500
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Hi,

The main issue I have with 2543bis-09 is that it's not clear with clear
examples that case were To tag is needed and not, cause in latest call flows
draft which been updated to reflect 2543bis-09 it may contain missing To tag
which I hope it just an typing error, I already I sent an email early for
clarification about this.

To my understanding
* To tag is added to any provisional response and final responses to all
requests, like it shown in section 11.1 line 1783, were no to tag in OPTIONS
request but in section 11.2 line 1818 there's To tag, and so 2543bis-09 is
missing To tag in F2 message, section 24.1


* I don't think (a) is true, cause I can not find any statement in the
2543bis-.09 that prove it, plus in section 17.1.1.3 which state that ACK
message for non-2xx differ from the To header field in the original request
by the addition of the tag parameter, it does not state that a provisional
response was sent early in order to have a To tag in the ACK message

Hope this helps.

Thanks.


Hussam Jarada
Ulticom Inc.

 -----Original Message-----
From: 	sip-admin@ietf.org [mailto:sip-admin@ietf.org]  On Behalf Of Tom-PT
Taylor
Sent:	Wednesday, April 24, 2002 2:14 PM
To:	'sip@ietf.org'
Subject:	[Sip] To-tag Corner Case

Section 8.2.6.2 of 2543bis says a To tag MUST be added to any provisional
response (except perhaps 100 Trying).  It goes on to say:

   The same tag MUST be used for all responses to that request, both
   final and provisional (again excepting the 100 (Trying)).

Section 12.1 says:

   Dialogs are created through the generation of non-failure responses
   to requests with specific methods. Within this specification, only
   2xx and 101-199 responses with a To tag to INVITE establish a dialog.

and of course, creation of a dialog involves the UAS adding a To-tag.

What I get out of this is:

(a) If the first response to an INVITE is a failure response, a To-tag MUST
NOT be added.

(b) If the first response to an INVITE is a 1xx (greater than 100), then a
To-tag MUST be present in that response and in ALL succeeding responses,
even if the final response is a failure response.

Some unexpected tags in the call flows document stimulated this observation,
which I present for your confirmation.

Tom Taylor
taylor@nortelnetworks.com
Ph. +1 613 736 0961 (ESN 396 1490)


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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From daemon@ns.ietf.org  Wed Apr 24 16:53:49 2002
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Subject: Re: [Sip] Matching responses to transactions (error in 9th draft?)
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: frank.derks@philips.com, sip@ietf.org
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Hi,

>Determining whether a response is a
>retransmission is actually hard, and not needed.

Is this true that a UAC doesn't need to determine whether a response is a
retransmission or not?
If so, wouldn't the UAC repeat the same processing for duplicate responses
which would impact performance, especially when the response includes SDP?

Thanks,
YoungSun



                                                                                             
                    Jonathan                                                                 
                    Rosenberg              To:     frank.derks@philips.com                   
                    <jdrosen@dynami        cc:     sip@ietf.org, (bcc: Youngsun              
                    csoft.com>             Song/Telcordia)                                   
                                           Subject:     Re: [Sip] Matching responses to      
                    04/10/02 12:59         transactions (error in 9th draft?)                
                    AM                                                                       
                                                                                             
                                                                                             







frank.derks@philips.com wrote:
>
> Section 17.1.3 of the 9th draft of the SIP specification states the
> following:
>
> "A response that matches a transaction matched by a previous response is
>
>  considered to be a restransmission of that response."
>
> This statement is preceded by two "matching rules", one about a matching
>
> branch parameter in the topmost Via header and one about a matching
> request method in the CSeq.
>
> Suppose that multiple responses are sent as the result of an INVITE.
> E.g.
> first a 183 Trying, then a 180 Ringing and then a 200 OK. All of these
> would match with the request, but according to the above statement, the
> 180 Ringing and the 200 OK would seem to be regarded as retransmissions
> of the 183 Trying. This can surely not be the intention.

Nope. This is an error in the spec. I think the right thing is to strike
the sentence entirely. The state machines are correct as defined, based
on the matching rules as defined. Determining whether a response is a
retransmission is actually hard, and not needed. Its not just the same
response code, since the same response could come from different
downstream UA. Its not a combination of response code and tags, since
the same UA could generate multiple provisional responses of the same
type (two 183, each with a different RSeq).

Thanks for pointing this out.

-Jonathan R.

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

_______________________________________________
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From daemon@ns.ietf.org  Thu Apr 25 00:16:56 2002
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Date: Wed, 24 Apr 2002 23:41:51 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Salva Rey Calatayud wrote:
> 
> Isn't there a mechanism whereby both UAs get back to an agreed
> state? How can the negotiation process continue properly
> otherwise?

It seems senseless to negotiate state with an implementation that is
broken. Its sort of like trying to talk louder in an attempt to
communicate with someone that doesn't even speak your language.
Tempting, but pointless.

If something breaks badly in the negotiation, I would recommend sending
a BYE.

-Jonathan R.

> On 10 Apr 2002, at 0:43, Jonathan Rosenberg wrote:
> 
> > THe spec does not detail error handling for the infinite different
> ways
> > in which they might occur. The choice is at the discretion of the
> > implementation.
> >
> > -Jonathan R.
> >
> > Salva Rey Calatayud wrote:
> > >
> > > Hi,
> > >
> > >         how should a UA act when let's say the other endpoint
> doesn't
> > > behave compliantly to the O/A exchange model. For instance, we
> > > send an offer, and none of the associated messages that could
> > > contain an answer does (an UPDATE with a 200OK without
> > > answer, INVITE with offer and none of the provisional response nor
> > > final 200OK contain an answer )
> > >
> > > thanks,
> > > Salva
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Apr 25 01:43:24 2002
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CC: steffen.fries@mchp.siemens.de, sip@ietf.org
Subject: Re: [Sip] SIP message integrity
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Michael Thomas wrote:
> 
> Steffen Fries writes:
>  > >    If you run SIP over UDP, TLS is useless and IPsec
>  > >    is a better choice. Also: the digest schemes don't
>  > >    address confidentiality.
>  > That is true, but is confidentiallity especially for the first
>  > hop necessary? Imagine a scenario [...]
> 
>    It's equally easy for me to imagine scenarios like
>    cable MSO's not turing on DOCSIS BPI (which I've heard
>    is quite common), wireless AP's with braindead
>    L2 encryption, and DSL providers not doing
>    anything at all. When you need it, you need it.


These are merely examples of poor practices in the wild. Once a few well
placed lawsuits are lodged I bet this practices ceases.



> 
>    As for the middle boxen you posit, this is the
>    price of non-transparency. Your middle box will
>    have an equal amount of trouble adultering a
>    message covered by a keyed hash if it doesn't
>    posess a key. Some of us think this is a feature.
> 
>                    Mike
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From daemon@optimus.ietf.org  Thu Apr 25 01:52:52 2002
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Steffen Fries wrote:
> 
> >    As for the middle boxen you posit, this is the
> >    price of non-transparency. Your middle box will
> >    have an equal amount of trouble adultering a
> >    message covered by a keyed hash if it doesn't
> >    posess a key. Some of us think this is a feature.
> 
> You are certainly right, but when no NAT/PAT is involved there
> is no need from the middle-box to alter the message. It would
> just need to trace the messages to open/close the appropriate
> ports on the firewall.
> 
> Anyway, I was interested if there are more scenarios evident
> were security in terms of message integrity protection is
> necessary and additional security protocols are not favoured to
> provide this service.

How about internal office communications, once deemed secure, to a
certain extent with the exception of a few PBX admins, which can now be
eavesdropped on by anyone with a LAN connection a sniffer, and assuming
enough knowledge to be dangerous in a switched environment, that can
capture and replay media of a confidential nature such as high level
conferencing.....

This scenario actually fits the need for additional security
protocols...but thats because I am concerned along those same lines.

Along the lines of not wanting additional security protocols...besides
issues related to dynamic pinholing, signaling modification for
NAT/NAPT, bypassing intrusion detection systems is another area of
concern. If certain signaling fields are encrypted unauthorized access
can be masked. I think that encryption of the media stream and signaling
authentication makes more sense. 

> 
> Steffen________________________________________________________
> ___
> Steffen Fries,     Siemens AG, CT IC 3
> Otto-Hahn-Ring 6,  D-81730 Munich, Germany
> Phone:  (+49) 89 / 636-53403, Fax  :  (+49) 89 / 636-48000
> Email:  Steffen.Fries@mchp.siemens.de
> ___________________________________________________________
>     This email was written on 100% recycled bytes!
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From daemon@optimus.ietf.org  Thu Apr 25 01:53:10 2002
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Steffen Fries wrote:
> 
> >    If you run SIP over UDP, TLS is useless and IPsec
> >    is a better choice. Also: the digest schemes don't
> >    address confidentiality.
> That is true, but is confidentiallity especially for the first
> hop necessary? Imagine a scenario, where you have a SIP
> telephony service provider and a small size company uses this
> service. The company will probably have a firewall deployed. If
> you use IPSec or TLS to protect the link between the user in
> the company and the telephony service provider, the user will
> not be able to make a phone call, since the port information of
> the RTP communication is encrypted and the firewall does not
> possess the appropriate key.


I would recomend that the firewall not be the endpoint for encryption,
or if it is then it will have the pre-share key.


%--%      +---------+        /-----\    /-------\    +----------+
 /\-------| firewall|--------| VPN |----|SP VPN |----| SP Proxy |
====      +---------+        \-----/    \-------/    +----------+
                                \___IPSec___/ 
                            or       
                     \__________IPSec____/

In the above scenario either way, the signaling must be taken care of
prior to encapsulation/decapsulation.

I agree with you, otherwise, the firewall cannot act on SIP signaling.

This only provides confidentiality between the customer Edge and the SP
edge. Internally on the customer network data is still "in the clear"

Chris
> 
> Steffen
> ___________________________________________________________
> Steffen Fries,     Siemens AG, CT IC 3
> Otto-Hahn-Ring 6,  D-81730 Munich, Germany
> Phone:  (+49) 89 / 636-53403, Fax  :  (+49) 89 / 636-48000
> Email:  Steffen.Fries@mchp.siemens.de
> ___________________________________________________________
>     This email was written on 100% recycled bytes!
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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Date: Thu, 25 Apr 2002 08:27:55 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: AC Mahendran <mahendra@qualcomm.com>
CC: sip@ietf.org, Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sip] Reason draft
References: <5.1.0.14.2.20020423140015.02756df8@clea.qualcomm.com> <5.1.0.14.2.20020424085924.02848580@clea.qualcomm.com>
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Hi,

well, in this case we would be providing functionality that does not
exist when we send a final response (rather than a 155).

Thefore, if we allow multiple Reason lines in a 155, it would be natural
to allow Reason also for final responses...

I would like to hear opinions about this.

Regards,

Gonzalo

AC Mahendran wrote:
> 
> Gonzalo:
> 
> The 155 (Update Requested) response can definitely have multiple reasons.
> 
> For eg:  If authentication is required and at the same time the UAS doesn't
> support a particular extension in the request, the UAS can respond with two
> Reason codes:
> 
> Reason: SIP ;cause=401 ;text="Authentication needed"
> Reason: SIP;cause=420  ;text="Bad Extension"
> 
> The corresponding headers, WWW-Authenticate & Unsupported, will provide
> more information needed in the UPDATE.
> 
> I really don't see the need in having this restriction in the specification.
> 
> thanks,
> AC
> 
> At 08:19 AM 4/24/2002 +0300, Gonzalo Camarillo wrote:
> >Hello,
> >
> >First of all, I need to fix the BNF in order to allow multiple Reason
> >lines. It will be fixed in the next release.
> >
> >About your comment, which scenario do you have in mind for needing two
> >Reason codes using the same protocol? Could you provide us with a
> >particular example?
> >
> >Thanks,
> >
> >Gonzalo
> >
> >AC Mahendran wrote:
> > >
> > > Gonzalo:
> > >
> > > Section 2 of the draft states the following:
> > > "A SIP message MAY contain more than one Reason values (i.e., multiple
> > > Reason lines), but all of them MUST have different protocol values (e.g.,
> > > one SIP and another Q.850)"
> > >
> > > Why is the restriction that the Reason values MUST have multiple protocol
> > > values ? For example, What if I want to provide multiple reasons (using the
> > > same protocol) as to why I am CANCELing a particular request ?
> > >
> > > I don't see any need for having the restriction.
> > >
> > > thanks,
> > > AC
> > >
> > > At 01:10 PM 4/23/2002 +0300, Gonzalo Camarillo wrote:
> > > >Hi,
> > > >
> > > >as previously suggested in the mailing list, I have removed the reason
> > > >specific codes from the Reason draft. Until it appears in the archives,
> > > >you can fetch it from:
> > > >
> > > >http://www.cs.columbia.edu/~gonzalo/draft-ietf-sip-reason-00.txt
> > > >
> > > >I would appreciate comments on the new format.
> > > >
> > > >Thanks,
> > > >
> > > >Gonzalo
> > > >--
> > > >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > > >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > > >Telecom R&D               Fax   :  +358  9 299 30 52
> > > >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > > >Finland                   http://www.hut.fi/~gonzalo
> > > >
> > > >_______________________________________________
> > > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >This list is for NEW development of the core SIP Protocol
> > > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > > >Use sipping@ietf.org for new developments on the application of sip
> >
> >--
> >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> >Telecom R&D               Fax   :  +358  9 299 30 52
> >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> >Finland                   http://www.hut.fi/~gonzalo
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

_______________________________________________
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From daemon@optimus.ietf.org  Thu Apr 25 02:18:51 2002
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Date: Thu, 25 Apr 2002 01:45:46 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: ysong@telcordia.com
CC: frank.derks@philips.com, sip@ietf.org
Subject: Re: [Sip] Matching responses to transactions (error in 9th draft?)
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ysong@telcordia.com wrote:
> 
> Hi,
> 
> >Determining whether a response is a
> >retransmission is actually hard, and not needed.
> 
> Is this true that a UAC doesn't need to determine whether a response is
> a
> retransmission or not?
> If so, wouldn't the UAC repeat the same processing for duplicate
> responses
> which would impact performance, especially when the response includes
> SDP?

No, there wouldn't need to be a repeat of the same processing. The state
machine for the client transaction tells you whether you need additional
processing. If a response matches the transaction, and the transaction
is in the completed state, you resend the ACK - thats it. The state
machine does not tell you to pass the response up to the TU, so no
further processing occurs. 

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

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From daemon@optimus.ietf.org  Thu Apr 25 02:18:54 2002
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Date: Thu, 25 Apr 2002 07:43:54 +0200
To: "Tom-PT Taylor"<taylor@nortelnetworks.com>,
        "'sip@ietf.org'" <sip@ietf.org>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Subject: Re: [Sip] To-tag Corner Case
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At 09:14 PM 4/24/2002, Tom-PT Taylor wrote:
>Section 8.2.6.2 of 2543bis says a To tag MUST be added to any provisional
>response (except perhaps 100 Trying).  It goes on to say:
>
>   The same tag MUST be used for all responses to that request, both 
>   final and provisional (again excepting the 100 (Trying)). 
>
>Section 12.1 says:
>
>   Dialogs are created through the generation of non-failure responses
>   to requests with specific methods. Within this specification, only
>   2xx and 101-199 responses with a To tag to INVITE establish a dialog.
>
>and of course, creation of a dialog involves the UAS adding a To-tag.
>
>What I get out of this is:
>
>(a) If the first response to an INVITE is a failure response, a To-tag MUST
>NOT be added.

I don't see how that is implied. Actually I think doing so would be badly 
restrictive (for example, To-tags are one way how to make an upstream UAC 
mirror piece of state in ACK).

-Jiri


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From daemon@optimus.ietf.org  Thu Apr 25 02:42:29 2002
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Subject: Re: [Sip] Timer D in the INVITE client transaction MUST inform the 
 TUwhen it fires????
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frank.derks@philips.com wrote:
> 
> 17.1.12 (lines 3312, 3313) states:
> 
> "If timer D fires while the client transaction is in the "Completed"
> state,
>  the client transaction MUST move to the terminated state, and it MUST
>  inform the TU of the timeout."
> 
> This does not apply to the cases when timers I, J, K fire in the other
> types
> of transactions. Moreover, I do not think that the above statement for
> timer
> D is correct: there is no purpose in informing the transaction user. The
> 
> sentence should therefore be removed.

I agree. 

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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Subject: Re: [Sip] To-tag Corner Case
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Jiri Kuthan wrote:
> 
> At 09:14 PM 4/24/2002, Tom-PT Taylor wrote:
> >Section 8.2.6.2 of 2543bis says a To tag MUST be added to any
> provisional
> >response (except perhaps 100 Trying).  It goes on to say:
> >
> >   The same tag MUST be used for all responses to that request, both
> >   final and provisional (again excepting the 100 (Trying)).
> >
> >Section 12.1 says:
> >
> >   Dialogs are created through the generation of non-failure responses
> >   to requests with specific methods. Within this specification, only
> >   2xx and 101-199 responses with a To tag to INVITE establish a
> dialog.
> >
> >and of course, creation of a dialog involves the UAS adding a To-tag.
> >
> >What I get out of this is:
> >
> >(a) If the first response to an INVITE is a failure response, a To-tag
> MUST
> >NOT be added.
> 
> I don't see how that is implied. Actually I think doing so would be
> badly
> restrictive (for example, To-tags are one way how to make an upstream
> UAC
> mirror piece of state in ACK).

Jiri is right - this is not implied at all. In fact, bis is quite
explicit about adding tags to all responses. Section 8.2.6.2 reads:

If a request contained a To tag in the request, the To header field in
the response MUST equal that of 1358
the request. However, if the To header field in the request did not
contain a tag, the URI in the To header 1359
field in the response MUST equal the URI in the To header field;
additionally, the UAS MUST add a tag to the To header field in the
response (with the exception of the 100 (Trying) response, in which a
tag MAY be 1361
present). This serves to identify the UAS that is responding, possibly
resulting in a component of a dialog 1362
ID. The same tag MUST be used for all responses to that request, both
final and provisional (again excepting 1363
the 100 (Trying)). Procedures for generation of tags are defined in
Section 19.3. 1364


The second sentence above is explicit that the UAS MUST add a tag to the
To header in the response, and the only exception is 100 trying. This
section is not specific to any type of response, and presents general
guidelines for constructing a response. Thus, it applies to all
responses except the 100 trying.


-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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Date: Thu, 25 Apr 2002 02:18:00 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
CC: AC Mahendran <mahendra@qualcomm.com>, sip@ietf.org
Subject: Re: [Sip] Reason draft
References: <3CC793DB.2FDF4248@lmf.ericsson.se>
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Gonzalo Camarillo wrote:
> 
> Hi,
> 
> well, in this case we would be providing functionality that does not
> exist when we send a final response (rather than a 155).
> 
> Thefore, if we allow multiple Reason lines in a 155, it would be natural
> to allow Reason also for final responses...
> 
> I would like to hear opinions about this.

I would rather not allow this, for the reason Gonzalo cites. The normal
processing of a request does not allow multiple error reports in a
single response. Since the generation of 155 is based on following the
normal request processing, but just changing the value of the status
code, it should not be possible to see multiple error status codes in
the 155. 

-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 25 03:13:47 2002
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To: "Fuxbruner, Amihay" <Amihay_Fuxbruner@icomverse.com>
CC: "'Ulrich Prakash'" <uprakash@netlab.hcltech.com>, sip@ietf.org
Subject: Re: [Sip] [Sip-implementors] Question on CANCEL - 
 backwards-compatibility
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"Fuxbruner, Amihay" wrote:
> 
> Does sending 200 OK is the formal meaning of no-op ?

The confusing word here is "behavior". In this context, it is meant to
refer to processing of this requet at layers above the transaction
layer. I agree it could be clearer. We will try to reword during authors
49 hours.

-Jonathan R.

> -----Original Message-----
> From: Ulrich Prakash [ mailto:uprakash@netlab.hcltech.com
> <mailto:uprakash@netlab.hcltech.com> ]
> Sent: Wednesday, March 20, 2002 9:46 PM
> To: Fuxbruner, Amihay; sip-implementors@cs.columbia.edu
> Subject: Re: [Sip-implementors] Question on CANCEL -
> backwards-compatibility
> 
> Hi,
> 
> please refer the following text from the draft.
> 
> Section 9.2 of bis-09, line 1497:
> If the original request was an INVITE,the UAS SHOULD immediately respond
> to the INVITE with a 487 (Request Terminated).
> 
> The behavior upon reception of a CANCEL request for any other method
> defined in this specification is effectively no-op.
> 
> Hence it should be a No operation.
> 
> Thanks,
> Prakash.
> 
> -----Original Message-----
> From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com
> <mailto:jdrosen@dynamicsoft.com> ]
> Sent: Wednesday, March 20, 2002 6:59 PM
> To: Fuxbruner, Amihay
> Cc: sip@ietf.org
> Subject: Re: [Sip] Question on CANCEL - backwards-compatibility
> 
> "Fuxbruner, Amihay" wrote:
> >
> > Hi,
> >
> > Early bis drafts support CANCEL for any request.
> > How UAS (bis 09) should respond to CANCEL for none-INVITE request from
> 
> > UAC (early bis) ?
> >
> 
> Send a 200 OK and do nothing else. This is discussed in the final
> paragraph of 9.2 of bis, which reads:
> 
> Regardless of the method of the original request, as long as the CANCEL
> matched an existing transac- tion, the UAS answers the CANCEL request
> itself with a 200 (OK) response. This response is constructed 1501
> following the procedures described in Section 8.2.6 noting that the To
> tag of the response to the CANCEL 1502
> and the To tag in the response to the original request SHOULD be the
> same. The response to CANCEL is 1503
> passed to the server transaction for transmission. 1504
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net <http://www.jdrosen.net>                   PH:
> (973) 952-5000
> http://www.dynamicsoft.com <http://www.dynamicsoft.com>

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

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To: "Hussam Jarada" <hussam.jarada@ulticomdal.com>
Cc: sip@ietf.org, "'Tom-PT Taylor'" <taylor@nortelnetworks.com>
Subject: RE: [Sip] To-tag Corner Case
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The issue shouldn't be whether there are clear examples, but whether there 
is
clear specification text. In cases where the specification leads to 
difficult
to read text to make the text correct, examples can be given.

Regards,

Frank



Hi,

The main issue I have with 2543bis-09 is that it's not clear with clear
examples that case were To tag is needed and not, cause in latest call 
flows
draft which been updated to reflect 2543bis-09 it may contain missing To 
tag
which I hope it just an typing error, I already I sent an email early for
clarification about this.

To my understanding
* To tag is added to any provisional response and final responses to all
requests, like it shown in section 11.1 line 1783, were no to tag in 
OPTIONS
request but in section 11.2 line 1818 there's To tag, and so 2543bis-09 is
missing To tag in F2 message, section 24.1


* I don't think (a) is true, cause I can not find any statement in the
2543bis-.09 that prove it, plus in section 17.1.1.3 which state that ACK
message for non-2xx differ from the To header field in the original 
request
by the addition of the tag parameter, it does not state that a provisional
response was sent early in order to have a To tag in the ACK message

Hope this helps.

Thanks.


Hussam Jarada
Ulticom Inc.

 -----Original Message-----
From:            sip-admin@ietf.org [mailto:sip-admin@ietf.org]  On Behalf 
Of Tom-PT
Taylor
Sent:            Wednesday, April 24, 2002 2:14 PM
To:              'sip@ietf.org'
Subject:                 [Sip] To-tag Corner Case

Section 8.2.6.2 of 2543bis says a To tag MUST be added to any provisional
response (except perhaps 100 Trying).  It goes on to say:

   The same tag MUST be used for all responses to that request, both
   final and provisional (again excepting the 100 (Trying)).

Section 12.1 says:

   Dialogs are created through the generation of non-failure responses
   to requests with specific methods. Within this specification, only
   2xx and 101-199 responses with a To tag to INVITE establish a dialog.

and of course, creation of a dialog involves the UAS adding a To-tag.

What I get out of this is:

(a) If the first response to an INVITE is a failure response, a To-tag 
MUST
NOT be added.

(b) If the first response to an INVITE is a 1xx (greater than 100), then a
To-tag MUST be present in that response and in ALL succeeding responses,
even if the final response is a failure response.

Some unexpected tags in the call flows document stimulated this 
observation,
which I present for your confirmation.

Tom Taylor
taylor@nortelnetworks.com
Ph. +1 613 736 0961 (ESN 396 1490)


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--=_alternative 0027C3B5C1256BA6_=
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<br><font size=2 face="Courier"><br>
The issue shouldn't be whether there are clear examples, but whether there is</font>
<br><font size=2 face="Courier">clear specification text. In cases where the specification leads to difficult</font>
<br><font size=2 face="Courier">to read text to make the text correct, examples can be given.</font>
<br>
<br><font size=2 face="Courier">Regards,</font>
<br>
<br><font size=2 face="Courier">Frank</font>
<br>
<br>
<br>
<br><font size=2 face="Courier">Hi,<br>
<br>
The main issue I have with 2543bis-09 is that it's not clear with clear<br>
examples that case were To tag is needed and not, cause in latest call flows<br>
draft which been updated to reflect 2543bis-09 it may contain missing To tag<br>
which I hope it just an typing error, I already I sent an email early for<br>
clarification about this.<br>
<br>
To my understanding<br>
* To tag is added to any provisional response and final responses to all<br>
requests, like it shown in section 11.1 line 1783, were no to tag in OPTIONS<br>
request but in section 11.2 line 1818 there's To tag, and so 2543bis-09 is<br>
missing To tag in F2 message, section 24.1<br>
<br>
<br>
* I don't think (a) is true, cause I can not find any statement in the<br>
2543bis-.09 that prove it, plus in section 17.1.1.3 which state that ACK<br>
message for non-2xx differ from the To header field in the original request<br>
by the addition of the tag parameter, it does not state that a provisional<br>
response was sent early in order to have a To tag in the ACK message<br>
<br>
Hope this helps.<br>
<br>
Thanks.<br>
<br>
<br>
Hussam Jarada<br>
Ulticom Inc.<br>
<br>
 -----Original Message-----<br>
From: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;sip-admin@ietf.org [mailto:sip-admin@ietf.org] &nbsp;On Behalf Of Tom-PT<br>
Taylor<br>
Sent: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Wednesday, April 24, 2002 2:14 PM<br>
To: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 'sip@ietf.org'<br>
Subject: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [Sip] To-tag Corner Case<br>
<br>
Section 8.2.6.2 of 2543bis says a To tag MUST be added to any provisional<br>
response (except perhaps 100 Trying). &nbsp;It goes on to say:<br>
<br>
 &nbsp; The same tag MUST be used for all responses to that request, both<br>
 &nbsp; final and provisional (again excepting the 100 (Trying)).<br>
<br>
Section 12.1 says:<br>
<br>
 &nbsp; Dialogs are created through the generation of non-failure responses<br>
 &nbsp; to requests with specific methods. Within this specification, only<br>
 &nbsp; 2xx and 101-199 responses with a To tag to INVITE establish a dialog.<br>
<br>
and of course, creation of a dialog involves the UAS adding a To-tag.<br>
<br>
What I get out of this is:<br>
<br>
(a) If the first response to an INVITE is a failure response, a To-tag MUST<br>
NOT be added.<br>
<br>
(b) If the first response to an INVITE is a 1xx (greater than 100), then a<br>
To-tag MUST be present in that response and in ALL succeeding responses,<br>
even if the final response is a failure response.<br>
<br>
Some unexpected tags in the call flows document stimulated this observation,<br>
which I present for your confirmation.<br>
<br>
Tom Taylor<br>
taylor@nortelnetworks.com<br>
Ph. +1 613 736 0961 (ESN 396 1490)<br>
<br>
<br>
_______________________________________________<br>
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This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
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<br>
<br>
_______________________________________________<br>
Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
This list is for NEW development of the core SIP Protocol<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sipping@ietf.org for new developments on the application of sip<br>
</font>
<br>
<br>
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From: "Steffen Fries" <steffen.fries@mchp.siemens.de>
Organization: Siemens AG
To: cmartin@dasecurenetworks.com
Date: Thu, 25 Apr 2002 10:27:05 +0200
MIME-Version: 1.0
Subject: Re: [Sip] SIP message integrity
Reply-to: steffen.fries@mchp.siemens.de
CC: Michael Thomas <mat@cisco.com>, sip@ietf.org
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> How about internal office communications, once deemed secure, to a
> certain extent with the exception of a few PBX admins, which can now be
> eavesdropped on by anyone with a LAN connection a sniffer, and assuming
> enough knowledge to be dangerous in a switched environment, that can
> capture and replay media of a confidential nature such as high level
> conferencing.....
> 
> This scenario actually fits the need for additional security
> protocols...but thats because I am concerned along those same lines.
> 
Encryption of the media stream in corporate environments is 
certainly an important requirement. It may also be interesting 
for carrier grade solutions.

> Along the lines of not wanting additional security protocols...besides
> issues related to dynamic pinholing, signaling modification for NAT/NAPT,
> bypassing intrusion detection systems is another area of concern. If
> certain signaling fields are encrypted unauthorized access can be masked.
> I think that encryption of the media stream and signaling authentication
> makes more sense.

Does this mean you would favour signaling authentication rather 
than signaling integrity? Is the reason for this the support of 
NAT/NAPT?

Steffen


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Folks,

I have put together a draft about how to use SigComp and SIP.

Until it appears in the archives, you can fetch it from:
http://www.cs.columbia.edu/~gonzalo/draft-camarillo-sip-compression-00.txt

Note that this draft does *not* describe a DNS based approach.

As usual, feedback is welcome.

Regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Security Mechanism Agreement for SIP Sessions
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-sip-sec-agree-00.txt
	Pages		: 16
	Date		: 24-Apr-02
	
SIP has a number of security mechanisms for hop-by-hop and end-to-end
protection. Some of the security mechanisms have been built in to the
SIP protocol, such as HTTP authentication or secure attachments. In
these mechanisms there are even alternative algorithms and parameters.
Currently it isn't possible to select which security mechanisms to use
over a connection. In particular, even if some mechanisms such as
OPTIONS were used to make this selection, the selection would be vul¡
nerable against the Bidding-Down attack.  This document defines a
header for negotiating the security mechanisms within SIP. A SIP
entity applying this mechanism must always require some minimum secu¡
rity (i.e. integrity protection) from all communicating parties in
order to secure the negotiation, but the negotiation can agree on
which specific minimum security is used.

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From daemon@optimus.ietf.org  Thu Apr 25 10:57:45 2002
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Date: Thu, 25 Apr 2002 08:38:09 -0500
From: Alan Johnston <alan.johnston@wcom.com>
Subject: RE: [Sip] To-tag Corner Case
In-reply-to: <001301c1ebca$dc51bcf0$ee9e82cc@ulticomdal.com>
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Hussam Jarada wrote: 

> To my understanding
> * To tag is added to any provisional response and final 
> responses to all requests, like it shown in section 11.1 line 
> 1783, were no to tag in OPTIONS request but in section 11.2 
> line 1818 there's To tag, and so 2543bis-09 is missing To tag 
> in F2 message, section 24.1
> 

The correct use of tags is shown in Section 11 with the OPTIONS.
Section 24.1 does contain an error in the lack of a To tag in the
response to the REGISTER.  This is the same error in the Call Flows.

The only change to tag usage in the Call Flows documents relates to
proxy behavior in forwarding non-2xx final responses.  It the past, the
call flows showed proxies replacing the existing tag in the non-2xx
response with its own tag.  Now, if possible, the proxy keeps the tag
when it proxies the response, which can help in troubleshooting -
figuring out which server sent the error.

Thanks,
Alan Johnston
WorldCom
sip:alan@siptest.wcom.com


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From daemon@optimus.ietf.org  Thu Apr 25 11:11:52 2002
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Subject: [Sip] Re: [rohc] SigComp and SIP
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Gonzalo,

I think this is definitely a step in the right direction. I'll note that
this is yet another thing that wouldn't work without loose routing :)

A few comments:

* Section 3 leaves out an important means for obtaining the URI -
configuration. Ultimately, we want to move towards configuration of
outbound proxies through URIs, NOT IP addresses. One reason is that we
can support supplemantal information, just like this, as part of the
URI. 

* I think a client MUST NOT send a sigcomp request to a server if it
doesn't know whether it supports compression. Section 3 provides cases
in which you can sort of guess whether it does. I don't like that. The
guesses are just that - guesses, and may be wrong. Indeed, the
heuristics are somewhat ill defined. For example:

>  o In a previous transaction, the client obtained a URI for the
>           same server with the parameter comp=sigcomp (e.g., a Contact
>           header field in a 2xx response for a previous OPTIONS request
>           or an entry in the route set of a previous dialog).

How do I know that the OPTIONS request this time was routed to the same
element that got it last time? Indeed, if, last time, the contact URI in
the 2xx to OPTIONS had a comp param, and this time, it doesn't, isn't
this a good sign that it has NOT been routed to the same UA?


* I think its worth noting a few things about this approach. First, a
proxy that gets a request uncompressed, and wants to get it compressed
in subsequent requests, can return a 301 with a contact header that has
the comp parameter. This will result in a caching of the binding in the
client that sent the request. Secondly, a SIP element that implements
this will need to be prepared to receive compressed or uncompressed
messages on the same port, and thus will need to look at the cookie in
the topmost bits to demux.



> A response is sent to the host in the sent-by parameter of the Via
>    header field. If the topmost Via header field contains the parameter
>    comp=sigcomp, the response SHOULD be compressed. Otherwise, the
>    response SHOULD NOT be compressed.

I would say MUST NOT be compressed. The criteria is - will things work
if you violate this? If I send a compressed response to something which
doesn't support it, the response is discarded. The result is total
failure of the transaction. There is no recovrey. So, it should be MUST
NOT.

>  A proxy performing Record-Route inspects the Record-Route and the
>    Contact header fields in the response. It looks for the URI of the
>    next upstream (closer to the user agent client) hop in the route set.
>    If this URI contains the parameter comp=sigcomp, the proxy SHOULD add
>    comp=sigcomp to its entry in the Record-Route header field.

Why? If the upstream element doesn't support this extension, the sigcomp
parameter is ignored, and the message is sent uncompressed. Thats what
you want. The decision about whether to insert a URI with the comp
parameter should be entirely a local one.

* section 7 - i think these considerations are addressed by sigcomp
itself. You should really discuss whether the proposed mechanism
introduces additional issues.


Thanks,
Jonathan R.


Gonzalo Camarillo wrote:
> 
> Folks,
> 
> I have put together a draft about how to use SigComp and SIP.
> 
> Until it appears in the archives, you can fetch it from:
> http://www.cs.columbia.edu/~gonzalo/draft-camarillo-sip-compression-00.t
> xt
> 
> Note that this draft does *not* describe a DNS based approach.
> 
> As usual, feedback is welcome.
> 
> Regards,
> 
> Gonzalo
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
> 
> _______________________________________________
> Rohc mailing list
> Rohc@ietf.org
> https://www1.ietf.org/mailman/listinfo/rohc

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From daemon@optimus.ietf.org  Thu Apr 25 11:15:10 2002
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Date: Thu, 25 Apr 2002 09:27:55 -0500
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
CC: AC Mahendran <mahendra@qualcomm.com>, sip@ietf.org
Subject: Re: [Sip] Reason draft
References: <5.1.0.14.2.20020423140015.02756df8@clea.qualcomm.com> <5.1.0.14.2.20020424085924.02848580@clea.qualcomm.com> <3CC793DB.2FDF4248@lmf.ericsson.se>
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Gonzalo Camarillo wrote:

> Hi,
> 
> well, in this case we would be providing functionality that does not
> exist when we send a final response (rather than a 155).
> 
> Thefore, if we allow multiple Reason lines in a 155, it would be natural
> to allow Reason also for final responses...
> 
> I would like to hear opinions about this.


Currently in SIP, a non-2xx final response contains the reason why the
session failed (in case the request was establishing a session, to
begin with).  A non-2xx final response is *not* the aggregation of all
particular reasons why the response failed.  I think that allowing such
an aggregation would be an invitation to open a can of worms as one
figures out which headers to include/preclude in an aggregated response
(not to mention the receiving side, which will have to deduce what went
wrong based on presence/absence of such headers).

Furthermore, what if 2 things went wrong, but fixing one automatically
fixes the other (for instance, a UAS received a very long R-URI with an
unknown scheme, so it sends 2 Reason headers, one containing 414
"Request-URI too long" and the other 416 "Unsupported URI Scheme".  A
UAC may attempt to fix 414 by using an alternate shorter R-URI, which
just happens to be a URI that contains a scheme that the UAS
understands).

I think what the I-D says right now:

   "A SIP message MAY contain more than one Reason values (i.e.,
   multiple Reason lines), but all of them MUST have different
   protocol values (e.g., one SIP and another Q.850)"

should stand.  This mirrors the current response behavior in SIP.

Regards,

- vijay

-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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Subject: Re: [Sip] Matching responses to transactions (error in 9th draft?)
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
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>
>> Hi,
>>
> >Determining whether a response is a
> >>retransmission is actually hard, and not needed.
>>
>> Is this true that a UAC doesn't need to determine whether a response is
>> a
>> retransmission or not?
>> If so, wouldn't the UAC repeat the same processing for duplicate
>> responses
>> which would impact performance, especially when the response includes
>> SDP?

>No, there wouldn't need to be a repeat of the same processing. The state
>machine for the client transaction tells you whether you need additional
>processing. If a response matches the transaction, and the transaction
>is in the completed state, you resend the ACK - thats it. The state
>machine does not tell you to pass the response up to the TU, so no
>further processing occurs.
>
>-Jonathan R.

In case of 1xx received, the INVITE client transaction in the Proceeding
state tells you to pass the response up to the TU. In this case, is the TU
responsible for determining the duplicate response or for repeating the
same processing treating the duplicate response as a new response?

Thanks,
YoungSun


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From daemon@optimus.ietf.org  Thu Apr 25 13:03:05 2002
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Subject: [Sip] Updated Digest-AKA draft
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Hi All,

I've submitted an updated version of the Digest-AKA draft to the I-D directories. Until it appears there, a copy is available at:

	http://www.iki.fi/aki.niemi/draft-ietf-sip-digest-aka-01.txt

This version includes mostly editorial changes and some clarifications with an updated section on IANA considerations.

Comments are welcome!

Cheers,
Aki

---

Abstract

   The Hypertext Transfer Protocol (HTTP) Authentication Framework
   includes two authentication schemes: Basic and Digest.  Both schemes
   employ a shared secret based mechanism for access authentication.
   The Authentication and Key Agreement (AKA) mechanism performs user
   authentication and session key distribution in Universal Mobile
   Telecommunications System (UMTS) networks.  AKA is a challenge-
   response based mechanism that uses symmetric cryptography.  This memo
   specifies an AKA based one-time password generation mechanism for
   HTTP Digest access authentication.

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From daemon@optimus.ietf.org  Thu Apr 25 13:16:23 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Jiri Kuthan
	 <kuthan@fokus.gmd.de>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] To-tag Corner Case
Date: Thu, 25 Apr 2002 11:51:39 -0400
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Then must the statement I quote from 12.1 be fixed, or did I go too far in
equating the addition of a To-tag to establishment of a dialog?

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Thursday, April 25, 2002 2:13 AM
To: Jiri Kuthan
Cc: Taylor, Tom-PT [CAR:B800:EXCH]; 'sip@ietf.org'
Subject: Re: [Sip] To-tag Corner Case




Jiri Kuthan wrote:
> 
> At 09:14 PM 4/24/2002, Tom-PT Taylor wrote:
> >Section 8.2.6.2 of 2543bis says a To tag MUST be added to any
> provisional
> >response (except perhaps 100 Trying).  It goes on to say:
> >
> >   The same tag MUST be used for all responses to that request, both
> >   final and provisional (again excepting the 100 (Trying)).
> >
> >Section 12.1 says:
> >
> >   Dialogs are created through the generation of non-failure responses
> >   to requests with specific methods. Within this specification, only
> >   2xx and 101-199 responses with a To tag to INVITE establish a
> dialog.
> >
> >and of course, creation of a dialog involves the UAS adding a To-tag.
> >
> >What I get out of this is:
> >
> >(a) If the first response to an INVITE is a failure response, a To-tag
> MUST
> >NOT be added.
> 
> I don't see how that is implied. Actually I think doing so would be
> badly
> restrictive (for example, To-tags are one way how to make an upstream
> UAC
> mirror piece of state in ACK).

Jiri is right - this is not implied at all. In fact, bis is quite
explicit about adding tags to all responses. Section 8.2.6.2 reads:

If a request contained a To tag in the request, the To header field in
the response MUST equal that of 1358
the request. However, if the To header field in the request did not
contain a tag, the URI in the To header 1359
field in the response MUST equal the URI in the To header field;
additionally, the UAS MUST add a tag to the To header field in the
response (with the exception of the 100 (Trying) response, in which a
tag MAY be 1361
present). This serves to identify the UAS that is responding, possibly
resulting in a component of a dialog 1362
ID. The same tag MUST be used for all responses to that request, both
final and provisional (again excepting 1363
the 100 (Trying)). Procedures for generation of tags are defined in
Section 19.3. 1364


The second sentence above is explicit that the UAS MUST add a tag to the
To header in the response, and the only exception is 100 trying. This
section is not specific to any type of response, and presents general
guidelines for constructing a response. Thus, it applies to all
responses except the 100 trying.


-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 25 14:42:50 2002
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ysong@telcordia.com wrote:
> 
> >
> >> Hi,
> >>
> > >Determining whether a response is a
> > >>retransmission is actually hard, and not needed.
> >>
> >> Is this true that a UAC doesn't need to determine whether a response
> is
> >> a
> >> retransmission or not?
> >> If so, wouldn't the UAC repeat the same processing for duplicate
> >> responses
> >> which would impact performance, especially when the response includes
> >> SDP?
> 
> >No, there wouldn't need to be a repeat of the same processing. The
> state
> >machine for the client transaction tells you whether you need
> additional
> >processing. If a response matches the transaction, and the transaction
> >is in the completed state, you resend the ACK - thats it. The state
> >machine does not tell you to pass the response up to the TU, so no
> >further processing occurs.
> >
> >-Jonathan R.
> 
> In case of 1xx received, the INVITE client transaction in the Proceeding
> state tells you to pass the response up to the TU. In this case, is the
> TU
> responsible for determining the duplicate response or for repeating the
> same processing treating the duplicate response as a new response?

For a bis compliant element, there is no differing treatment of
duplicate provisional responses. All are forwarded upstream by proxies,
all are associated with a dialog and possibly rendered to the user at a
UA. The reliability of provional responses spec defines reliability for
1xx, which has a notion of retransmits. It provides a separate header
used to determine retransmits. That processing is not done at the
transaction layer, but in the TU.

-Jonathan R.


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

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From daemon@optimus.ietf.org  Thu Apr 25 14:43:51 2002
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CC: Jiri Kuthan <kuthan@fokus.gmd.de>, "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] To-tag Corner Case
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Tom-PT Taylor wrote:
> 
> Then must the statement I quote from 12.1 be fixed, or did I go too far
> in equating the addition of a To-tag to establishment of a dialog?

Yes, you went to far. 

-Jonathan R.

> 
> -----Original Message-----
> From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com
> <mailto:jdrosen@dynamicsoft.com> ]
> Sent: Thursday, April 25, 2002 2:13 AM
> To: Jiri Kuthan
> Cc: Taylor, Tom-PT [CAR:B800:EXCH]; 'sip@ietf.org'
> Subject: Re: [Sip] To-tag Corner Case
> 
> Jiri Kuthan wrote:
> >
> > At 09:14 PM 4/24/2002, Tom-PT Taylor wrote:
> > >Section 8.2.6.2 of 2543bis says a To tag MUST be added to any
> > provisional
> > >response (except perhaps 100 Trying).  It goes on to say:
> > >
> > >   The same tag MUST be used for all responses to that request, both
> > >   final and provisional (again excepting the 100 (Trying)).
> > >
> > >Section 12.1 says:
> > >
> > >   Dialogs are created through the generation of non-failure
> responses
> > >   to requests with specific methods. Within this specification, only
> 
> > >   2xx and 101-199 responses with a To tag to INVITE establish a
> > dialog.
> > >
> > >and of course, creation of a dialog involves the UAS adding a To-tag.
> 
> > >
> > >What I get out of this is:
> > >
> > >(a) If the first response to an INVITE is a failure response, a
> To-tag
> > MUST
> > >NOT be added.
> >
> > I don't see how that is implied. Actually I think doing so would be
> > badly
> > restrictive (for example, To-tags are one way how to make an upstream
> > UAC
> > mirror piece of state in ACK).
> 
> Jiri is right - this is not implied at all. In fact, bis is quite
> explicit about adding tags to all responses. Section 8.2.6.2 reads:
> 
> If a request contained a To tag in the request, the To header field in
> the response MUST equal that of 1358
> the request. However, if the To header field in the request did not
> contain a tag, the URI in the To header 1359
> field in the response MUST equal the URI in the To header field;
> additionally, the UAS MUST add a tag to the To header field in the
> response (with the exception of the 100 (Trying) response, in which a
> tag MAY be 1361
> present). This serves to identify the UAS that is responding, possibly
> resulting in a component of a dialog 1362
> ID. The same tag MUST be used for all responses to that request, both
> final and provisional (again excepting 1363
> the 100 (Trying)). Procedures for generation of tags are defined in
> Section 19.3. 1364
> 
> The second sentence above is explicit that the UAS MUST add a tag to the
> 
> To header in the response, and the only exception is 100 trying. This
> section is not specific to any type of response, and presents general
> guidelines for constructing a response. Thus, it applies to all
> responses except the 100 trying.
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net <http://www.jdrosen.net>                   PH:
> (973) 952-5000
> http://www.dynamicsoft.com <http://www.dynamicsoft.com>

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

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From daemon@optimus.ietf.org  Thu Apr 25 15:17:19 2002
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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inline.

Bob Penfield wrote:
> 
> Last night at the meeting I raised the issue that it is burdensome for
> the
> UAC to figure out what the problem is with a request when a 155
> provisional
> response is received. For example, there would be no way to tell that a
> 416
> (Unsupported URI scheme) problem exists. There were 2 suggestions to
> solve
> this problem.
> 
> 1. Include the proposed Reason header (or a simple version of it) which
> would indicate the associated 4xx error code.
> 
> 2. Include a status-line in a sipfrag body.
> 
> The first option was rejected because the Reason header will not be
> ready in
> time for the 2nd bundle (manyfolks, privacy, call-auth, etc.) and it has
> also been sent back to SIPPING for a better definition of requirements.
> The
> idea of a simplified reason header was rejected because we don't want to
> add
> too many headers.
> 
> The second option does not seem promising because of issues raised with
> the
> sipfrag MIME type (I'm not aware of what those are).
> 
> My fear is that we would need a new option tag when the Reason header
> was
> done because it would impose slightly different behavior requirements on
> the
> UAS and UAC.
> 
> After further investigation, it appears that only 416 is a problem and
> all
> other reason can be determined by the presence of a specific header (as
> described in the draft). Therefore, if we just added another status code
> (say 156) to cover 416, I think we would have everything covered.
> 
> 401: 155 with WWW-Authenticate
> 407: 155 with Proxy-Authenticate
> 415: 155 with Accept/Accept-Encoding/Accept-Language
> 416: 156
> 420: 155 with Unsupported
> 423: 155 with Min-Expires
> 488: 155 with Warning
> 
> This still requires the UAC to check the headers. And presence of an
> Accept
> header does not necessarily imply 415, unless we change the draft to say
> that it does. We could specify that all the other headers should be
> checked
> first.
> 
> Another alternative would be to define a 15x status codes for each
> corresponding 4xx code so that the UAS can be explicit about the
> problem.
> 
> 401&407: 151
> 415: 152
> 416: 153
> 420: 154
> 423: 155
> 488: 156
> 
> We could use 150 as a place holder for use with the future Reason header
> and
> indicate that the UAC should CANCEL if it cannot figure out what the
> problem
> is.
> 
> The new status codes may be overkill.

I don't like this. It effectively requires us to register two response
codes for every repairable error response. I would rather use Reason,
thats what its meant for.

-Jonathan R.

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

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From daemon@optimus.ietf.org  Thu Apr 25 15:22:29 2002
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Bob Penfield wrote:
> 
> Section 4 says the UPDATE may contain an offer any time after a dialog
> is
> established. Although section 5.3 says it must follow the offer/answer
> rules
> you may want to point out in section 4 that the UPDATE can only contain
> an
> offer if allowed under the offer/answer rules.

This section is meant to overview operation, and doesn't specify
normative behavior. So, I don't want to get into details. I reworded to
say:

Afterwards,
the UAC can generate an UPDATE method that contains
an SDP offer \citenorm{draft-ietf-mmusic-sdp-offer-answer} for the
purposes of updating the session.

> 
> Section 5.3 talks about the Supported header in the response, but
> section
> 5.2.1 mandates the response contain the Require header. This should be
> made
> consistent. Otherwise it is not clear why the Require:update is needed.
> Also, the language in 5.3 implies that the fact that it is a 155
> response is
> sufficient.

There are two different conditions here:

Condition 1: the UAS sent a 155. In this case, the UAC really needs to
send an update to get it out of the wait state it is in. This is the
reason for the MUST level inclusion of Require in the 155. Effectively,
processing of the 155 requires an extension, and that extension is
separate from the support of the UPDATE method.

Condition 2: the uAS just sent a vanilla reliable provisional response
(i.e., 183). The UAC can send an update if it likes. UPDATE is a method
supported by the UAS. So, it is included in an Allow header, just like
any other method. Thus, the text in 5.3 is wrong, since it says
Supported. It needs to say Allow. Allow is mentioned in section 5.1 as
something the UAC can include. It is supposed to be symmetric.


> 
> Since we have changed the strength of the UAS requirement for generating
> a
> 155 from MAY to SHOULD, I suggest that we change the addition of
> Require:update in the proxy from a SHOULD to a MAY. If the UAS does not
> support update, you'll end up with 3 exchanges if there is a repairable
> error in the INVITE instead of 2. I don't see any great advantage in the
> proxy requiring update.

OK, I think that is sensible.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From daemon@ns.ietf.org  Thu Apr 25 18:39:50 2002
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Subject: [Sip] a modest proposal for update
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Folks,

As I was integrating comments on UPDATE for the final rev for iesg, I
begin to rethink our decision on keeping the HERFP stuff (155) as part
of this draft. The current usage really specifies to behaviors; they
even have different ways of indicating support for them (Allow: UPDATE
and Supported/Require: update). So, I would like to propose that we
excise the HERFP stuff, and proceed with UPDATE alone as a mechanism for
modifying session parameters. 

Here are some of the pros/cons:

pros:

  + cleaner separation of what is probably two separate things
  + we can revert to using the Reason header, since that aspect of it
would go in the HERFP document, not in UPDATE baseline
  + the HERFP stuff is a little immature still. I would prefer to have
some implementation experience with it before going to RFC.
  + the HERFP stuff is less time sensitive; we need UPDATE for the
purposes of session modification (manyfolks needs it)

cons:

  - somewhat major change at a fairly late point in the game, although
it should be fairly easy to excise


Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From daemon@ns.ietf.org  Thu Apr 25 20:44:24 2002
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Steffen Fries wrote:
> 
> > How about internal office communications, once deemed secure, to a
> > certain extent with the exception of a few PBX admins, which can now be
> > eavesdropped on by anyone with a LAN connection a sniffer, and assuming
> > enough knowledge to be dangerous in a switched environment, that can
> > capture and replay media of a confidential nature such as high level
> > conferencing.....
> >
> > This scenario actually fits the need for additional security
> > protocols...but thats because I am concerned along those same lines.
> >
> Encryption of the media stream in corporate environments is
> certainly an important requirement. It may also be interesting
> for carrier grade solutions.
> 
> > Along the lines of not wanting additional security protocols...besides
> > issues related to dynamic pinholing, signaling modification for NAT/NAPT,
> > bypassing intrusion detection systems is another area of concern. If
> > certain signaling fields are encrypted unauthorized access can be masked.
> > I think that encryption of the media stream and signaling authentication
> > makes more sense.
> 
> Does this mean you would favour signaling authentication rather
> than signaling integrity? Is the reason for this the support of
> NAT/NAPT?

No I would prefer both actually, but I am inclined to be one of the more
paranoid.  :^)

But yes the reasoning for my comment is due to the fact that to permit
signaling through sip enabled nat/napt, at a minimum the SDP must not be
encrypted. That is not to say that I dont want full encryption....I do,
but thats another discussion and a different topic.


> 
> Steffen
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Fri Apr 26 04:55:34 2002
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From: "Salva Rey Calatayud" <salreyca@teleco.upv.es>
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Date: Wed, 26 Apr 2000 10:35:24 +0200
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Hi,


	I have a doubt regarding the concept of a dialog when a session 
is made up by several participants.


	is there any relation between the call-id of all the dialogs that 
make up a session?


	Here is an extract from rfc-2543-bis09 


<bold><color><param>0100,0100,0100</param><FontFamily><param>Times New Roman</param>Abstract

</bold>The Session Initiation Protocol (SIP) is an application-layer control 
(signaling) protocol for creating, <FontFamily><param>Arial</param><smaller><smaller>15

<FontFamily><param>Times New Roman</param><bigger><bigger>modifying, and terminating sessions with one or more participants. <FontFamily><param>Arial</param><smaller><smaller>16




<bigger><bigger>(309) ...<FontFamily><param>Times New Roman</param><bigger>. SIP can also invite participants to already existing 
sessions, such as multicast conferences.</color><FontFamily><param>Arial</param><smaller>


 (line 594) <color><param>0100,0100,0100</param><FontFamily><param>Times New Roman</param><bigger>The most important method in SIP is the <FontFamily><param>Arial</param>INVITE 
<FontFamily><param>Times New Roman</param>method, which is used to establish a session between participants. 
A session is a collection of participants, and streams of media 
between them,... Section 13 discusses how sessions are initiated, 
resulting in one or more SIP dialogs.


(2069) when multiple 2xx responses are received from different 
remote UAs (because the <FontFamily><param>Arial</param>INVITE <FontFamily><param>Times New Roman</param>forked), each 2xx establishes a 
different dialog. All these dialogs are part of the sa
me call.


	Now<FontFamily><param>Arial</param><smaller> it seems to me that all these participants correspond to 
the endpoint where a single user can be reached, what about the 
other "participants" in the session? How are they "invited"? Are two 
different concepts of session being used here, namely, at a sip 
signalling level with only endpoint, and at a call level?


thanks,

Salva</color>




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Security Mechanism Agreement for SIP Sessions
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-sip-sec-agree-00.txt
	Pages		: 16
	Date		: 24-Apr-02
	
SIP has a number of security mechanisms for hop-by-hop and end-to-end
protection. Some of the security mechanisms have been built in to the
SIP protocol, such as HTTP authentication or secure attachments. In
these mechanisms there are even alternative algorithms and parameters.
Currently it isn't possible to select which security mechanisms to use
over a connection. In particular, even if some mechanisms such as
OPTIONS were used to make this selection, the selection would be vul¡
nerable against the Bidding-Down attack.  This document defines a
header for negotiating the security mechanisms within SIP. A SIP
entity applying this mechanism must always require some minimum secu¡
rity (i.e. integrity protection) from all communicating parties in
order to secure the negotiation, but the negotiation can agree on
which specific minimum security is used.

A URL for this Internet-Draft is:
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_______________________________________________
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From daemon@optimus.ietf.org  Fri Apr 26 06:41:06 2002
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From: "lakshmi" <lakshmi@ipgen-india.com>
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Hi all,

In message-waiting-02 draft, hname in message-summery 
syntax is given as 

message-headers = hname ":" SP hvalue CRLF 
hname = #(alphanum | "-" | "_")

i think it should be 1* in place of #. bcos if multiple
hname comes according rfc2543bis-02 will be separated by comma.


summary-line = media-type ":" SP new "/" old  [ urgent ] CRLF 
   new = WHOLENUMBER 
   old = WHOLENUMBER 
   new-urgent = WHOLENUMBER 
   old-urgent  = WHOLENUMBER 
   WHOLENUMBER = *DIGIT 

in the above syntax WHOLENUMBER should be 1*Digit.
bcos * means zero or multiples.

Anybody clarify abt the following syntax.

Thanks 
Lakshmi



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr 26 07:08:17 2002
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Hi all,

In message-waiting-02 draft, hname in message-summery 
syntax is given as 

message-headers = hname ":" SP hvalue CRLF 
hname = #(alphanum | "-" | "_")

i think it should be 1* in place of #. bcos if multiple
hname comes according rfc2543bis-02 will be separated by comma.


summary-line = media-type ":" SP new "/" old  [ urgent ] CRLF 
   new = WHOLENUMBER 
   old = WHOLENUMBER 
   new-urgent = WHOLENUMBER 
   old-urgent  = WHOLENUMBER 
   WHOLENUMBER = *DIGIT 

in the above syntax WHOLENUMBER should be 1*Digit.
bcos * means zero or multiples.

Anybody clarify abt the following syntax.

Thanks 
Lakshmi



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr 26 07:43:59 2002
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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sip@ietf.org
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Date: 26 Apr 2002 13:06:32 +0200
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Subject: [Sip] REFER drafts
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I've just submitted three new refer related drafts.
Until they hit the repository, you can get them at

http://www.nostrum.com/~rjsparks/draft-sparks-sip-mimetypes-03.txt
http://www.nostrum.com/~rjsparks/draft-sparks-sip-refer-split-00.txt
http://www.nostrum.com/~rjsparks/draft-sparks-sip-referredby-split-00.txt

Mimetypes is ready to go I think.

I propose resubmitting refer-split-00 as refer-03 and sending that 
through last call and to the IESG now.

We can then continue our discussion of referred-by in the 
referredby-split draft.


RjS

p.s. I will be online for a few more hours, and then only
intermittently until the interim meeting. 



_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr 26 08:40:31 2002
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Date: Fri, 26 Apr 2002 07:42:22 -0500
From: Chris Martin <cmartin@dasecurenetworks.com>
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To: steffen.fries@mchp.siemens.de, Michael Thomas <mat@cisco.com>,
        sip@ietf.org
Subject: Re: [Sip] SIP message integrity
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Chris Martin wrote:
> 
> Steffen Fries wrote:
> >
> > > How about internal office communications, once deemed secure, to a
> > > certain extent with the exception of a few PBX admins, which can now be
> > > eavesdropped on by anyone with a LAN connection a sniffer, and assuming
> > > enough knowledge to be dangerous in a switched environment, that can
> > > capture and replay media of a confidential nature such as high level
> > > conferencing.....
> > >
> > > This scenario actually fits the need for additional security
> > > protocols...but thats because I am concerned along those same lines.
> > >
> > Encryption of the media stream in corporate environments is
> > certainly an important requirement. It may also be interesting
> > for carrier grade solutions.
> >
> > > Along the lines of not wanting additional security protocols...besides
> > > issues related to dynamic pinholing, signaling modification for NAT/NAPT,
> > > bypassing intrusion detection systems is another area of concern. If
> > > certain signaling fields are encrypted unauthorized access can be masked.
> > > I think that encryption of the media stream and signaling authentication
> > > makes more sense.
> >
> > Does this mean you would favour signaling authentication rather
> > than signaling integrity? Is the reason for this the support of
> > NAT/NAPT?
> 
> No I would prefer both actually, but I am inclined to be one of the more
> paranoid.  :^)
> 
> But yes the reasoning for my comment is due to the fact that to permit
> signaling through sip enabled nat/napt, at a minimum the SDP must not be
> encrypted. That is not to say that I dont want full encryption....I do,
> but thats another discussion and a different topic.


I should clarify this further by stating that I would like to see full
encryption of signaling and media, but I dont want to hamper the efforts
of firewall vendors in their effort to facilitate SIP, which most have
done or are in process of doing. There are now at least 5 B2BUA style
firewall products and 3 application layer gateway style firewall vendors
with firewall solutions available or about to be available. There are
also about 5 other vendors providing other alternatives to firewalls to
address private address space and/or dynamic pinhole capability for SIP.
Many of these are well known vendors. 

Also many clients offer NAT features that would be useless with this
level of encryption initially.

It almost appears that to provide the level of integrity and
confidentiality that may be required in the future, the requirement
would be to provide a VPN capability in the client for media and
signaling and to proxy all such traffic to a carrier based VPN gateway
for decapsulation/encapsulation or between endpoints for the same
action. But then you get to the private address issues again and
circumvention of security policy (IDS cant detect unauthorized
communcations, firewalls cant enforce security policy via filters,
etc...).


> 
> >
> > Steffen
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Apr 26 08:43:04 2002
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Date: Fri, 26 Apr 2002 08:14:34 -0400
Subject: [Sip] I-D ACTION:draft-ietf-sip-digest-aka-01.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: HTTP Digest Authentication Using AKA
	Author(s)	: A. Niemi, J. Arkko, V. Torvinen
	Filename	: draft-ietf-sip-digest-aka-01.txt
	Pages		: 18
	Date		: 25-Apr-02
	
The Hypertext Transfer Protocol (HTTP) Authentication Framework
includes two authentication schemes: Basic and Digest.  Both schemes
employ a shared secret based mechanism for access authentication.
The Authentication and Key Agreement (AKA) mechanism performs user
authentication and session key distribution in Universal Mobile
Telecommunications System (UMTS) networks.  AKA is a challenge-
response based mechanism that uses symmetric cryptography.  This memo
specifies an AKA based one-time password generation mechanism for
HTTP Digest access authentication.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-digest-aka-01.txt

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	a MIME-compliant mail reader.  Different MIME-compliant mail readers
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From daemon@ns.ietf.org  Fri Apr 26 14:59:33 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: "'Rohan Mahy'" <rohan@cisco.com>,
        "'Brian Rosen'" <brian.rosen@marconi.com>,
        "'Joerg Ott'" <jo@ipdialog.com>, <jari.arkko@ericsson.com>
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I'd like to request final wg comments on the newly-renamed:

http://www.ietf.org/internet-drafts/draft-ietf-sip-sec-agree-00.txt


Jari Arkko will coordinate feedback. We hope to bump this to IETF Last
Call on April 30. Of course, one can still make comments even then, but
it's time we hatched this egg (do you eggree?).

--
Dean 






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From daemon@ns.ietf.org  Fri Apr 26 15:02:01 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
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Please finish digesting:

http://www.ietf.org/internet-drafts/draft-ietf-sip-digest-aka-01.txt


ss soon as possible. Aki will coordinate comments, I believe. We'd like
to send this to the IESG for IETF Last Call by April 30, if possible.

--
Dean



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From daemon@ns.ietf.org  Fri Apr 26 19:30:08 2002
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From: "Sriram Parameswar"<sriramp@nortelnetworks.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] REFER drafts
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Hi Robert,

Good job on the cutting out Referred-By from the main REFER draft. Small
nit:

1. draft-sparks-sip-referredby-split-00.txt section 3.3.2.2 - not sure why
we need to reproduce the multiple REFER case. Anyways if you prefer to keep
it, then please change 'cseq' -to-> 'id' in the Event header of flow F7.
2. The events-draft
(http://www.ietf.org/internet-drafts/draft-ietf-sip-events-05.txt) in
section 8.2 show that the header Subscription-State is mandatory in a NOTIFY
request. We never use this header in the two REFER drafts (refer-split and
referredby-split). It is my opinion that we have to have this header to be
in compliance with the events draft.

Regards,

Sriram
__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
Sent: Friday, April 26, 2002 6:07 AM
To: sip@ietf.org
Subject: [Sip] REFER drafts


I've just submitted three new refer related drafts.
Until they hit the repository, you can get them at

http://www.nostrum.com/~rjsparks/draft-sparks-sip-mimetypes-03.txt
http://www.nostrum.com/~rjsparks/draft-sparks-sip-refer-split-00.txt
http://www.nostrum.com/~rjsparks/draft-sparks-sip-referredby-split-00.txt

Mimetypes is ready to go I think.

I propose resubmitting refer-split-00 as refer-03 and sending that 
through last call and to the IESG now.

We can then continue our discussion of referred-by in the 
referredby-split draft.


RjS

p.s. I will be online for a few more hours, and then only
intermittently until the interim meeting. 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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------_=_NextPart_001_01C1ED75.9A0945A0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: [Sip] REFER drafts</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi Robert,</FONT>
</P>

<P><FONT SIZE=3D2>Good job on the cutting out Referred-By from the main =
REFER draft. Small nit:</FONT>
</P>

<P><FONT SIZE=3D2>1. draft-sparks-sip-referredby-split-00.txt section =
3.3.2.2 - not sure why we need to reproduce the multiple REFER case. =
Anyways if you prefer to keep it, then please change 'cseq' -to-&gt; =
'id' in the Event header of flow F7.</FONT></P>

<P><FONT SIZE=3D2>2. The events-draft (<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-events-05.txt=
" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-eve=
nts-05.txt</A>) in section 8.2 show that the header Subscription-State =
is mandatory in a NOTIFY request. We never use this header in the two =
REFER drafts (refer-split and referredby-split). It is my opinion that =
we have to have this header to be in compliance with the events =
draft.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
<BR><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Robert Sparks [<A =
HREF=3D"mailto:rsparks@dynamicsoft.com">mailto:rsparks@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Friday, April 26, 2002 6:07 AM</FONT>
<BR><FONT SIZE=3D2>To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] REFER drafts</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>I've just submitted three new refer related =
drafts.</FONT>
<BR><FONT SIZE=3D2>Until they hit the repository, you can get them =
at</FONT>
</P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.nostrum.com/~rjsparks/draft-sparks-sip-mimetypes-03.t=
xt" =
TARGET=3D"_blank">http://www.nostrum.com/~rjsparks/draft-sparks-sip-mime=
types-03.txt</A></FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.nostrum.com/~rjsparks/draft-sparks-sip-refer-split-00=
.txt" =
TARGET=3D"_blank">http://www.nostrum.com/~rjsparks/draft-sparks-sip-refe=
r-split-00.txt</A></FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.nostrum.com/~rjsparks/draft-sparks-sip-referredby-spl=
it-00.txt" =
TARGET=3D"_blank">http://www.nostrum.com/~rjsparks/draft-sparks-sip-refe=
rredby-split-00.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>Mimetypes is ready to go I think.</FONT>
</P>

<P><FONT SIZE=3D2>I propose resubmitting refer-split-00 as refer-03 and =
sending that </FONT>
<BR><FONT SIZE=3D2>through last call and to the IESG now.</FONT>
</P>

<P><FONT SIZE=3D2>We can then continue our discussion of referred-by in =
the </FONT>
<BR><FONT SIZE=3D2>referredby-split draft.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>RjS</FONT>
</P>

<P><FONT SIZE=3D2>p.s. I will be online for a few more hours, and then =
only</FONT>
<BR><FONT SIZE=3D2>intermittently until the interim meeting. </FONT>
</P>
<BR>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

</BODY>
</HTML>
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_______________________________________________
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: HTTP Digest Authentication Using AKA
	Author(s)	: A. Niemi, J. Arkko, V. Torvinen
	Filename	: draft-ietf-sip-digest-aka-01.txt
	Pages		: 18
	Date		: 25-Apr-02
	
The Hypertext Transfer Protocol (HTTP) Authentication Framework
includes two authentication schemes: Basic and Digest.  Both schemes
employ a shared secret based mechanism for access authentication.
The Authentication and Key Agreement (AKA) mechanism performs user
authentication and session key distribution in Universal Mobile
Telecommunications System (UMTS) networks.  AKA is a challenge-
response based mechanism that uses symmetric cryptography.  This memo
specifies an AKA based one-time password generation mechanism for
HTTP Digest access authentication.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-digest-aka-01.txt

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From daemon@optimus.ietf.org  Sat Apr 27 06:38:26 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <brian.rosen@marconi.com>, <jo@ipdialog.com>,
        "'Allison Mankin'" <mankin@isi.edu>,
        "'Duncan Mills'" <Dunc.Mills@btinternet.com>
Date: Sat, 27 Apr 2002 05:06:10 -0500
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Subject: [Sip] Please review draft-mills-sip-access-network-info (P-headers)
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Another P-header draft has been sent to internet-drafts and should be
available soon. In the meantime, please use:

http://www.softarmor.com/sipwg/drafts/draft-mills-sip-access-network-inf
o-00.txt


This is an individual informational draft being "expert reviewed" by SIP
WG under the sip-change process. We need to get it in final format for
the RFC editor very soon.

In additional to technical review, we need to hep assure that the
document meets the standards for IETF publication, things like format,
references, etc. -- the "nits review".

Please copy the editor, Duncan Mills <dunc.mills@btinternet.com>, on
feedback.

Thanks, y'all.

--
Dean


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From daemon@optimus.ietf.org  Sat Apr 27 06:43:44 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <Rohan@cicso.com>, <brian.rosen@cisco.com>,
        "'Allison Mankin'" <mankin@isi.edu>, <jo@ipdialog.com>,
        <gonzallo.camarillo@ericsson.com>
Date: Sat, 27 Apr 2002 05:25:18 -0500
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Subject: [Sip] Quick WGLC,  Reason draft
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We need to process the Reason draft. The UPDATE method is dependent on
it.

http://search.ietf.org/internet-drafts/draft-ietf-sip-reason-00.txt

Editor Gonzalo Camarillo will coordinate responses -- please copy him.

Thanks,

--
Dean Willis


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Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Mon Apr 29 05:07:18 2002
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From: Tan Ya-Ching  ICM N PG U ID A 1 <Ya-Ching.Tan@icn.siemens.de>
To: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Cc: 3GPP_TSG_CN_WG1 <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Subject: RE: [Sip] Revised Path draft submitted
Date: Mon, 29 Apr 2002 10:30:49 +0200
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Some comments on draft-willis-sip-path-03 :

The draft states that any proxy processing a REGISTER request can add its
own URI or a Path element referencing another node to the REGISTER message.
For the response to the REGISTER, it states that "The registrar reflects the
accumulated Path back into the REGISTER response, and intermediate nodes
propogate this back toward the originating UA". There is no mention of how a
proxy can add or modify the Path headers in the response. But in the
addition of Path to SIP Table 2, the Path in 2xx is marked as "amr", where a
= add and m = modify. Is this a typo ?

The Path header is supposed to have a "syntax very similar to the
Record-Route header". SIP bis-09 states that "The URI placed in the
Record-Route header field value MUST be a SIP URI. This URI  MUST contain an
lr parameter". I think this should be a requirement for the Path header too
and should be mentioned in the draft.

Two minor side comments :

- The addition of Path to SIP Table 2 should be to Table 3 instead. Table 3
is for header fields starting with P-Z.

- Path header should be added to Table 1 of bis-09 too.

- In the example message flow, the ordering of the two Path elements in F6
should be reversed.

-------
Ya-Ching Tan


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From daemon@ns.ietf.org  Mon Apr 29 05:07:32 2002
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From: Tan Ya-Ching  ICM N PG U ID A 1 <Ya-Ching.Tan@icn.siemens.de>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
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Subject: Recall: [Sip] Revised Path draft submitted
Date: Mon, 29 Apr 2002 10:33:10 +0200
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Tan Ya-Ching  ICM N PG U ID A 1 would like to recall the message, "[Sip]
Revised Path draft submitted".

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From daemon@ns.ietf.org  Mon Apr 29 05:37:05 2002
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From: Tan Ya-Ching  ICM N PG U ID A 1 <Ya-Ching.Tan@icn.siemens.de>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: [Sip] Revised Path draft vs the revised Service Route Discovery d
	raft
Date: Mon, 29 Apr 2002 11:10:42 +0200
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Hi folks,

Sorry about the confusion. I'm a bit behind on the SIP list and just read
the draft-willis-sip-path-03 and commented on it. Then I discovered the
revised service route discovery draft and thought that it was supposed to
replace the Path draft. However, it seems that the Path header draft still
stands according to the following mail. Or does it not ?



> -----Original Message-----
> From: ext AC Mahendran [mailto:mahendra@qualcomm.com]
> Sent: 22 April, 2002 19:49
> To: Dean Willis; sip@ietf.org
> Cc: 3GPP_TSG_CN_WG1
> Subject: Re: [Sip] Revised Service Route Discovery Draft
> 
> Also, doesn't the "Path" header provide the same functionality as the 
> "P-Service-Route" ?

In IETF SIP WG is was impossible to reach consensus on using
Path for both - originated and terminated - session cases.
Thus, the Path header will be used for terminated session
cases only, whereas the P-Service-Route header only affects
the originated session cases.



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Subject: RE: [Sip] z9hG4bK forever ???
Date: Mon, 29 Apr 2002 10:26:14 +0100
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

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The special arragement option depends on how the network operator deals with
providing it.
 
If the network operator gives it to all and everyone, then it is as you say.
Essentially as the From field. This requires no further consideration.
 
When it was specified however, one major use was identified as being from
enterprise networks, in fact they were the main proponents of such a usage.
If one follows the private network ISDN signalling specifications (ISO PSS1
and ECMA QSIG) then enterprise delivered values are screened in the
enterprise network, thus as far as the enterprise network is concerned, they
are regarded as network asserted rather than as user asserted. The only
reason for the special arrangement coding (user provided unscreened) in this
case is that the public network operators wanted a distinction between
public network asserted identities and private network asserted identities. 
 
So far the point is therefore that while enterprise networks can and do
assert identities, they may not be fully trusted. In the ISDN, it is trusted
sufficiently in order to deliver the CLIP service to the remote user, but is
not trusted sufficiently for malicious call identification. If we are to
achieve 100% mapping to the ISDN/PSTN, then we may need to reflect this
distinction.
 
I certainly do not believe that we should treat the unscreened coding in all
cases as mapping to the From header, as this will not give a 100% mapping in
the reverse direction. However, in the longer term, providing some values
indicating who provided the assertion may resolve this problem. In this case
dummy values such as "non-public network operator" could be used as the
asserting identity when such values are mapped.
 
In the short term, as a large number of public network operators do not
allow the special arrangement to exist, it may be best to ignore the special
arrangement, or at least the finer details of the mapping.
 
Keith
 
 

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: 18 April 2002 12:00
To: 'Dean Willis'; William Marshall; sip@ietf.org
Subject: RE: [Sip] z9hG4bK forever ???



I think we're about at the end of this one. 

Two final points from me: 
1) Dean's desire to avoid having the From: field mangled unless he has
specifically requested it is reasonable. I still think  that the
subscriber's wishes could overrule this though.

This is no different from the ISDN 'Special Arrangement' where the user
inserts any number they like in the Calling Party Number field. The network
passes it though with no checking, changing, nothing - I think this is
analogous to what Dean wants for the From field.

But the network still removes this information if the subscriber wants
anonymity. The stated purpose of the field is user identity, which implies
Personal Data of the subscriber. This is enough to argue that the network
must remove it if privacy is required, and this is certainly the application
of current legislation to ISDN.

What is the stated purpose of the From field - I do not think it is
sufficiently different for us to be _sure_ that we can avoid the
requirement.

So, the munging may still be required, but I certainly agree it should be
avoided wherever possible. 

2) In formal terms, 'B2BUA == NOT Proxy'. We all know there will be/are
devices which behave very much like a proxy, but perform some additional
functions, such as these privacy services. This is fine. Everyone should be
aware that with only two terms defined 'B2BUA' and 'Proxy', such devices
fall into the B2BUA category.

...Mark 

> -----Original Message----- 
> From: Dean Willis [ mailto:dean.willis@softarmor.com
<mailto:dean.willis@softarmor.com> ] 
> Sent: 17 April 2002 17:36 
> To: William Marshall; sip@ietf.org 
> Subject: Re: [Sip] z9hG4bK forever ??? 
> 
> 
> Baill said: 
> > All this discussion of proxy modification of the From and 
> To headers, 
> > and the need for a B2BUA, seems to me to be a short-term 
> problem only. 
> > 
> > RFC3261 (2543bis-09) specifies matching of dialog-ids based only 
> > on Call-ID and tags, not on display-name or on addr-spec.  So 
> > when all endpoints are compliant with RFC3261, there will be no 
> > problem with a proxy modifying the display-name or addr-spec in 
> > the From header. 
> 
> I disagree. I don't want the network mangling my user-to-user headers 
> unless I've specifically requested it to do so. Specifically, the 
> non-authenticated, non-mangled, non-protected From: field 
> offers utility 
> which is DIFFERENT than that provided by network authenticated calling 
> party ID, and I don't want to sacrifice that utility just because this 
> is a capability the widely-deployed PSTN (as opposed to say Q.SIG) 
> didn't have . . . 
> 
> -- 
> Dean 
> 
> 
> 
> 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
<https://www1.ietf.org/mailman/listinfo/sip>  
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> 


------_=_NextPart_001_01C1EF5F.E887F3C0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>RE: [Sip] z9hG4bK forever ???</TITLE>

<META content="MSHTML 5.50.4915.500" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>The 
special arragement option depends on how the network operator deals with 
providing it.</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>If the 
network operator gives it to all and everyone, then it is as you say. 
Essentially as the From field. This requires no further 
consideration.</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>When 
it was specified however, one major use was identified as being from enterprise 
networks, in fact they were the main proponents of such a usage. If one follows 
the private network ISDN signalling specifications (ISO PSS1 and ECMA QSIG) then 
enterprise delivered values are screened in the enterprise network, thus as far 
as the enterprise network is concerned, they are regarded as network asserted 
rather than as user asserted. The only reason for the special arrangement coding 
(user provided unscreened) in this case is that the public network operators 
wanted a distinction between public network asserted identities and private 
network asserted identities. </FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>So far 
the point is therefore that while enterprise networks can and do assert 
identities, they may not be fully trusted. In the ISDN, it is trusted 
sufficiently in order to deliver the CLIP service to the remote user, but is not 
trusted sufficiently for malicious call identification. If we are to achieve 
100% mapping to the ISDN/PSTN, then we may need to reflect this 
distinction.</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>I 
certainly do not believe that we should treat the unscreened coding in all cases 
as mapping to the From header, as this will not give a 100% mapping in the 
reverse direction. However, in the longer term, providing some values indicating 
who provided the assertion may resolve this problem. In this case dummy values 
such as "non-public network operator" could be used as the asserting identity 
when such values are mapped.</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff size=2>In the 
short term, as a large number of public network operators do not allow the 
special arrangement to exist, it may be best to ignore the special arrangement, 
or at least the finer details of the mapping.</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2>Keith</FONT></SPAN></DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=087053013-22042002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> 18 April 2002 
  12:00<BR><B>To:</B> 'Dean Willis'; William Marshall; 
  sip@ietf.org<BR><B>Subject:</B> RE: [Sip] z9hG4bK forever 
  ???<BR><BR></FONT></DIV>
  <P><FONT size=2>I think we're about at the end of this one.</FONT> </P>
  <P><FONT size=2>Two final points from me:</FONT> <BR><FONT size=2>1) Dean's 
  desire to avoid having the From: field mangled unless he has specifically 
  requested it is reasonable. I still think&nbsp; that the subscriber's wishes 
  could overrule this though.</FONT></P>
  <P><FONT size=2>This is no different from the ISDN 'Special Arrangement' where 
  the user inserts any number they like in the Calling Party Number field. The 
  network passes it though with no checking, changing, nothing - I think this is 
  analogous to what Dean wants for the From field.</FONT></P>
  <P><FONT size=2>But the network still removes this information if the 
  subscriber wants anonymity. The stated purpose of the field is user identity, 
  which implies Personal Data of the subscriber. This is enough to argue that 
  the network must remove it if privacy is required, and this is certainly the 
  application of current legislation to ISDN.</FONT></P>
  <P><FONT size=2>What is the stated purpose of the From field - I do not think 
  it is sufficiently different for us to be _sure_ that we can avoid the 
  requirement.</FONT></P>
  <P><FONT size=2>So, the munging may still be required, but I certainly agree 
  it should be avoided wherever possible.</FONT> </P>
  <P><FONT size=2>2) In formal terms, 'B2BUA == NOT Proxy'. We all know there 
  will be/are devices which behave very much like a proxy, but perform some 
  additional functions, such as these privacy services. This is fine. Everyone 
  should be aware that with only two terms defined 'B2BUA' and 'Proxy', such 
  devices fall into the B2BUA category.</FONT></P>
  <P><FONT size=2>...Mark</FONT> </P>
  <P><FONT size=2>&gt; -----Original Message-----</FONT> <BR><FONT size=2>&gt; 
  From: Dean Willis [<A 
  href="mailto:dean.willis@softarmor.com">mailto:dean.willis@softarmor.com</A>]</FONT> 
  <BR><FONT size=2>&gt; Sent: 17 April 2002 17:36</FONT> <BR><FONT size=2>&gt; 
  To: William Marshall; sip@ietf.org</FONT> <BR><FONT size=2>&gt; Subject: Re: 
  [Sip] z9hG4bK forever ???</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; </FONT><BR><FONT size=2>&gt; Baill said:</FONT> <BR><FONT 
  size=2>&gt; &gt; All this discussion of proxy modification of the From and 
  </FONT><BR><FONT size=2>&gt; To headers,</FONT> <BR><FONT size=2>&gt; &gt; and 
  the need for a B2BUA, seems to me to be a short-term </FONT><BR><FONT 
  size=2>&gt; problem only.</FONT> <BR><FONT size=2>&gt; &gt;</FONT> <BR><FONT 
  size=2>&gt; &gt; RFC3261 (2543bis-09) specifies matching of dialog-ids based 
  only</FONT> <BR><FONT size=2>&gt; &gt; on Call-ID and tags, not on 
  display-name or on addr-spec.&nbsp; So</FONT> <BR><FONT size=2>&gt; &gt; when 
  all endpoints are compliant with RFC3261, there will be no</FONT> <BR><FONT 
  size=2>&gt; &gt; problem with a proxy modifying the display-name or addr-spec 
  in</FONT> <BR><FONT size=2>&gt; &gt; the From header.</FONT> <BR><FONT 
  size=2>&gt; </FONT><BR><FONT size=2>&gt; I disagree. I don't want the network 
  mangling my user-to-user headers</FONT> <BR><FONT size=2>&gt; unless I've 
  specifically requested it to do so. Specifically, the</FONT> <BR><FONT 
  size=2>&gt; non-authenticated, non-mangled, non-protected From: field 
  </FONT><BR><FONT size=2>&gt; offers utility</FONT> <BR><FONT size=2>&gt; which 
  is DIFFERENT than that provided by network authenticated calling</FONT> 
  <BR><FONT size=2>&gt; party ID, and I don't want to sacrifice that utility 
  just because this</FONT> <BR><FONT size=2>&gt; is a capability the 
  widely-deployed PSTN (as opposed to say Q.SIG)</FONT> <BR><FONT size=2>&gt; 
  didn't have . . .</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; 
  --</FONT> <BR><FONT size=2>&gt; Dean</FONT> <BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; </FONT><BR><FONT size=2>&gt; 
  _______________________________________________</FONT> <BR><FONT size=2>&gt; 
  Sip mailing list&nbsp; <A target=_blank 
  href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</A></FONT> 
  <BR><FONT size=2>&gt; This list is for NEW development of the core SIP 
  Protocol</FONT> <BR><FONT size=2>&gt; Use sip-implementors@cs.columbia.edu for 
  questions on current sip</FONT> <BR><FONT size=2>&gt; Use sipping@ietf.org for 
  new developments on the application of sip</FONT> <BR><FONT size=2>&gt; 
  </FONT></P></BLOCKQUOTE></BODY></HTML>

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	Title		: Internet Media Types message/sipfrag
	Author(s)	: R. Sparks
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	Pages		: 6
	Date		: 26-Apr-02
	
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This document defines the private SIP extension header
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: The Referred-By Header Field
	Author(s)	: R. Sparks
	Filename	: draft-sparks-sip-referredby-split-00.txt
	Pages		: 10
	Date		: 26-Apr-02
	
This document defines the Referred-By header field.  This SIP
extension is used with REFER to carry information about the Referror
to the resource indicated by the Refer-To URI when that URI is a SIP
URI.

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http://www.ietf.org/internet-drafts/draft-sparks-sip-referredby-split-00.txt

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	Title		: The Refer Method
	Author(s)	: R. Sparks
	Filename	: draft-sparks-sip-refer-split-00.txt
	Pages		: 16
	Date		: 26-Apr-02
	
This document defines the REFER method.  This SIP extension requests
that the recipient REFER to a resource provided in the request.  This
can be used to enable many applications, including Call Transfer.

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From daemon@optimus.ietf.org  Mon Apr 29 08:38:53 2002
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From: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
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To: sip@ietf.org, Dean Willis <dwillis@dynamicsoft.com>
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Subject: [Sip] 3 I-Ds about P- headers
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Hi:

I have just submitted 3 new internet drafts about 3 new P- headers.
They will probably be available in the IETF repository in the next few days. In the meantime, you can download them from:

http://standards.ericsson.net/sip/drafts/draft-garcia-sip-visited-network-id-00.txt

http://standards.ericsson.net/sip/drafts/draft-garcia-sip-called-party-id-00.txt

http://standards.ericsson.net/sip/drafts/draft-garcia-sip-associated-uri-00.txt

Regards,

       Miguel
-- 
Miguel-Angel Garcia                     Oy LM Ericsson AB
                                        Jorvas, Finland
mailto:Miguel.A.Garcia@ericsson.com     Phone:  +358 9 299 3553
mailto:Miguel.A.Garcia@piuha.net        Mobile: +358 40 5140002

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From daemon@optimus.ietf.org  Mon Apr 29 08:46:10 2002
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From: Tan Ya-Ching  ICM N PG U ID A 1 <Ya-Ching.Tan@icn.siemens.de>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>,
        "'AC Mahendran'"
	 <mahendra@qualcomm.com>, sip@ietf.org
Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Mon, 29 Apr 2002 14:23:28 +0200
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| -----Original Message-----
| From: Dean Willis [mailto:dwillis@dynamicsoft.com]
| Sent: 24 April 2002 12:33
| 
| Well, the return code of the Path MAY be similar to the value of
| P-Service-Route. It might not. The Path is an exact subset of the
| traversal path of the REGISTER request. 

No, it needs not be. draft-willis-sip-path-03 states that "It is also
possible for a proxy with specific knowledge of network topology to add a
path element referencing another node, thereby allowing construction of a
Path which is discongruent with the route taken by the REGISTER request".

It is not clear to me whether the Path header and the P-Service-Route header
can co-exist in a 200 OK response to a REGISTER. If Path header exists in
the REGISTER request, the Path draft expects the registrar to reflect
accumulated Path back into the REGISTER response. The Service-Route draft on
the other hand says that the registrar may choose to return a
P-Service-Route header in any successful 200 OK response. Should the
registrar not return the Path if it wants to insert the Service-Route ?

Regards,
Ya-Ching

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From daemon@optimus.ietf.org  Mon Apr 29 09:58:38 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: "'Tan Ya-Ching  ICM N PG U ID A 1'" <Ya-Ching.Tan@icn.siemens.de>,
        "'AC Mahendran'" <mahendra@qualcomm.com>, <sip@ietf.org>
Cc: <bernhard.honeisen@nokia.com>
Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Mon, 29 Apr 2002 08:21:55 -0500
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Tan Ya-Ching said:
> Dean Said:
> | -----Original Message-----
> | From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> | Sent: 24 April 2002 12:33
> | 
> | Well, the return code of the Path MAY be similar to the value of 
> | P-Service-Route. It might not. The Path is an exact subset of the 
> | traversal path of the REGISTER request.
> 
> No, it needs not be. draft-willis-sip-path-03 states that "It 
> is also possible for a proxy with specific knowledge of 
> network topology to add a path element referencing another 
> node, thereby allowing construction of a Path which is 
> discongruent with the route taken by the REGISTER request".

Ah! Thanks for pointing that out. That paragraph is left over from an
earlier version of the draft, and needs to be deleted. My fault. Thanks
for the careful reading!

 
> It is not clear to me whether the Path header and the 
> P-Service-Route header can co-exist in a 200 OK response to a 
> REGISTER. If Path header exists in the REGISTER request, the 
> Path draft expects the registrar to reflect accumulated Path 
> back into the REGISTER response. The Service-Route draft on 
> the other hand says that the registrar may choose to return a 
> P-Service-Route header in any successful 200 OK response. 
> Should the registrar not return the Path if it wants to 
> insert the Service-Route ?

They may both be present.

--
Dean


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From daemon@optimus.ietf.org  Mon Apr 29 10:10:33 2002
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Subject: [Sip] Regarding sending of Min-SE header in the re-INVITE
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Hello
     I have doubt regarding the following  statement of the
draft-ietf-sip-session-timer-08.txt

"The UAC MUST insert the Min-SE header into a re-INVITE request for a
particular call leg if it has ever received a 422 response to a   previous
INVITE on the same leg"

When a UAC receives 422 response for a INVITE request, it retries with by
removing the tag field.
So once the call is established, there will not be any Min-SE for that call
leg.

The example in draft also says same thing.

What value in the Min-SE header should go out ?.


The draft also specifes

"In an INVITE that is   within a call leg with an active session timer, the
header SHOULD be   present, and SHOULD contain the the larger of the
current value of   the session interval and the value of the Min-SE, if
present"

So how can be the re-INVITE get 422 response, so that next re-INVITE will
go out with Min-SE header.


Any replies,suggestions,pointers in this regard would be helpful.

Regds
Pankaj Nath







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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Subject: Re: [Sip] Quick WGLC,  Reason draft
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Dean Willis wrote:

> We need to process the Reason draft. The UPDATE method is dependent on
> it.
> 
> http://search.ietf.org/internet-drafts/draft-ietf-sip-reason-00.txt
> 
> Editor Gonzalo Camarillo will coordinate responses -- please copy him.


Gonzalo:

Some comments, divided into nits and non-nits comments.

Non-nits comments
-----------------
1) The I-D says that:
    "Initially, the Reason header field defined here appears to be
     most useful for BYE and CANCEL requests, but it can appear in
     any request"

    I seem to recall on the list that the Reason header field was limited
    to requests within a dialog only.  Is this restriction no longer
    applicable.  The I-D implies that a request generated, for instance,
    by a 3PCC controller establishing a session with party B may contain
    a Reason header.

2) If I am not mistaken, the ABNF restricts the Reason field to appear
    only once.  Since it can appear multiple times in a SIP message (and
    be separable by commas), we should change the ABNF to reflect this.

Nits comments
-------------
1) The ABNF of Reason sets the Protocol field to:
    Protocol = SIP / Q.850 / token
    but the paragraph under the ABNF states that the value "SIP/2.0" has
    been defined for a SIP status code.  However, the examples still
    contain the value "SIP", not "SIP/2.0".  To be consistent, we should
    either mention the value "SIP" in the paragraph below the ABNF, or
    change the ABNF and examples to contain "SIP/2.0".

2) The following paragraph appears run-on:
    "When an INVITE is rejected, not because the call is declined,
    but because some aspect of the request was not acceptable, if
    the INVITE was forked, the error response is not forwarded
    towards the UAC by the forking proxy. This problem is known
    as..."
    Suggested modifications:
    "An INVITE can sometimes be rejected not because the session
    initiation was declined, but becuause some aspect of the request
    was not acceptable; for instance, if the INVITE forked and
    resulted in a rejection, which may never be forwarded to the
    client unless all the other branches also reject the request.
    This problem is known as ..."

3) The four examples in Section 2 correspond roughly to the examples
    of section 3.  Maybe they should be repeated (or presented for
    the first time) in the relevant sub-sections of Section 3.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>,
        "'Tan Ya-Ching  ICM N PG U ID A 1'" <Ya-Ching.Tan@icn.siemens.de>,
        "'AC Mahendran'" <mahendra@qualcomm.com>, <sip@ietf.org>
Cc: <bernhard.honeisen@nokia.com>
Subject: RE: [Sip] Revised Service Route Discovery Draft
Date: Mon, 29 Apr 2002 10:22:21 -0500
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Ok, it's apparently my day for making stupid responses. I hope it's not
becoming a habit.

Please allow me to clarify what I meant to say (as opposed to what I
actually said, which was wrong).

A proxy adding a Path element to a REGISTER request MAY construct that
element so that it actually refers to a different proxy. It's actually
possible to do something similar in Record-Route, although I'm not aware
of any systems that have actually implemented this. 

So it is not technically correct to say that the Path is an exact subset
of traversal path of the RESGISTER. Rather, it is "usually" an exact
subset of the traversal path of the REGISTER.

But this is still very different from the value returned by
P-Service-Route.

From the UA's perspective, the Path response means "These are the
proxies that have been added to my registration, such that messages
going from my home proxy to me will visit these proxies". It deals with
UA-terminated messages, i.e., when the UA us acting as a UAS.

From the UA's perspective, the P-Service-Route response means "This is a
route set that I can combine with other routing knoweldge I may have so
that I can get services from a particular set of service proxies." It
deals with UA-originated messages, i.e., when the UA is acting as a UAC.

--
Dean


> Tan Ya-Ching said:
> > Dean Said:
> > | -----Original Message-----
> > | From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> > | Sent: 24 April 2002 12:33
> > | 
> > | Well, the return code of the Path MAY be similar to the value of
> > | P-Service-Route. It might not. The Path is an exact subset of the 
> > | traversal path of the REGISTER request.
> > 
> > No, it needs not be. draft-willis-sip-path-03 states that "It
> > is also possible for a proxy with specific knowledge of 
> > network topology to add a path element referencing another 
> > node, thereby allowing construction of a Path which is 
> > discongruent with the route taken by the REGISTER request".
> 
> Ah! Thanks for pointing that out. That paragraph is left over 
> from an earlier version of the draft, and needs to be 
> deleted. My fault. Thanks for the careful reading!


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From daemon@ns.ietf.org  Mon Apr 29 12:15:14 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Subject: [Sip] Timer C, Timer B, and Spiraling
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Some comments on -09bis on Timers C, B, and spiraling:

1) In SIP, proxies can fork either in serial or in parallel.  Consider
    serial forking for this discussion.  Normally, in the presence of
    multiple Contact headers to an address-of-record, a proxy will sort
    the Contacts based on their q value and fork serially to each
    Contact.  If Contact 1 did not elicit a final response within a
    certain time, Contact 2 will be tried, and so on.  Problem is,
    what exactly is the time for a proxy to give up on the current
    Contact and try the next one?

    Reading -09bis, it appears that timer C can serve this function.
    But the initial value of timer C is too high (3 minutes, and it is
    a MUST to set it to at least 3 mins), so that will not work.  Another
    possible candidate is the transaction timeout; if a proxy allows a
    client transaction to continue (T1*64 seconds = 32 seconds) until it
    times out, the time between trying the next Contact (32 seconds) is
    too large for the caller to keep holding.  So, this does not work
    either.

    Should -09bis provide some guidance on timers to use in proxy cores
    if serial forking is being used and > 1 Contact header is present,
    or should this be left upto the implementer?

2) Timer B controls transaction timeouts in INVITE client trans-
    actions.  The state diagram in Figure 5 is silent on what to
    do if Timer B fires while control is in "Proceeding" state; i.e.
    received a 180, but no final response.  The correct behavior
    should be to sent a CANCEL and enter "Terminated" state.

3) Regarding finding the spiraled R-R on responses in order to
    modify it (section 16.7, step 8), won't it be the case that
    the first R-R in a response corresponds to the inner-most
    spiral?  Since the inner-most spiral will generate the first
    response, can't we just get the first R-R header on a response
    in order to modify it?

If I mis-represented something and the solution is in -09bis, please
let me know.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and eServices
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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Subject: Re: [Sip] I-D ACTION:draft-mills-sip-access-network-info-00.txt
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Duncan:

Section 6, paragraph 2 says:
"The P-Access-Network-Info header MAY be inserted by a UA. Proxies, MUST 
NOT add to or modify the contents of the P-Access Network-Info".

However, in the formal definition of "P-Access-Network-Info" we have the 
values of "acm" for proxies. This contradicts the previous statement.

thanks,
AC

At 07:52 AM 4/29/2002 -0400, Internet-Drafts@ietf.org wrote:
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>
>
>         Title           : The SIP Access Network Info header
>         Author(s)       : D. Mills
>         Filename        : draft-mills-sip-access-network-info-00.txt
>         Pages           :
>         Date            : 26-Apr-02
>
>This document defines the private SIP extension header
>P-Access-Network-Info. This mechanism is useful in SIP networks that
>are partitioned, such as 3G wireless networks which are partitioned at
>the SIP layer into 'access' and 'home' networks. SIP User Agents may
>use this header to relay information about the access network to
>serving proxies in their home network. The serving proxy may then use
>this information to optimize services for the UA. For example, a 3GPP
>terminal uses this header to pass information about the access network
>such as radio access technology and cell ID to its home service.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-mills-sip-access-network-info-00.txt
>
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From: "Dean Willis" <dean.willis@softarmor.com>
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Cc: "3GPP_TSG_CN_WG1" <3GPP_TSG_CN_WG1@LIST.ETSI.FR>, <amankin@isi.edu>,
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Subject: [Sip] Revised P-Service-Route header draft, please review
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I've submitted -02 of the P-Service-Route header to internet-drafts. In
the meantime, you can get it from:

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.txt
http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.htm
l
http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.xml

I think this is getting fairly close to final. Please eyeball it
carefully.

This draft is an individual informational documenting a P-header
requested by 3GPP. We'll be processing it by "expert review". Current
plans require review to be complete by May 10 or thereabouts.

Thanks,

--
Dean Willis



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From daemon@ns.ietf.org  Mon Apr 29 13:29:41 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>, "3GPP_TSG_CN_WG1" <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Cc: <bernhard.honeisen@nokia.com>, <amankin@isi.edu>,
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        <brian.rosen@marconi.com>
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I've sent -04 into internet-drafts. In the meantime, please review from:

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.txt
http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.html
http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.xml

We'll probably be calling for WGLC "very soon".

Thanks,

--
Dean


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From daemon@ns.ietf.org  Mon Apr 29 14:07:32 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: "'Rohan Mahy'" <rohan@cisco.com>, <jo@ipdialog.com>,
        <brian.rosen@marconi.com>, <mankin@isi.edu>
Date: Mon, 29 Apr 2002 12:49:12 -0500
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Subject: [Sip] Expert Review for P-Headers Drafts
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You've already seen a few and will see many more requests for us to
provide expert review for P-header drafts.

P-headers, if you haven't been following, are "private, preliminary, or
proprietary" extension headers. They are published as informational
drafts, general as individual contributions. The process is set out in
draft-tsv-sipchange and is based on the process defined in RFC 2026 for
informational documents.

Expert Review is basically the equivalent of a Working Group Last Call
for this sort of draft. It should take a couple of weeks, and may
involve multiple updates of the document. As always, getting feedback to
the editors as soon as reasonably possible is a Good Thing.

Our tasks in reviewing these include:

1) Establish that the document is "publication grade" for IETF -- in the
right format, properly referenced, and effectively using requirements
language. Bascially, a "nits review".

2) Detect significant overlaps with chartered general purpose work. In
general, we wish to avoid P-headers that overlap with some problem we've
already identified and are working on solving, especially if the general
solution is near-term.

3) Detect serious security issues. P-headers MAY proceed with security
questions, but we need to make sure that the issues are adequately
documented and will probably propose fixes.

4) Look for underlying generalities that we may use to guide our work
going forward. We may notice multiple occurrences of certain clases of
problems cropping up, which often means that there is some underlying
more general problem which if solved would produce broad benefits.

Thanks,

--
Dean Willis


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From daemon@ns.ietf.org  Mon Apr 29 14:17:15 2002
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Date: Mon, 29 Apr 2002 10:52:17 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
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Hi,

I just submitted an update to replaces.  This draft has a substantial
change, in that it specifies Proxy behavior for replacing early dialogs.
(The proxy behavior is only pertinent to forking proxies).  The proposed
mechanism should work, but it is complex.  We can either:

a) leave the proposed mechanism in the draft,
b) explicitly choose not to address call control for early dialogs, or
c) split out matching of transactions at forking proxies in another draft

I invite your comments on this (hopefully last) open issue.

Until it appears in the archive, you can fetch a copy from the
supplemental web site.  An html version is also available.

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-replaces-01.txt

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-replaces-01.html

many thanks,
-rohan



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From: "Vijay K. Gurbani" <vkg@lucent.com>
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CC: sip@ietf.org, 3GPP_TSG_CN_WG1 <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Subject: Re: [Sip] Revised Path draft (-04) submitted
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Dean Willis wrote:

> I've sent -04 into internet-drafts. In the meantime, please review from:
> 
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.txt
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.html
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-path-04.xml


Dean:

Some comments:

Section 1:
    "UA wishes to register with REGISTRAR. However, due to network
    topology, UA must use P1 as an "outbound proxy", and all messages
    between UA1 and REGISTRAR must also traverse P1, P2, and P3 before
            ^^^
    reaching REGISTRAR."

There isn't any UA1, only "UA".  To be consistent with Section 4.5 and
4.6, you may want to re-label "UA" in the figure to "UA1".

In the sentence,

    "Eventually, REGISTRAR or a service proxy closely related to it
    will receive a message for UA. It needs to know which proxies
    must be transited by that message in order to get back to UA..."

we are talking about *request* initiated by the REGISTRAR destined to
UA, right?  If so, can't we reword it as:

    "Eventually, REGISTRAR or a service proxy closely related to it
    will need to send a request to the UA.  It needs to know which
    proxies must be transited by that request in order to get to UA..."

Generally speaking, in many places where you use "message", I believe
you mean "requests".  If so, can we substitute the term request, since
a "message" may be a response, for instance, which will follow the
Via paths.

In Section 4.6, the requests from P1->P2 show P1 adding itself to
the Path list (initially set to null).  The branch of P1 does NOT
contain the "magic cookie" (z9hG4bK).  Likewise for P3->REGISTRAR.
Since P1 and P3 supports "lr", it is a good bet that they also support
the Via "magic cookie".  Of course, this is an obscure nit, but all the
same...

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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To: Dean Willis <dean.willis@softarmor.com>
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Dean Willis wrote:

> I've submitted -02 of the P-Service-Route header to internet-drafts. In
> the meantime, you can get it from:
> 
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.txt
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.htm
> l
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-svcrtdisco-02.xml
> 
> I think this is getting fairly close to final. Please eyeball it
> carefully.


Dean:

Some comments:

Section 1, bullet 5:
    5. Performing an operation equivalent to record-routing in a
       REGISTER transaction between the UA and the associated
       registrar, then storing that route in the UA and reusing it
       as a service route on future messages originating from the UA.
       While efficient, this constrains the service route for proxy
       operations to be congruent with the route taken by the REGISTER
       message.

I interpret the last sentence as follows:
    "While efficient, this constrains the service route for proxy
     operations to be congruent with the Route set formed at the UA
     from the Path headers in a 200 OK (REGISTER)."

On an initial read of this I-D, I thought that the Path extension may
help.  However, the problem appears to be, if we use the Path extension,
that the service route will necessarily be equivalent to the one
formed when a Path-header knowledgable UA gets back a 200 OK (REGISTER)
with multiple Paths.  At least to me, this was not immediately apparent
from the wording in bullet 5. I don't know if my suggested wording
makes it more easier to understand or not...

Also, should the Path I-D be referenced here?

Also, what happens if a 200 OK (REGISTER) has multiple Path headers
*and* a P-Service-Route header?  The I-D does provide some guidance
in Section 5.1: "In general, the service route set is appended to any
locally configured route needed to egress the access proxy chain."
I think this should be highlighted and maybe included in the examples
in Section 5.

That's all.  Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Cc: <sip@ietf.org>, "'3GPP_TSG_CN_WG1'" <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Subject: RE: [Sip] Revised Path draft (-04) submitted
Date: Mon, 29 Apr 2002 14:47:39 -0500
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Vijay wrote:
> 
> Section 1:
>     "UA wishes to register with REGISTRAR. However, due to network
>     topology, UA must use P1 as an "outbound proxy", and all messages
>     between UA1 and REGISTRAR must also traverse P1, P2, and P3 before
>             ^^^
>     reaching REGISTRAR."
> 
> There isn't any UA1, only "UA".  To be consistent with 
> Section 4.5 and 4.6, you may want to re-label "UA" in the 
> figure to "UA1".

Rats. Thought I fixed that one. Thanks.

> In the sentence,
> 
>     "Eventually, REGISTRAR or a service proxy closely related to it
>     will receive a message for UA. It needs to know which proxies
>     must be transited by that message in order to get back to UA..."
> 
> we are talking about *request* initiated by the REGISTRAR 
> destined to UA, right?  If so, can't we reword it as:
> 
>     "Eventually, REGISTRAR or a service proxy closely related to it
>     will need to send a request to the UA.  It needs to know which
>     proxies must be transited by that request in order to get 
> to UA..."

Actually, I'm thinking a third party will send a request to REGISTRAR
(it's a proxy too, right) that is detined for UA. The mindset is a "home
service proxy".

> 
> Generally speaking, in many places where you use "message", I 
> believe you mean "requests".  If so, can we substitute the 
> term request, since a "message" may be a response, for 
> instance, which will follow the Via paths.

Reasonable.

> In Section 4.6, the requests from P1->P2 show P1 adding 
> itself to the Path list (initially set to null).  The branch 
> of P1 does NOT contain the "magic cookie" (z9hG4bK).  
> Likewise for P3->REGISTRAR. Since P1 and P3 supports "lr", it 
> is a good bet that they also support the Via "magic cookie".  
> Of course, this is an obscure nit, but all the same...

Good point.


Excellent feedback. Thanks!

--
Dean


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From daemon@ns.ietf.org  Mon Apr 29 16:28:41 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Revised P-Service-Route header draft, please review
Date: Mon, 29 Apr 2002 15:04:10 -0500
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Vijay wrote:
> Section 1, bullet 5:
>     5. Performing an operation equivalent to record-routing in a
>        REGISTER transaction between the UA and the associated
>        registrar, then storing that route in the UA and reusing it
>        as a service route on future messages originating from the UA.
>        While efficient, this constrains the service route for proxy
>        operations to be congruent with the route taken by the REGISTER
>        message.
> 
> I interpret the last sentence as follows:
>     "While efficient, this constrains the service route for proxy
>      operations to be congruent with the Route set formed at the UA
>      from the Path headers in a 200 OK (REGISTER)."

Yep, that's what I meant. I was just trying not to refer to Path draft .
. .
 
> On an initial read of this I-D, I thought that the Path 
> extension may help.  However, the problem appears to be, if 
> we use the Path extension, that the service route will 
> necessarily be equivalent to the one formed when a 
> Path-header knowledgable UA gets back a 200 OK (REGISTER) 
> with multiple Paths.  At least to me, this was not 
> immediately apparent from the wording in bullet 5. I don't 
> know if my suggested wording makes it more easier to 
> understand or not...



> Also, should the Path I-D be referenced here?

I was hoping not to.

> 
> Also, what happens if a 200 OK (REGISTER) has multiple Path headers
> *and* a P-Service-Route header?  The I-D does provide some 
> guidance in Section 5.1: "In general, the service route set 
> is appended to any locally configured route needed to egress 
> the access proxy chain." I think this should be highlighted 
> and maybe included in the examples in Section 5.

Then I have failed to make something clear. Path and P-Service-Route are
completely independent in specification.

The Path header reports a proxy sequence, associated with a specific
registration, that will be traversed by requests that have visited the
home service proxy associated with the registrar. This is "just so the
UA knows who's in the loop". It is not used, in any way whatsover, to
route messages out of the UA.

The P-Service-Route header provides a proxy sequence that the UA MAY use
in order to request services from a specific proxy cluster.

They MAY have some of the same values in them, just as two different
INVITES may also traverse the same proxy.


How can I make this more clear?


--
Dean


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From daemon@ns.ietf.org  Mon Apr 29 16:58:21 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Dean Willis wrote:

> Vijay wrote:

[...]

> Yep, that's what I meant. I was just trying not to refer to Path draft .

[...]

>>Also, should the Path I-D be referenced here? 
> I was hoping not to.


Any idea why the reluctance?

 
>>Also, what happens if a 200 OK (REGISTER) has multiple Path headers
>>*and* a P-Service-Route header?  The I-D does provide some 
>>guidance in Section 5.1: "In general, the service route set 
>>is appended to any locally configured route needed to egress 
>>the access proxy chain." I think this should be highlighted 
>>and maybe included in the examples in Section 5.
>>
> 
> Then I have failed to make something clear. Path and P-Service-Route are
> completely independent in specification.


Right; on closer introspection, disregard what I wrote above; I am not
debating that Path and P-Service-Route are dependent on each other.
They are not; but since Path and P-Service-Route *appear* to be doing
similar things for requests, i.e. *force* them to follow a certain path,
there is some equivalence that needs to be strongly pointed out.

 > How can I make this more clear?

Maybe something as simple as the following will do:
"The Path mechanism is applicable for requests arriving at a UA from the
home domain, and the P-Service-Route mechanism is applicable for forcing
requests from the UA to visit the home domain"

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@ns.ietf.org  Mon Apr 29 18:53:25 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Revised P-Service-Route header draft, please review
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 . . . 
> 
> Any idea why the reluctance?
> 

Well, it hardly seems right to make a full header dependent on a
P-header, and the full header is more debatable so I did not wish to
make them sequence the other way either. In general, normative
dependencies, especially on works not yet complete, are better when
avoided.

> Right; on closer introspection, disregard what I wrote above; I am not
> debating that Path and P-Service-Route are dependent on each other.
> They are not; but since Path and P-Service-Route *appear* to be doing
> similar things for requests, i.e. *force* them to follow a 
> certain path,
> there is some equivalence that needs to be strongly pointed out.
> 
>  > How can I make this more clear?
> 
> Maybe something as simple as the following will do:
> "The Path mechanism is applicable for requests arriving at a 
> UA from the
> home domain, and the P-Service-Route mechanism is applicable 
> for forcing
> requests from the UA to visit the home domain"

How about:

In Path:

"The routing established by the Path header mechanism applies only to to
requests transiting or originating in the home domain."

In P-Service-Route:

"The routing established by P-Service-Route applies only to requests
originating in the UA and transiting or terminating in the home domain."

--
Dean


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From daemon@ns.ietf.org  Mon Apr 29 18:58:24 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Cc: "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, <brian.rosen@marconi.com>,
        <jo@ipdialog.com>, "'Allison Mankin'" <mankin@isi.edu>
Date: Mon, 29 Apr 2002 17:34:29 -0500
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Subject: [Sip] Request for Expert Review, draft-garcia-sip-associated-uri
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The SIP WG has been asked to provide expert review on the individual
informational track draft-garcisip-associated-URI.

This draft provides a lightweight mechanism proposed by 3GPP by which
registrars may inform registering UAs of additional identities (not
listed in the To: header) that the registrar will assoicate with the
registered contact.

We would like to conclude review of this by March 10. 

The current version of the draft has been sent to internet-drafts and is
also available at:

http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-associated-uri-00
.txt


Please copy comments to the list and to the author,
miguel.a.garcia@ericsson.com.

Thanks,

--
Dean Willis






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Dean Willis wrote:

>>Any idea why the reluctance?
> 
> Well, it hardly seems right to make a full header dependent on a
> P-header, and the full header is more debatable so I did not wish to
> make them sequence the other way either. In general, normative
> dependencies, especially on works not yet complete, are better when
> avoided.


I see.

 
> How about:
> 
> In Path:
> 
> "The routing established by the Path header mechanism applies only to to
> requests transiting or originating in the home domain."
> 
> In P-Service-Route:
> 
> "The routing established by P-Service-Route applies only to requests
> originating in the UA and transiting or terminating in the home domain."

Sounds reasonable.

Thanks, Dean.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@ns.ietf.org  Mon Apr 29 19:16:38 2002
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	Mon, 29 Apr 2002 17:49:24 -0500
From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, <brian.rosen@marconi.com>,
        <jo@ipdialog.com>, "'Allison Mankin'" <mankin@isi.edu>
Date: Mon, 29 Apr 2002 17:49:05 -0500
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Subject: [Sip] Expert Review for draft-garcia-sip-called-party-id
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The SIP WG has been asked to provide expert review on the individual
informational track draft-garcia-sip-called-party-id. 

This is a P-header proposed by 3GPP to provide a lightweight mechanism
for the retention-in-signaling of the public identity targeted by a
request which is retargeted by a serving proxy.

The draft has been sent to internet-drafts and is also available at:

http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-called-party-id-0
0.txt

We would like to complete the review by about April 10.

Please copy comments to the list and to the author
miguel.a.garcia@ericsson.com.

--
Dean


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From daemon@ns.ietf.org  Mon Apr 29 19:19:07 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <brian.rosen@marconi.com>, <jo@ipdialog.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, "'Allison Mankin'" <mankin@isi.edu>,
        "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>
Date: Mon, 29 Apr 2002 17:54:45 -0500
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Subject: [Sip] Expert Review for draft-garcia-sip-visited-network-id
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The SIP Working Group has been requested to provide expert review on the
individual informational draft-garcia-sip-visited-network-id. This
documents a p-header proposed by 3GPP to provide a lightweight mechanism
for identifying the originating domain of a request in a network
composed of adjacent admininistrative domains with pairwise trust and
mutual authentication.

The draft has been posted to internet-drafts and is also available from:

http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-visited-network-i
d-00.txt	

We would like to complete the review process by April 10.

Please copy comments to the list and to the author,
miguel.a.garcia@ericsson.com.

Thanks,

--
Dean Willis


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From daemon@optimus.ietf.org  Tue Apr 30 05:52:07 2002
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From: Tan Ya-Ching  ICM N PG U ID A 1 <Ya-Ching.Tan@icn.siemens.de>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>, sip@ietf.org,
        3GPP_TSG_CN_WG1
	 <3GPP_TSG_CN_WG1@LIST.ETSI.FR>
Cc: bernhard.honeisen@nokia.com, amankin@isi.edu,
        "'Rohan Mahy'"
	 <rohan@cisco.com>, jo@ipdialog.com,
        brian.rosen@marconi.com
Subject: RE: [Sip] Revised Path draft (-04) submitted
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Dean,

Just nits:

- The first line of Path in Table 3 should have a "o" under REG.

- Perhaps you want to add the to the note in 4.7 that for simplication, all
headers added by nodes between UA2 and the REGISTRAR are omitted from the
messages.

Ya-Ching

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From daemon@optimus.ietf.org  Tue Apr 30 07:01:46 2002
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------=_NextPart_000_00CF_01C1F044.65EF0490
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Hi,

Is it possible that a retransmission of a (non invite)request goes over =
tcp, when the original request was sent over udp?

Thanks
Matthias Konrad



------=_NextPart_000_00CF_01C1F044.65EF0490
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Is it possible that a retransmission of =
a (non=20
invite)request goes over tcp, when the original request was sent over=20
udp?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Matthias Konrad</FONT></DIV>
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From daemon@optimus.ietf.org  Tue Apr 30 07:35:48 2002
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Private SIP extension for Associated URI
	Author(s)	: M. Garcia
	Filename	: draft-garcia-sip-associated-uri-00.txt
	Pages		: 4
	Date		: 29-Apr-02
	
This memo describes a private extension to SIP in the form of a
P-Associated-URI header. A UAC registers a URI to a registrar. Then
the registrar informs the UAC of the associated URIs to its registered
URI. This information is conveyed in the P-Associated-URI header,
carried in a 200 OK response to a REGISTER request.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Private SIP extension for Called Party Identity
	Author(s)	: M. Garcia
	Filename	: draft-garcia-sip-called-party-id-00.txt
	Pages		: 6
	Date		: 29-Apr-02
	
This memo describes a private extension to SIP in the form of a
P-Called-Party-ID header. The home serving proxy inserts this header
typically in an INVITE, en-route to its destination. The header is
populated with the URI received by the proxy in the request. The UAS
identifies which ID out of several IDs the invitation was sent to (for
example, the user may be using simultaneously a personal and a
business SIP URI to receive invitation to sessions). The UAS may use
the information to render different distinctive audiovisual alerting
tones, depending on the ID used to receive the invitation to the
session.

A URL for this Internet-Draft is:
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	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-garcia-sip-called-party-id-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20020429151306.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-garcia-sip-called-party-id-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-garcia-sip-called-party-id-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20020429151306.I-D@ietf.org>

--OtherAccess--

--NextPart--



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 07:38:36 2002
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Message-Id: <200204301115.HAA12944@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
CC: sip@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 30 Apr 2002 07:15:05 -0400
Subject: [Sip] I-D ACTION:draft-sparks-sip-mimetypes-03.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Internet Media Types message/sipfrag
	Author(s)	: R. Sparks
	Filename	: draft-sparks-sip-mimetypes-03.txt
	Pages		: 5
	Date		: 29-Apr-02
	
This document registers the message/sipfrag MIME media type.  This
type is similar to message/sip , but allows fragments of well formed
SIP messages to be used for the same tunelling purposes as message/
sip.  In addition to end-to-end security uses , message/sipfrag is
used with the REFER method  to tunnel information about the status of
a referrenced request.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sparks-sip-mimetypes-03.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
	"get draft-sparks-sip-mimetypes-03.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
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NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20020429151228.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-sparks-sip-mimetypes-03.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-sparks-sip-mimetypes-03.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20020429151228.I-D@ietf.org>

--OtherAccess--

--NextPart--



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 07:44:02 2002
Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged))
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Received: (from daemon@localhost)
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Message-Id: <200204301115.HAA13052@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
CC: sip@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 30 Apr 2002 07:15:30 -0400
Subject: [Sip] I-D ACTION:draft-garcia-sip-visited-network-id-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Private SIP extension for Visited Network Identifier
	Author(s)	: M. Garcia
	Filename	: draft-garcia-sip-visited-network-id-00.txt
	Pages		: 5
	Date		: 29-Apr-02
	
This memo describes a private extension to SIP in the form of a
P-Visited-Network-ID header. The contents of the header identify each
of the visited networks the message traversed en-route to the home
network.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-garcia-sip-visited-network-id-00.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
	"get draft-garcia-sip-visited-network-id-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-garcia-sip-visited-network-id-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20020429151316.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-garcia-sip-visited-network-id-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-garcia-sip-visited-network-id-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20020429151316.I-D@ietf.org>

--OtherAccess--

--NextPart--



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 07:54:59 2002
Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged))
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Message-Id: <200204301115.HAA13102@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
CC: sip@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 30 Apr 2002 07:15:39 -0400
Subject: [Sip] I-D ACTION:draft-beckmann-sip-reg-event-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Registration Event package
	Author(s)	: G. Mayer, M. Beckmann
	Filename	: draft-beckmann-sip-reg-event-00.txt
	Pages		: 9
	Date		: 29-Apr-02
	
This draft defines an event package that allows a network entity to
request to be notified of changes in the registration state of a
particular user.  Subscription and notification of registration state
is supported by defining an event package within the general  SIP
event notification event framework.  This event package is based on
the Presence event package as specified in [3] and therefore this
draft only describes the deltas to the Presence event package.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-beckmann-sip-reg-event-00.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
	"get draft-beckmann-sip-reg-event-00.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-beckmann-sip-reg-event-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20020429151336.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-beckmann-sip-reg-event-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-beckmann-sip-reg-event-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20020429151336.I-D@ietf.org>

--OtherAccess--

--NextPart--



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 08:10:58 2002
Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged))
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Received: (from daemon@localhost)
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Date: 30 Apr 2002 04:25:00 -0000
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Received: from unknown (152.1.162.22) by rediffmail.com via HTTP; 30 Apr 2002 04:25:00 -0000
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From: "ruhiya  " <ruhiya@rediffmail.com>
Reply-To: "ruhiya  " <ruhiya@rediffmail.com>
To: sip@ietf.org
Content-type: text/plain;
	format=flowed
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Subject: [Sip] SIP protocol stack
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Hi,

SIP is a application layer protocol, I would like to know the 
complete protocol stack for SIP.

Actually if I want to design a router which can suppoert SIP what 
would be the minimun set of protocols that the router should 
support and at which layer?

Any help or idea would be appreciated.

Thanks,
Ruhiya
_________________________________________________________
Click below to visit monsterindia.com and review jobs in India or 
Abroad
http://monsterindia.rediff.com/jobs



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 09:25:16 2002
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Message-Id: <200204301115.HAA13118@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
CC: sip@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 30 Apr 2002 07:15:43 -0400
Subject: [Sip] I-D ACTION:draft-willis-sip-scvrtdisco-02.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension Header for Service Route Discovery in 
                          Private Networks
	Author(s)	: D. Willis, B. Hoeneisen
	Filename	: draft-willis-sip-scvrtdisco-02.txt
	Pages		: 15
	Date		: 29-Apr-02
	
This document proposes a private SIP extension header used in
conjunction with responses to REGISTER messages to provide a
mechanism by which a registrar may inform a registering UA of a
service route that the UA may use to request outbound services from
the registrar's domain.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-willis-sip-scvrtdisco-02.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-willis-sip-scvrtdisco-02.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-willis-sip-scvrtdisco-02.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20020429151345.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-willis-sip-scvrtdisco-02.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-willis-sip-scvrtdisco-02.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20020429151345.I-D@ietf.org>

--OtherAccess--

--NextPart--



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 09:37:06 2002
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: The SIP Replaces Header
	Author(s)	: B. Biggs, R. Dean, R. Mahy
	Filename	: draft-ietf-sip-replaces-01.txt
	Pages		: 25
	Date		: 29-Apr-02
	
This document proposes a new header for use with the SIP call control
architecture.  The Replaces header is used in peer-to-peer call
control to logically replace an existing SIP dialog with a new SIP
dialog.  This primitive can be used to enable a variety of features,
for example: 'Attended Transfer' and 'Retrieve from Call Park'.  Note
that definition of these example features is non-normative.

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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 10:37:20 2002
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ruhiya,
this is not the correct forum to ask questions about how sip is to 
be implemented. there is a separate list for that on the sip home 
page.
sip is an application layer protocol. as long as your router is 
able to handle regulare tcp/ip communications, there is no need to 
do anything in particular. although we would love it if your 
router secretly gave priority to traffic that involves ports 5004, 
5060, 5005. but i am just being silly.
if you need more information on implementational aspects please 
post it to the implementor's list or you can email me.
- farhan
_________________________________________________________
Click below to visit monsterindia.com and review jobs in India or 
Abroad
http://monsterindia.rediff.com/jobs


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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 11:10:23 2002
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Date: Tue, 30 Apr 2002 09:40:05 -0500
From: Suryaram Alladi <Suryaram.Alladi@alcatel.com>
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I have a question related to SDP information in SIP-T. Is the following
SDP format allowed in SIP-T? If not is there an alternate method to
achieve the same? The application is negotiation of SDP by media
gateways supporting Megaco  and SIP-T is used between two MGCs to
transport the SDP information.


v=0
c=IN IPV ipddress
m=media virtualConnId RTP/AVP 0
v=0
c=ATM NSAP atm_address
 m=media virtualConnId AAL1/ATMF 0

Thanks
Suri



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 11:15:09 2002
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Date: Tue, 30 Apr 2002 16:30:14 +0200
To: sip@ietf.org
From: Jiri Kuthan <jiri@iptel.org>
Cc: Jan Janak <J.Janak@sh.cvut.cz>
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Subject: [Sip] REGISTER Processing
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There seems to be insufficient guidance in bis-09 on usage of Call-ID
in REGISTERs. The only thing which I found there is in 10.2.4:

  "A UA SHOULD use the same Call-ID for all registrations during a
   single boot cycle. "

I think that does NOT require Call-ID to change across reboots. UAs
with no memory are not explicitely mandated to change Call-ID (ignoring
that bis asks to deny REGISTERs with Call-ID== and CSeq <= values in registrar's
bindings). If the UACs actually do not change Call-ID, REGISTERs after reboots 
with low CSeq will be ignored.

Which actually happened during two tests in SIPIt.

Also the SHOULD strengh seems to weak -- 'SHOULD' MUST be used if
not doing so does not break things, which is not the case here.

--

Other confusing thing is which SIP reply a request failure for out-of-order REGISTERs 
in line 1740 implies. I could connect it to the next paragraph "....if binding 
update fails...the request MUST fail with 500..." but I am not sure that 500 is 
a good choice.

--

Yet anothet thing, which Jan pointed out, is that handling of out-or-order
in lines 1738-40 also includes retransmissions (CSeq and CallID in request and
bindings equal). Sending some negative reply upstream would in that case 
cause the upstream UAC to consider registration failed.

-Jiri


--
Jiri Kuthan            http://iptel.org/~jiri/



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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 11:39:01 2002
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Date: Tue, 30 Apr 2002 11:12:55 -0400
Subject: [Sip] I-D ACTION:draft-willis-sip-path-04.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension for Registering Non-Adjacent Contacts
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-path-04.txt
	Pages		: 15
	Date		: 29-Apr-02
	
The REGISTER function is used in a SIP system primarily to associate
a temporary contact address with an address-of-record.  This contact
is generally in the form of a URI, such as Contact:
<sip:alice@pc33.atlanta.com> and is generally dynamic and associated
with the IP address or hostname of the SIP UA.  The problem is that
network topology may be that there are one or more SIP proxies
between the UA and the registrar, such that any message from the
user's home network to the registered UA must traverse these proxies.
The REGISTER method does not give us a mechanism to discover and
record this sequence of proxies in the registrar for future use.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-willis-sip-path-04.txt

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From daemon@optimus.ietf.org  Tue Apr 30 13:45:16 2002
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From: "Tom-PT Taylor"<taylor@nortelnetworks.com>
To: "'Suryaram Alladi'" <Suryaram.Alladi@alcatel.com>, sip@ietf.org
Subject: RE: [Sip] SDP information in SIP body
Date: Tue, 30 Apr 2002 13:24:47 -0400
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SIP only allows one session description.  (SIP-T is SIP with interworking
rules and encapsulation.)  Thus what you would have to put in your SDP is

v=0
m=media port RTP/AVP 0
c=IN IPV ipddress
m=media virtualConnId AAL1/ATMF 0
c=ATM NSAP atm_address

In other words, the c= lines now become media-level rather than
session-level in scope, and the two transports are presented as simultaneous
streams from which either the far end chooses one (setting
port/virtualConnId to 0 to reject the other) or supports both.  If the
latter happens, the originator has to send an UPDATE to indicate which
alternative it prefers, again by setting port/virtualConnId to 0 to reject
the other.


-----Original Message-----
From: Suryaram Alladi [mailto:Suryaram.Alladi@alcatel.com]
Sent: Tuesday, April 30, 2002 10:40 AM
To: sip@ietf.org
Subject: [Sip] SDP information in SIP body


I have a question related to SDP information in SIP-T. Is the following
SDP format allowed in SIP-T? If not is there an alternate method to
achieve the same? The application is negotiation of SDP by media
gateways supporting Megaco  and SIP-T is used between two MGCs to
transport the SDP information.


v=0
c=IN IPV ipddress
m=media virtualConnId RTP/AVP 0
v=0
c=ATM NSAP atm_address
 m=media virtualConnId AAL1/ATMF 0

Thanks
Suri



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 14:11:10 2002
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Extension for Registering Non-Adjacent Contacts
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-path-04.txt
	Pages		: 15
	Date		: 29-Apr-02
	
The REGISTER function is used in a SIP system primarily to associate
a temporary contact address with an address-of-record.  This contact
is generally in the form of a URI, such as Contact:
<sip:alice@pc33.atlanta.com> and is generally dynamic and associated
with the IP address or hostname of the SIP UA.  The problem is that
network topology may be that there are one or more SIP proxies
between the UA and the registrar, such that any message from the
user's home network to the registered UA must traverse these proxies.
The REGISTER method does not give us a mechanism to discover and
record this sequence of proxies in the registrar for future use.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-willis-sip-path-04.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 14:45:56 2002
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Date: Tue, 30 Apr 2002 11:17:56 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: sip@ietf.org
cc: john.loughney@nokia.com
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Subject: [Sip] new AAA requirements draft -- discussion on the SIPPING list
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Hi,

FYI: I just posted this announcement about AAA requirements to SIPPING.
Comments on the draft should get discussed on the SIPPING list.

thanks,
-rohan

--------------

Hi,

John Loughney has just submitted a first draft rewrite of the SIP AAA
requirements.  Until it appears in the archive, you can download it from:

http://www-nrc.nokia.com/sua/draft-loughney-sip-aaa-req-00.txt

Comments are solicited on the SIPPING list only.

John will also be leading discussion on this topic at the interim meeting
next week.



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 15:32:13 2002
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	Tue, 30 Apr 2002 14:06:52 -0500
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, <brian.rosen@marconi.com>,
        <jo@ipdialog.com>, "'Allison Mankin'" <mankin@isi.edu>
Subject: RE: [Sip] Expert Review for draft-garcia-sip-called-party-id
Date: Tue, 30 Apr 2002 14:06:32 -0500
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That's a due date of May 10, 2002, not April 10.

Thanks,

--
Dean

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Monday, April 29, 2002 5:49 PM
> To: sip@ietf.org
> Cc: 'Miguel A. Garcia'; 'Rohan Mahy'; 
> brian.rosen@marconi.com; jo@ipdialog.com; 'Allison Mankin'
> Subject: [Sip] Expert Review for draft-garcia-sip-called-party-id
> 
> 
> 
> The SIP WG has been asked to provide expert review on the 
> individual informational track draft-garcia-sip-called-party-id. 
> 
> This is a P-header proposed by 3GPP to provide a lightweight 
> mechanism for the retention-in-signaling of the public 
> identity targeted by a request which is retargeted by a serving proxy.
> 
> The draft has been sent to internet-drafts and is also available at:
> 
http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-called-party-id-0
0.txt

We would like to complete the review by about April 10.

Please copy comments to the list and to the author
miguel.a.garcia@ericsson.com.

--
Dean


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 15:32:58 2002
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	Tue, 30 Apr 2002 14:05:18 -0500
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>, <sip@ietf.org>
Cc: "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, <brian.rosen@marconi.com>,
        <jo@ipdialog.com>, "'Allison Mankin'" <mankin@isi.edu>
Subject: RE: [Sip] Request for Expert Review, draft-garcia-sip-associated-uri
Date: Tue, 30 Apr 2002 14:04:57 -0500
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May 10, 2002. Not March 10. I have no excuse.

--
Dean

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Monday, April 29, 2002 5:34 PM
> To: sip@ietf.org
> Cc: 'Miguel A. Garcia'; 'Rohan Mahy'; 
> brian.rosen@marconi.com; jo@ipdialog.com; 'Allison Mankin'
> Subject: [Sip] Request for Expert Review, 
> draft-garcia-sip-associated-uri
> 
> 
> 
> The SIP WG has been asked to provide expert review on the 
> individual informational track draft-garcisip-associated-URI.
> 
> This draft provides a lightweight mechanism proposed by 3GPP 
> by which registrars may inform registering UAs of additional 
> identities (not listed in the To: header) that the registrar 
> will assoicate with the registered contact.
> 
> We would like to conclude review of this by March 10. 
> 
> The current version of the draft has been sent to 
> internet-drafts and is also available at:
> 
http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-associated-uri-00
.txt


Please copy comments to the list and to the author,
miguel.a.garcia@ericsson.com.

Thanks,

--
Dean Willis






_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 15:36:02 2002
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	Tue, 30 Apr 2002 14:04:34 -0500
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: <brian.rosen@marconi.com>, <jo@ipdialog.com>,
        "'Rohan Mahy'" <rohan@cisco.com>, "'Allison Mankin'" <mankin@isi.edu>,
        "'Miguel A. Garcia'" <Miguel.A.Garcia@ericsson.com>
Subject: RE: [Sip] Expert Review for draft-garcia-sip-visited-network-id
Date: Tue, 30 Apr 2002 14:04:14 -0500
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Ok, make that due date May 10, not April 10. That's 2002 . . .

I hate it when I make a wrong turn in the time machine and end up in the
middle of last month.

--
dean

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Monday, April 29, 2002 5:55 PM
> To: sip@ietf.org
> Cc: brian.rosen@marconi.com; jo@ipdialog.com; 'Rohan Mahy'; 
> 'Allison Mankin'; 'Miguel A. Garcia'
> Subject: [Sip] Expert Review for draft-garcia-sip-visited-network-id
> 
> 
> 
> The SIP Working Group has been requested to provide expert 
> review on the individual informational 
> draft-garcia-sip-visited-network-id. This documents a 
> p-header proposed by 3GPP to provide a lightweight mechanism 
> for identifying the originating domain of a request in a 
> network composed of adjacent admininistrative domains with 
> pairwise trust and mutual authentication.
> 
> The draft has been posted to internet-drafts and is also 
> available from:
> 
http://www.softarmor.com/sipwg/drafts/draft-garcia-sip-visited-network-i
d-00.txt	

We would like to complete the review process by April 10.

Please copy comments to the list and to the author,
miguel.a.garcia@ericsson.com.

Thanks,

--
Dean Willis


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use
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From daemon@optimus.ietf.org  Tue Apr 30 15:44:49 2002
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Date: Tue, 30 Apr 2002 12:25:12 -0700 (Pacific Daylight Time)
From: Rohan Mahy <rohan@cisco.com>
To: sip@ietf.org, <sipping@ietf.org>
cc: rohan@cisco.com
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Subject: [Sip] Updated agenda for SIP/SIPPING Interim Meeting
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Hi,

Below is updated agenda information for the interim:  (Note that I
rearranged several of the slots within each day.)

SIP/SIPPING Interim Meeting
May 6-7, 2002
Room S225  (2nd Floor of South Hall)
Las Vegas Convention Center
3150 Paradise Road at Convention Center Drive
Las Vegas, Nevada, USA

The meeting space and network connectivity is being sponsored by
pulver.com's SIP Summit (pulver is part of Key3Media who also runs
Interop).  If you get lost, just ask where SIP summit is.
You do not need to register, but if you have not already done so, please
let me know if you are coming so I can get a reasonable estimate of
attendance.  If you want a shirt, please bring some cash.

Also please note that unfortunately the wireless network won't be up until
Monday afternoon.

thanks,
-rohan

Rohan Mahy
SIPPING co-chair

------------

Monday, May 6

08:30  Welcome, Logistics, Agenda Bashing -- Rohan Mahy

09:00  Multiparty Apps Framework and Conferencing Requirements --
Dan Petrie, Orit Levin
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-framework-00.txt
http://www.softarmor.com/sipping/drafts/draft-levin-sipping-conferencing-requirements-00.txt

10:30  REFER / Replaces / cc-transfer / service examples -- Robert Sparks,
Rohan Mahy, Alan Johnston
http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-02.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-refer-split-00.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-referredby-split-00.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-sec-options-00.txt
http://www.ietf.org/internet-drafts/draft-sparks-sip-mimetypes-03.txt
http://www.ietf.org/internet-drafts/draft-ietf-sip-replaces-01.txt
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-cc-transfer-05.txt
http://www.softarmor.com/sipping/drafts/draft-ietf-sipping-service-examples-01.txt

13:00  Lunch

14:00  The call-info and conference-info packages -- Jonathan Rosenberg
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-call-package-01.txt

15:30  Content Indirection
http://www.ietf.org/internet-drafts/draft-olson-sipping-content-indirect-00.txt

16:30  Binding Published data to SIP -- Ben Campbell
very long mailing list thread beginning with:
http://www1.ietf.org/mail-archive/working-groups/sipping/current/msg01544.html

17:15  Key Events/ Stimulus Markup -- Jonathan Rosenberg
http://www.ietf.org/internet-drafts/draft-culpepper-sipping-app-interact-reqs-00.txt
http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-markup-00.txt

18:00  Wrap

Tuesday, May 7

08:30  AAA requirements (Requirements) -- John Loughney and Gonzalo
Camarillo
http://www.ietf.org/internet-drafts/draft-calhoun-sip-aaa-reqs-04.txt
http://www.softarmor.com/sipping/drafts/draft-loughney-sip-aaa-req-00.txt

10:30  URI as service indicator -- Ben Campbell
RFC 3087

11:15  Request History (Requirements) -- Mary Barnes and Mark Watson
http://www.ietf.org/internet-drafts/draft-watson-sipping-request-history-01.txt

12:00  Lunch

13:00  Privacy and Network Asserted Identity (Direction Forward) -- Cullen
Jennings
http://www.softarmor.com/sipping/drafts/draft-watson-sipping-nai-reqs-00.txt
http://www.ietf.org/internet-drafts/draft-ietf-sip-privacy-04.txt
http://www.ietf.org/internet-drafts/draft-peterson-sip-longterm-privacy-00.txt
http://www.ietf.org/internet-drafts/draft-peterson-sip-identity-00.txt

16:30  Future Security Needs (Discussion)
http://www.ietf.org/internet-drafts/draft-ietf-sip-sec-agree-00.txt
http://www.ietf.org/internet-drafts/draft-undery-sip-auth-00.txt
http://www.softarmor.com/sipwg/drafts/draft-thomas-sip-sec-req-00.txt

18:00 Wrap



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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 17:27:09 2002
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From: "chen Zhang" <c_zhang88@hotmail.com>
To: sip@ietf.org
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Subject: [Sip] SIP and MPLS
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Hi,
SIP is an application layer protocol and origins as a large scale multiparty 
conferencing protocol.
MPLS is an advanced forwarding scheme.

Is it possible to join the SIP and MPLS? If it is, we can get
following benefits:
1. using SIP to set up the LSP.
2. providing the higher quality in the Internet.

IF you have any comments about that, Please let me know.

Thank you for your attention

Best regards

Chen



_________________________________________________________________
Send and receive Hotmail on your mobile device: http://mobile.msn.com



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 18:10:37 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'chen Zhang'" <c_zhang88@hotmail.com>, <sip@ietf.org>
Subject: RE: [Sip] SIP and MPLS
Date: Tue, 30 Apr 2002 16:50:37 -0500
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I believe Jon Crowcroft and Mark Gibson suggested something like this at
IETF 46 and barely escaped afterwards. So it's definitely a possibility.
However, it didn't appear at the time that there was much interest in
pursuing it at the IETF.

The slides from their presentation should be at:

http://www.softarmor.com/sipwg/meets/ietf46/slides/pres-gibson-sip-qos-r
esv-sip-46.PDF

And the related draft is probably still in the draft "morgue" on that
site.

--
Dean Willis


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of chen Zhang
> Sent: Tuesday, April 30, 2002 11:05 AM
> To: sip@ietf.org
> Subject: [Sip] SIP and MPLS
> 
> 
> Hi,
> SIP is an application layer protocol and origins as a large 
> scale multiparty 
> conferencing protocol.
> MPLS is an advanced forwarding scheme.
> 
> Is it possible to join the SIP and MPLS? If it is, we can get 
> following benefits: 1. using SIP to set up the LSP. 2. 
> providing the higher quality in the Internet.
> 
> IF you have any comments about that, Please let me know.
> 
> Thank you for your attention
> 
> Best regards
> 
> Chen
> 
> 
> 
> _________________________________________________________________
> Send and receive Hotmail on your mobile device: http://mobile.msn.com
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 18:29:44 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'chen Zhang'" <c_zhang88@hotmail.com>, sip@ietf.org
Subject: RE: [Sip] SIP and MPLS
Date: Tue, 30 Apr 2002 18:11:40 -0400
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At the moment, the IETF is treating MPLS as a half-baked cat let out of
the bag, and is trying to put it back in the bag, without success.
Asking to do something involving SIP and MPLS in the same breath is
likely to bring the "layer violation" police out in force.

However:

IF you could bring MPLS to an endpoint
AND you wanted to make the media go directly over an MPLS lsp
OR you used a media relay which, say, took in IP/UDP/RTP and
	pushed that into an LSP

THEN
You could consider putting together an SDP extension along the lines
of the ATM extension, and then you could specify how to create
an lsp to handle the media flow.  No changes to SIP would be required.
I doubt there is any benefit to doing anything with the signalling.

The label stacking mechanism of MPLS could conceivably be put to good
use in aggregating media flows that needed similar QoS handling.

While I think this is an interesting line of inquiry, I suspect:
	a) MPLS networks, today, don't offer any advantage vis a vis
		SIP/multimedia streams that IP networks don't already have
	b) You aren't going to see MPLS to the desktop for a LONG time

I do still wonder if a site-edge relay that mapped streams to lsps,
wouldn't be an interesting architecture.

Brian
	

> -----Original Message-----
> From: chen Zhang [mailto:c_zhang88@hotmail.com]
> Sent: Tuesday, April 30, 2002 12:05 PM
> To: sip@ietf.org
> Subject: [Sip] SIP and MPLS
> 
> 
> Hi,
> SIP is an application layer protocol and origins as a large 
> scale multiparty 
> conferencing protocol.
> MPLS is an advanced forwarding scheme.
> 
> Is it possible to join the SIP and MPLS? If it is, we can get
> following benefits:
> 1. using SIP to set up the LSP.
> 2. providing the higher quality in the Internet.
> 
> IF you have any comments about that, Please let me know.
> 
> Thank you for your attention
> 
> Best regards
> 
> Chen
> 
> 
> 
> _________________________________________________________________
> Send and receive Hotmail on your mobile device: http://mobile.msn.com
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Apr 30 18:29:56 2002
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From: "Hussam Jarada" <hussam.jarada@ulticomdal.com>
To: <sip@ietf.org>
Date: Tue, 30 Apr 2002 17:07:27 -0500
Message-ID: <004901c1f093$6aa1c550$ee9e82cc@ulticomdal.com>
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Subject: [Sip] tel URI in SIP spec bis-09
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Hi,

In section 7.1, line 787 states that:
[SIP elements MAY support Request-URIs with schemes other than "sip" and
"sips", for example the
"tel" URI scheme of RFC 2806]

But in section 8.1.12,  line 997 state that:
[All SIP implementations MUST support the SIP and URI scheme]

I think URI is just an error typing and it should be SIPS in line 997, cause
you have MAY in section 7.1 but MUST in section 8.1.12.

Thanks.

Hussam Jarada
Ulticom Inc.




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From daemon@optimus.ietf.org  Tue Apr 30 19:28:10 2002
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Date: Tue, 30 Apr 2002 18:00:01 -0500
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] SIP and MPLS
In-reply-to: <F191irbThA9QRmzeyNt00003d26@hotmail.com>
To: "'chen Zhang'" <c_zhang88@hotmail.com>, sip@ietf.org
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> Is it possible to join the SIP and MPLS? If it is, we can get 
> following benefits: 1. using SIP to set up the LSP. 2. 
> providing the higher quality in the Internet.

I believe this is a horrible idea, and as mentioned by Dean has been
discussed and firmly rejected before. The reason it is so detestful
IMHO, is the Internet goes over many 'nets' withouth making any
assumption about any specific layer 2 end-to-end. That what "Inter" is
all about. It's IP end-to-end, not anything else.
(Some may suspect that MPLS here is just a mutant of ATM...)

MPLS may have have certainly its good use, but this is not one of them
IMHO.

The assumption:
>> providing the higher quality in the Internet
...is also wrong with regard to the Internet. Just make a SIP phone call
using two SIP devices end-to-end and you will see QoS is just fine as
is, especially with the present bandwidth glut. Access QoS is a
different story, but MPLS does not apply there either.

There are the DiffSev and IntServ Internet standards and MPLS is one of
the L2 that can be used with them, but RTP packets will see just IP.

Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA
 

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of chen Zhang
> Sent: Tuesday, April 30, 2002 11:05 AM
> To: sip@ietf.org
> Subject: [Sip] SIP and MPLS
> 
> 
> Hi,
> SIP is an application layer protocol and origins as a large 
> scale multiparty 
> conferencing protocol.
> MPLS is an advanced forwarding scheme.
> 
> Is it possible to join the SIP and MPLS? If it is, we can get 
> following benefits: 1. using SIP to set up the LSP. 2. 
> providing the higher quality in the Internet.
> 
> IF you have any comments about that, Please let me know.
> 
> Thank you for your attention
> 
> Best regards
> 
> Chen
> 
> 
> 
> _________________________________________________________________
> Send and receive Hotmail on your mobile device: http://mobile.msn.com
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 


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From daemon@optimus.ietf.org  Tue Apr 30 20:08:36 2002
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Why does it need to?

-Jonathan R.

Bob Penfield wrote:
> 
> One more comment.
> 
> This may have already been covered by other comments, but I could not
> find a
> requirement that the UAC MUST receive the 200 OK to the PRACK for the
> 155
> response before sending an UPDATE.
> 
> cheers,
> (-:bob
> 
> Robert F. Penfield
> Chief Software Architect
> Acme Packet, Inc.
> 130 New Boston Street
> Woburn, MA 01801
> bpenfield@acmepacket.com

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



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From daemon@optimus.ietf.org  Tue Apr 30 20:10:58 2002
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Jo,

Thanks for your comments. Responses inline:

Jo Hornsby wrote:
> 
> Okay, so here's my comments.  Apologies for any dups.
> 
>  * Section 1; Paragraph 2;
>    Sentence beginning "While this is reasonable":
>    s/where it this/where this/

Fixed.


> 
>  * Section 3:
>    Would it perhaps be possible to also include
>    493?  Is there a case where a UAC could
>    automatically Do The Right Thing in this case?

Thats the question. I think you'd generally need to query the user,
although its not unreasonable for a user to configure their UA to
automatically retry in this case. 

> 
>  * Section 4; Paragraph 2:
>    I'm not quite sure what the differentiation
>    is between 2543bis and 100rel, but shouldn't
>    the reference here also be referencing the
>    RFCxxxx?

100rel is about reliability of provisional responses, which is not in
bis at all (anymore)... I think the reference is right as it stands.


> 
>  * Section 4; Paragraph 4:
>    I am unsure as to the motivation for this?
>    Wouldn't it just be better to always do 155
>    because, certainly at least in the first
>    instance, a large number of proxies are
>    going to be UPDATE-ignorant?

Hmm, thats a good point. I think this paragraph can stay as it is, but
instead, section 5.2.1 should say that, if supported, the uas SHOULD
request an update instead of generating the repairable error response. 


> 
>  * Section 4; Paragraph 5:
>    I'm not quite sure what the differentation
>    is between 2543 and sip-events, but shouldn't
>    the reference here also be referencing the
>    RFCyyyy?

[5] is sip-events, I don't get it?

> 
>  * Section 5.1; Paragraph 2:
>    I think the "MAY" worries me.  Is the
>    expectation that a large number of UAs are
>    going to support this quite quickly?

Only the market can decide that.

>     Else
>    this looks like something that could break
>    interoperability with a sledgehammer.
>    Perhaps there should be a note about how
>    this is slightly dangerous, unless the
>    UA has advanced knowledge of the network/
>    peer UA?

This really isn't different than any other extension. I'll note the
issue though.
> 
>  * Section 5.2.1; Paragraph 1:
>    See, here I think if we upgraded the "MAY"
>    to a "SHOULD", we could probably knock out
>    the bit about a proxy flicking the Require:
>    update switch on.  But, I'm worried I'm
>    missing something blindingly obvious.  &:(

No, that seems to be the right thing. Changed.

> 
>  * Section 5.2.2; Paragraph 2:
>    I was going to suggest using the Reason
>    header here (header inspection I found a
>    little too icky), but then I referred back
>    to Reason:, and now I get the feeling that
>    this change is already in the pipeline. &:)

Now that the draft has been split into two (UPDATE and 155), Reason is
part of the 155 stuff. It wasn't there initially for process reasons,
which are now moot.

> 
>  * Section 5.3; Paragraphs 2/3(motivational):
>    Minor point, but 100rel makes the creation
>    of an "early dialog" a MUST.  Obviously,
>    this does no harm, but perhaps the
>    redundancy will cause confusion?

THe 1xx need not have been sent reliably; 100rel requires it, but
baseline bis doesn't (although the words are kind of vague in the spec). 

> 
>  * Section 5.3; Paragraph 7:
>    s/proxy/UAC/

Fixed.

> 
>  * Section 5.3; Paragraph 7:
>    Ah.  Here's the placeholder for Reason:.
>    Now I'm betting this is a process issue --
>    there's no way that Reason is going to
>    be ready in time, right?

See above.

> 
>  * Section 6.1:
>    I'm a little confused by the placement of
>    this text, especially given the previous
>    paragraph, which says much the same thing?

Its just transition text; I'll clean up.

> 
>  * Section 6.1.1:
>    I'm a little confused by the "In the case
>    of INVITE".  Huh?  Is it like in the
>    case of a _session_ created by an INVITE?

UPDATE has the semantics of the request which created the dialog on
which its sent. However, I must admit that this makes me kind of
queasy... This bit about INVITE vs. SUBSCRIBE is only an issue for the
herfp usage, now in a separate draft. We have the opportunity now to
consider whether that is the actual semantic we want.

> 
>  * Section 6.1.1; Paragraph 3:
>    [BTW, vertical whitespace constitutes a
>    paragraph to my warped mind.  I apologize
>    if this is just silly.]
>    Isn't this actually a contradiction with
>    100rel, since that says that an answer
>    must appear in the first reliable
>    provisional response if there was an offer
>    in the INVITE?  I realize that the 155
>    is just a facade, but it probably needs
>    to be called out that this is in fact okay.

This is an odd exception, due to the fact that 155 isn't *really* the
response. Really, the response is an error response, and the SDP is the
capabilities of the UAS, as would be present in a 488.

> 
>  * Section 6.1.1; Paragraph 8;
>    Sentence beginning "If the UPDATE":
>    Should it be explicitly pointed out that
>    the "reliable provisional response" must
>    have beed PRACKed; and possibly responded
>    to, if the PRACK contained an offer?
>    In fact, there is some horrible potential
>    for out-of-order with 2xx-to-PRACK and
>    UPDATE here.  Hmmm....  This may be a
>    tad radical, but couldn't we just rule
>    out sending offers in PRACKs when a
>    dialog is UPDATE-capable?  The only
>    downside I can see is extra
>    request/response pairs when a UAC wants
>    to update session parameters at the exact
>    same time a UAS returns a provisional
>    response reliably.  This doesn't seem
>    that likely to me beyond the first
>    provisional response in an early dialog;
>    but I have to admit that I'm manyfolks-
>    ignorant, so it might be of utmost
>    importance there.

We had a whole thread on this. The conclusion was that the UAS needs to
be prohibited from sending an UPDATE with an offer if it has sent SDP in
a 1xx for which it hasn't gotten a PRACK yet. We can't disallow offers
in PRACK, since it is too late for that. The reason for it was pressure,
as I recall, from manyfolks, to allow for the baseline manyfolks
exchange without an extra transaction (remember the 3gpp guys want to
keep the transaction count down). 


> 
>  * Section 6.1.1; Last Paragraph:
>    s/Section 4.1/Section 14.1/

Fixed.

> 
>  * Section 6.2; Paragraph 2:
>    Might be nice to reference Section 12.2 of
>    RFC BBBB here.  Just a thought -- it'll
>    doubtless save my sanity later.

OK. Anything to save your sanity.

> 
>  * Section 6.2; Paragraph 4;
>    Sentence beginning "Once the application":
>    s/response to request/response to the request/

Fixed.

> 
>  * Section 6.2.1; Paragraph 2;
>    Sentence beginning "The original request":
>    I'd be inclined to append "for example".

OK.

> 
>  * Section 6.2.1; Paragraph 2:
>    Is this paving the way for retrying an
>    UPDATE?  (For example: challenge to INVITE
>    in 155; UPDATE with credentials; now the
>    UAS realizes that it doesn't grok the media,
>    so 488 in 155.)  This probably needs to
>    either be explicitly allowed or otherwise.

No, this paragraph is avoiding infinite 155 loops. You are talking about
a different proble, which I don't think is discussed, and needs to be. I
suppose it is OK to generate another 155 in this case. 

> 
>  * Section 7:
>    Personally, I see this as the wrong approach
>    for maintaining backward compatibility.
>    Although the work that the proxy has to do
>    is somewhat trivial, it seems far more likely
>    that a UA is going to support this extension
>    than a proxy.  This means that it's not
>    entirely unlikely that UAs will just not
>    use UPDATE because a proxy is forking but
>    not doing the Require.  Also, how likely is
>    it that a UA that isn't UPDATE-aware is
>    going to be fly enough to accept resubmitted
>    requests that are virtual merges?
>    Well, I've said my bit. &:)

Per other threads, this has been changed to a MAY.

> 
>  * Table 1/2; "proxy" column:
>    [I'm comparing the same table in bis-09]
>     o Alert-Info/Call-Info:
>       Is there any reason this is "am" here as
>       opposed to "ar" in bis-09?

No. Fixed.

>     o Contact: 3xx:
>       Should be "d"?

Yup.

>     o Content-Length:
>       Should be "ar".

yup.

>     o Error-Info:
>       Should be "a"?

yup.

>     o Organization:
>       Should be "ar".

Yup.

>     o Priority: through Record-Route::
>       Need aligning with Table 3 of bis-09.

yup.

>     o Require:
>       Not sure what "c" means here.

Fixed.

>     o Route:
>       Should be "adr".

yup.

>     o Via:/WWW-Authenticate::
>       Require alignment with Table 3 of bis-09.

Fixed.

> 
>  * Table 1/2; "UPDATE" column:
>     o Allow: 405:
>       Should be "m"?

yup.

>     o Contact: R/2xx:
>       Should be "m"?

Hmm. Its not strictly needed, but since its a target refresh request, it
probably has to be there....


>     o Min-Expires: 423:
>       Should be "m"?

For the subscribe usage, yes, but since update and herfp are split, the
baseline update won't need it, so its a -.

>     o Unsupported: 420:
>       Should be "m"?

yup.

>    Same issues with aligning with bis-09 Tables
>    in terms of Proxy-Authenticate/
>    Record-Route/Via/WWW-Authenticate; specifically,
>    the "where" column.

Right. Sadly, I copied this table from an older version of bis, before
we have a major overhaul, and it got out of sync. I re-aligned.

> 
>  * Table 2:
>    "copied with possible addition of tag" I don't
>    think makes sense here, since there's always
>    going to be a tag, right?

yup.

> 
>  * Section 12.3:
>    CopyPaste error?  (Should be talking about the
>    155, right?)

Right.

> 
> Finally, is any normative behaviour required for
> the case when an UPDATE is received after, say, the
> initial INVITE has completed unsuccessfully?
> An obvious case where this could happen, that I
> can think is where a CANCEL has been issued, and
> theres vague racing going on.  I'm guessing you
> just 487, but perhaps this should be spelled out?
> (Especially if you don't 487¬ (:&).

Yes, you 487. No different than any other mid-dialog request outside of
an active 
dialog.

Thanks for your comments.

-Jonathan R.

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



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From daemon@optimus.ietf.org  Tue Apr 30 20:46:32 2002
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Folks,

I just submitted an update to UPDATE, based on comments from the list.
Until it appaers in the archives, you can find it at:

http://www.jdrosen.net/papers/draft-ietf-sip-update-02.txt

The changes:

* excised 155, herfp, reason stuff
* table 1/2 aligned with bis
* fixed text to indicate that you can't send an UPDATE until you've
received the PRACK for a 1xx you sent that contained SDP.
* more rigorous processing defined

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

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