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Subject: [sipcore] I-D Action: draft-ietf-sipcore-location-conveyance-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : Location Conveyance for the Session Initiation Protocol
	Author(s)       : James Polk
                          Brian Rosen
                          Jon Peterson
	Filename        : draft-ietf-sipcore-location-conveyance-09.txt
	Pages           : 32
	Date            : 2011-09-04

   This document defines an extension to the Session Initiation
   Protocol (SIP) to convey geographic location information from one
   SIP entity to another SIP entity.  The SIP extension covers
   end-to-end conveyance as well as location-based routing, where SIP
   intermediaries make routing decisions based upon the location of the
   Location Target.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-location-conveyance-=
09.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-location-conveyance-0=
9.txt

From ivo.sedlacek@ericsson.com  Thu Sep  8 03:13:37 2011
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From: Ivo Sedlacek <ivo.sedlacek@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Sep 2011 12:15:25 +0200
Thread-Topic: RFC5626, reg-id usage after power up
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Subject: [sipcore] RFC5626, reg-id usage after power up
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Hello,

can anyone please help me with the following RFC5626 related question?

Suppose a UA registers with registrar over two flows. A Contact with featur=
e tag X is registered over one flow and a Contact with feature tag Y is reg=
istered over other flow.
Whenever the UA registers the Contact with the feature tag X, the reg-id=3D=
987 is used.
Whenever the UA registers the Contact with the feature tag Y, the reg-id=3D=
654 is used.

When the UA is powered up for the first time, the UA uses reg-id=3D987 when=
 registering over the first flow and uses reg-id=3D654 when registering ove=
r the second flow later on.
Then the UA is switched off.
When the UA is powered up again:
A)    must the UA use reg-id=3D987 when registering over the first flow and=
 reg-id=3D654 when registering over the second flow later on?
or
B)    can  the UA use reg-id=3D654 when registering over the first flow and=
 reg-id=3D987 when registering over the second flow later on?

RFC5626, section 4.2.1 talks about "the same sequence of reg-id values", bu=
t it is a little unclear.

----
4.2.1.  Initial Registrations
...
   A UAC conforming to this specification MUST include in the Contact
   header field, a "reg-id" parameter that is distinct from other
   "reg-id" parameters used in other registrations that use the same
   "+sip.instance" Contact header field parameter and AOR.  Each one of
   these registrations will form a new flow from the UA to the proxy.
>>>
   The sequence of reg-id values does not have to be sequential but MUST
   be exactly the same sequence of reg-id values each time the UA
   instance power cycles or reboots, so that the reg-id values will
   collide with the previously used reg-id values.
<<<
This is so the
   registrar can replace the older registrations.
...
----

Thanks for clarification.

Kind regards

Ivo Sedlacek




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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=3Dus-ascii">
<META content=3D"MSHTML 6.00.6002.18457" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2>Hello,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>can anyone please help me with the followi=
ng=20
RFC5626 related question?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Suppose a UA registers with registrar over=
 two=20
flows. A Contact with feature tag X is registered over one flow and a Conta=
ct=20
with feature tag Y is registered over other flow.<BR>Whenever the UA regist=
ers=20
the Contact with the feature tag X, the reg-id=3D987 is used.<BR>Whenever t=
he UA=20
registers the Contact with the feature tag Y, the reg-id=3D654 is=20
used.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT size=3D2>When the UA is powered up for the fi=
rst time,=20
the UA uses reg-id=3D987 when registering over the first flow and uses reg-=
id=3D654=20
when registering over the second flow later on.<BR>Then the UA is switched=
=20
off.<BR>When the UA is powered up again:<BR>A)&nbsp;&nbsp;&nbsp; must the U=
A use=20
reg-id=3D987 when registering over the first flow and reg-id=3D654 when reg=
istering=20
over the second flow later on?<BR>or<BR>B)&nbsp;&nbsp;&nbsp; can&nbsp; the =
UA=20
use reg-id=3D654 when registering over the first flow and reg-id=3D987 when=
=20
registering over the second flow later on?</FONT></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>RFC5626, section 4.2.1 talks about "the sa=
me=20
sequence of reg-id values", but it is a little unclear.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>----<BR>4.2.1.&nbsp; Initial=20
Registrations<BR>...<BR>&nbsp;&nbsp; A UAC conforming to this specification=
 MUST=20
include in the Contact<BR>&nbsp;&nbsp; header field, a "reg-id" parameter t=
hat=20
is distinct from other<BR>&nbsp;&nbsp; "reg-id" parameters used in other=20
registrations that use the same<BR>&nbsp;&nbsp; "+sip.instance" Contact hea=
der=20
field parameter and AOR.&nbsp; Each one of<BR>&nbsp;&nbsp; these registrati=
ons=20
will form a new flow from the UA to the proxy.<BR>&gt;&gt;&gt;<BR>&nbsp;&nb=
sp;=20
The sequence of reg-id values does not have to be sequential but=20
MUST<BR>&nbsp;&nbsp; be exactly the same sequence of reg-id values each tim=
e the=20
UA<BR>&nbsp;&nbsp; instance power cycles or reboots, so that the reg-id val=
ues=20
will<BR>&nbsp;&nbsp; collide with the previously used reg-id=20
values.<BR>&lt;&lt;&lt;<BR>This is so the<BR>&nbsp;&nbsp; registrar can rep=
lace=20
the older registrations.<BR>...<BR>----</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks for clarification.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Kind regards</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Ivo Sedlacek</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><BR></FONT>&nbsp;</DIV></BODY></HTML>

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From dworley@avaya.com  Thu Sep  8 08:18:25 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Ivo Sedlacek <ivo.sedlacek@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 8 Sep 2011 11:20:15 -0400
Thread-Topic: RFC5626, reg-id usage after power up
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Subject: Re: [sipcore] RFC5626, reg-id usage after power up
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> From: Ivo Sedlacek [ivo.sedlacek@ericsson.com]
>=20
> When the UA is powered up again:
> A)    must the UA use reg-id=3D987 when registering over the first flow
> and reg-id=3D654 when registering over the second flow later on?
> or
> B)    can  the UA use reg-id=3D654 when registering over the first flow
> and reg-id=3D987 when registering over the second flow later on?
>=20
> RFC5626, section 4.2.1 talks about "the same sequence of reg-id
> values", but it is a little unclear.

The important aspect is:

   so that the reg-id values will collide with the previously used
   reg-id values.  This is so the registrar can replace the older
   registrations.

Strictly, the registrar/proxy doesn't depend on the UA repeating
values of reg-id in order to operate correctly.  But the system
operates much better if, when a later registration is intended to be
the successor of an earlier registration, it uses the same reg-id.

In your case, it appears that the feature tags are important, so you
would want to use reg-id 987 when registering with feature tag X, and
reg-id 654 when registering with feature tag Y.

Dale

From internet-drafts@ietf.org  Fri Sep  9 04:41:57 2011
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-reqs-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : Requirements for indication of features supported by a S=
IP proxy
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
	Filename        : draft-ietf-sipcore-proxy-feature-reqs-01.txt
	Pages           : 8
	Date            : 2011-09-09

   The Session Initiation Protocol (SIP) &quot;Caller Preferences&quot; ext=
ension
   defined in RFC 3840 provides a mechanism that allows a SIP message to
   convey information relating to the originator&#39;s supported features/
   capabilities.  This document defines requirements for a mechanism
   that would allow SIP proxies to convey information relating to the
   proxy&#39;s supported features/capabilities.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-proxy-feature-reqs-0=
1.txt

Internet-Drafts are also available by anonymous FTP at:
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This Internet-Draft can be retrieved at:
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.txt

From christer.holmberg@ericsson.com  Fri Sep  9 04:44:02 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Fri, 9 Sep 2011 13:45:53 +0200
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Hi,

I've submitted a new version (-01) of the proxy-feature reqs draft.

Based on feedback, I've added a couple of requirements, and done some edito=
rial modifications in the requirements, and some editorial clarifications i=
n the use-cases.

Regards,

Christer


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<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">I've submitted a new version (-01) of the proxy-featu=
re reqs draft.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Based on feedback, I've added a couple of requirement=
s, and done some editorial modifications in the requirements, and some edit=
orial clarifications in the use-cases.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
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From iesg-secretary@ietf.org  Mon Sep 12 08:02:59 2011
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Subject: [sipcore] Protocol Action: 'Location Conveyance for the Session Initiation	Protocol' to Proposed Standard	(draft-ietf-sipcore-location-conveyance-09.txt)
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The IESG has approved the following document:
- 'Location Conveyance for the Session Initiation Protocol'
  (draft-ietf-sipcore-location-conveyance-09.txt) as a Proposed Standard

This document is the product of the Session Initiation Protocol Core
Working Group.

The IESG contact persons are Robert Sparks and Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-sipcore-location-conveyance/




      Technical Summary
        This document defines an extension to the Session Initiation
        Protocol (SIP) to convey geographic location information
        from one SIP entity to another SIP entity.  The SIP extension
        covers end-to-end conveyance as well as location-based
        routing, where SIP intermediaries make routing decisions
        based upon the location of the Location Target.

      Working Group Summary
        This work began in the (now concluded) SIPPING working group
        back in 2003, and progressed through the (also concluded)
        SIP working group before being finalized in the SIPCORE
        working group. The protocol mechanism underwent significant
        simplification in early 2010 to reflect a better understanding
        of the core requirements underpinning the work. Although
        this simplification was initially controversial, the working
        group did manage to achieve a rather strong consensus around
        the new mechanism after some additional changes.

      Document Quality
        This protocol mechanism is described in the 3GPP document
        24.229 as a component of certain emergency calling scenarios
        in IMS-based networks. It was developed in the SIP and
        SIPCORE working groups with input from GEOPRIV working group
        participants.

RFC Editor Note:

(updated to apply to -09)

1) In Section 4.4
OLD

  There is no guidance given in this document as to which
  locationValue, when more than one was present in the SIP request,
  is related to the Geolocation-Error code; meaning that, somehow
  not defined here, the LR just picks one to error.

NEW

  When more than one locationValue is present in a SIP request,
  this mechanism provides no indication which one the Geolocation-
  Error code corresponds to. If multiple errors are present, the
  LR applies local policy to select one.


2) Section 4.6, fourth paragraph, first sentence:
OLD
       location information via an HTTP
NEW
      location information via HTTP

3) Section 8.3, registration of geolocation-http:
OLD
    via an HTTP
NEW
   via HTTP

4) Please change the current inclusion of TLP 3.0 section 6.b to use the TLP 4.0 section 6.b.

From brett@broadsoft.com  Wed Sep 14 05:15:41 2011
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From: Brett Tate <brett@broadsoft.com>
To: "adam@nostrum.com" <adam@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 14 Sep 2011 05:17:33 -0700
Thread-Topic: draft-ietf-sipcore-rfc3265bis-03: comments concerning RFC 5057
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Subject: [sipcore] draft-ietf-sipcore-rfc3265bis-03: comments concerning RFC 5057
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Concerning the following draft-ietf-sipcore-rfc3265bis-03 section 4.1.2.2 s=
nippet indicating subscriber "MUST consider the subscription terminated", t=
he draft should potentially better indicate that the MUST does not preempt =
the RFC 5057 notes (such as for 482 and 483) which indicate situations wher=
e the dialog would not be terminated.

Section 4.1.2.2: "If a SUBSCRIBE request to refresh a subscription receives=
 a 404, 405, 410, 416, 480-485, 489, 501, or 604 response, the subscriber M=
UST consider the subscription terminated.  (See [RFC5057] for further detai=
ls and notes about the effect of error codes on dialogs and usages within d=
ialog, such as subscriptions)."

Do the RFC 5057 notes for 482 edge condition still apply?  It indicates an =
RFC 3263 advancing situation which might trigger a 482 response and allow t=
he subscription to remain.  If the RFC 5057 notes for 482 edge condition st=
ill apply, is it adequately indicated within the above snippet?

Thanks,
Brett


From ibc@aliax.net  Fri Sep 16 07:37:13 2011
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Subject: [sipcore] About draft-gurbani-sip-sipsec
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Hi all,

I would like to comment about draft-gurbani-sip-sipsec "The SIPSEC
Uniform Resource Identifier (URI)".
I like the idea proposed there. Even more taking into account that it
is based on the same mechanism for HTTP/CONNECT (widely deployed).



One of the drawback of this mechanism is that it mandates creating
multiple TCP connections between two proxies, this is:

                             P1               P2
Alice  --CONNECT---->------------------> ---------------------> Bob
Carol  --CONNECT---->------------------> ---------------------> Adam
Tomy  --CONNECT---->------------------> ---------------------> Mark

In a normal SIP scenario, if communication between P1 and P2 was TCP
or TLS, P1 would just mantain a single connection with P2. But using
CONNECT as stated in the draft, P1 needs to open 3 TCP connections
with P2 (as neither P1 or P2 can inspect the data transported).

I don't think this "limitation" is a showstopper, but just something
to take into account.



Another "issue": The draft requires that the TLS negotiation is made
by both endpoints (let's say two phones), so the UAS MUST have a TLS
certificate, which is uncommon, am I right?


And the last comment: proxies in the path cannot inspect the SIP
traffic through the established connection so they know nothing about
what's happening in the call or transactions. Proxies cannot do
accounting or whatever other action as they are not SIP proxies now,
but just TCP proxies. How does that fit with current SIP deployments?


Thanks a lot.





--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From vkg@bell-labs.com  Fri Sep 16 09:04:03 2011
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Date: Fri, 16 Sep 2011 11:08:43 -0500
From: "Vijay K. Gurbani" <vkg@bell-labs.com>
Organization: Bell Laboratories, Alcatel-Lucent
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Subject: Re: [sipcore] About draft-gurbani-sip-sipsec
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On 09/16/2011 09:39 AM, Iñaki Baz Castillo wrote:
> Hi all,
>
> I would like to comment about draft-gurbani-sip-sipsec "The SIPSEC
> Uniform Resource Identifier (URI)".
> I like the idea proposed there. Even more taking into account that it
> is based on the same mechanism for HTTP/CONNECT (widely deployed).

Thanks!

> Another "issue": The draft requires that the TLS negotiation is made
> by both endpoints (let's say two phones), so the UAS MUST have a TLS
> certificate, which is uncommon, am I right?

Uncommon, yes.  However, self-signed certificates would work (as long
as the fingerprint was exchanged by some other means).

> And the last comment: proxies in the path cannot inspect the SIP
> traffic through the established connection so they know nothing about
> what's happening in the call or transactions. Proxies cannot do
> accounting or whatever other action as they are not SIP proxies now,
> but just TCP proxies. How does that fit with current SIP deployments?

It does not.  Understandably, the rub against this work back then,
and I presume now as well, will be that the value proposition of
turning all the intermediaries into simple byte forwarders is
hard to justify.

However, if you are Mick O'Malley [1], then you'd be happy if
such a scheme was provided :-)

[1] "Counting from Zero", Alan Johnston, 2011.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
Web:   http://ect.bell-labs.com/who/vkg/

From ibc@aliax.net  Fri Sep 16 09:09:26 2011
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2011/9/16 Vijay K. Gurbani <vkg@bell-labs.com>:
> It does not. =C2=A0Understandably, the rub against this work back then,
> and I presume now as well, will be that the value proposition of
> turning all the intermediaries into simple byte forwarders is
> hard to justify.
>
> However, if you are Mick O'Malley [1], then you'd be happy if
> such a scheme was provided :-)

:)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From dean.willis@softarmor.com  Tue Sep 27 17:34:28 2011
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Date: Tue, 27 Sep 2011 19:37:11 -0500
From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [sipcore] About draft-gurbani-sip-sipsec
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On 9/16/11 9:39 AM, Iñaki Baz Castillo wrote:
> Hi all,
>
> I would like to comment about draft-gurbani-sip-sipsec "The SIPSEC
> Uniform Resource Identifier (URI)".
> I like the idea proposed there. Even more taking into account that it
> is based on the same mechanism for HTTP/CONNECT (widely deployed).
>

Thanks!

>
> Another "issue": The draft requires that the TLS negotiation is made
> by both endpoints (let's say two phones), so the UAS MUST have a TLS
> certificate, which is uncommon, am I right?

I understand that comodohacker has made certs widely available. You 
wanna be gmail for a day?

Seriously, yes, you're right. And this is a major "design feature", not 
unintended. It gives us real "end to end" authentication and privacy, 
subject to the strength of the certificate issuance process (see: 
comodohacker).

As Vijay pointed out, you can issue your own certificates if you and 
your partner work out the trust relationships outside of the usual CA 
process. The certificate used can also be dynamically varied with the 
target identity.

This works nicely in the enterprise space, for example.

Let's say as a consultant working for many clients, each gives me a cert 
to use when talking with them. My UA can key off the remote domain 
identifier in order to know which cert to present during handshake.

However, nothing about the protocol really demands mutual cert-auth. It 
should be possible to use HTTP-style cert-auth of the UAS and digest 
auth of the UAC.

On the other hand, you're going to need a cert to answer the phone, so 
you might as well use it when making calls, too.


> And the last comment: proxies in the path cannot inspect the SIP
> traffic through the established connection so they know nothing about
> what's happening in the call or transactions. Proxies cannot do
> accounting or whatever other action as they are not SIP proxies now,
> but just TCP proxies. How does that fit with current SIP deployments?

That's another feature, not a bug. Current SIP deployments are mostly 
MITM attacks waiting to happen.

Where this becomes incredibly important is P2PSIP (aka SIP/RELOAD). When 
some other bozo's box is acting as your proxy, do you really want it 
doing "accounting or whatever other action" on your SIP messages?

Further, the processing load on the proxies is hugely decreased. Not 
only can they stop worrying about wiretapping every SIP message (as such 
would be pointless), they no longer require the overhead of having to 
decrypt each message before logging it and then re-encrypting it before 
resending it. This makes them hundreds of times more scalable.

It's a win-win for everybody who matters.

Of course, this also works well for criminal conspiracies that have a 
good IT department and can mint and share their own certs out-of-band. 
I'm OK with that. Note that the presence of a communication between the 
two endpoints is still revealed; it's just the text of the signaling 
between them that is protected by the protocol, so snoops could still 
get a "pen register", which in the US is typically a warrantless request 
anyhow. True anonymity is much harder; what is needed is a TOR-ified 
version of the protocol, and that still reveals two half-calls.

--
Dean




--
Dean


--
Dean
