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VIPR Design Team conference call on Friday September 30th, from 9:05am to 9:55am
PST.

o Attendees:
  Mary Barnes
  Hakim Mehmood
  Muthu Arul Mozhi Perumal
  Jon Peterson
  Marc Petit-Huguenin
  John Ward
  Dean Willis

o Regrets:
  Daryl Malas
  Michael Procter

o Agenda

- - Security section of -overview (5 min)
- - SIP Intermediaries (10 min)
- - Capabilities (10 min)
- - RELOAD registration (5 min)
- - PVP privacy (10 min)
- - PVP method priority and registration (10 min)
- - PVP method entropy (10 min)
- - Replacement ticket by client certificate (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (5 min)

o Action items

- - Mary will send an improved text for the -overview Security Considerations
section to the mailing-list.
- - Marc will send an email describing the problem and use cases for SIP
intermediaries.

o Minutes

- - Security section of -overview

Marc: Mary published a new version of the -overview document.  Security section
still need some work.

Mary: My impression is that it's 80% there.  Not sure what it's in there will
get us past AD security review.

Hakim: Any concern on the test of phone numbers?

Mary: The idea of the attacker claiming phone numbers as its own was part of the
concerns that Richard Shockey brought forward.

Dean: There is potential for denial of service by claiming a lot of numbers.

Hakim: The idea with VIPR is that you bring your own resources.  There should be
a way for the overlay to reject excessive requests.

Mary: Does RELOAD handle that?

Dean: The fundamental problem with RELOAD is that someone can dedicate 10000's
CPU to mess up the overlay.  But that's a RELOAD problem, not a VIPR problem

Hakim: DHTs perform better as more nodes are added.

Dean: Only if those nodes are performing correctly.

Hakim: It's an attack on performance.

Jon: In the security section we must be careful to enumerate the limitations of
the system. One thing that is missing is the property of the token that is
carried in SIP. What are the protocols that are securing the actual network
interaction, e.g. TLS. Should list the counter-measures, for example for the
phone number mechanism it would be a combination of the overlay validation
mechanism and the generation of the SIP token.

Marc: The methods that are described in -pvp are just examples, and the security
of these should be deferred to the document where they are defined.

Mary: It's OK to point solution in other documents but we need to be sure that
we have a comprehensive security overview.

Jon: Is there an authentication problem, is there an integrity problem?  Then
saying that there is a token in this other draft.

Marc: So who want to contribute?

Mary: I can make a pass at the modifications suggested and send it to the
mailing-list.

- - SIP Intermediaries

Marc: We agreed on the previous call that SIP intermediaries should not be only
for toll bypass, but to improve the SIP situation.  I put a proposal together
and this proposal is to carry both the SDP to the PSTN gateway (aka SIPconnect
profile) and the SDP with SIP enhanced feature (aka SIPbeyond, which could be
defined in collaboration with RTCWEB) in a multipart/alternative body.

Jon: I am not initially a huge fan of this.  We want to keep VIPR as close as
possible to standard SIP.  As said in the previous call, the high level solution
is to be clear that VIPR is intended to provide feature transparency.  If you
just provide a great PSTN service, then you should not be doing this.

Marc: I agree 100%, but what VIPR domains who want to participate should do?

Hakim: I agree with Jon, the original intent of VIPR was to use SIP has is. I am
not sure there is a good solution. One of the approach is to collect all the
routes you can and then use a policy to choose one, a policy could be to choose
the route that have the best features, or your policy could be to choose the
least cost.

Marc: It's a different topic.  It is not about choosing the best route, but for
intermediate VIPR domains on an outgoing call.  We all agree that a domain that
is not improving SIP should not participate.  The question is what do we do when
an intermediate want to do the right thing.

Jon: Same answer that I gave last time, if we do that, we completely missed what
we want to do with VIPR.  Any entity that is advertizing a route to an endpoint
should doing so only if it is a route to the features of the endpoint.

Dean: It's almost a denial of service on the intent of VIPR.

Jon: We want to have a story to how VIPR is gonna work with federations, but I
see that as a distant goal for us.  I am not saying that intermediaries should
not participate, but we need to understand how to work with this kind of systems.

Marc: We all agree on this, but we are not proposing a solution.  I think that
we should not exclude intermediaries, and I disagree that we should do this
later.  If we exclude them then we will not have enough participants.

[The next 15 minutes of audio recording are lost.  Next call will have one more
recording point to prevent this to happen again.  Sorry for that]

Jon: ...we should not give up on the idea that we can have one route per phone
number provisioned in your VIPR entity.

John: We should definitively talk more about this.  For us to know as a
provider, a carrier, the capabilities of a given endpoint is something that will
be more and more possible.  So having a route to an endpoint at this capability
level is something that we really want support.

Jon:  If it is desirable for an endpoint to defer to an upstream entity, it's
perhaps possible to have a kind of negotiation.  There is other ways than
forcing this complexity.

John: Your issue is that it is the endpoint that should decide based on all the
entities that reported back.

Jon: I am trying to deal with the issue which is how do you deal with sending a
SIP INVITE that there is multiple places it could land and how do you build the
richest connection features.  How does an INVITE will reach the entity with the
richest feature, which is not a trivial problem to solve.

John: Unless each of these routes are reporting their supported feature set.

Hakim: But intermediaries will not really know.

John: I am not convinced that's a valid assumption. Suppose that a provider
knows the features supported by the devices.  Why shouldn’t this provider been
able to participate in VIPR?

Hakim: OK, but what about an entity one step further?

John: They shouldn't be advertizing it.

Hakim: Then how do you prevent it?

John: If you do not allow the proxy, then you can never, ever, have non-VIPR
players play with VIPR federations.

Hakim: I am not suggesting that. Everybody should publish, but we need to have a
way to pick the right route.

Marc:  Sorry to interrupt, but we are running out of time.  I will send another
email only describing the problem, and use cases. But we need discussion on the
mailing-list to accelerate decisions.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Mon Oct  3 09:23:28 2011
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Hi Muthu,

On 09/28/2011 11:10 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
> A fundamental question: Why would a provider want to advertise routes for a
> number for which neither of the enterprises is willing to pay for?

I am not sure why you are saying that.  When an enterprise uses a provider, it
is paying for the service of routing the calls to the PSTN, so why advertizing
the number in the VIPR overlay could not be part of this cost?

> Isn't ViPR based on the fact the provider is trusted for the first (PSTN)
> call? If the provider is fooling with the enterprise, won't the enterprise
> switch to a different provider? Or are we concerned about the security
> implications around routing subsequent calls through the provider? Aren't
> they applicable to the first call as well?
>
> If the enterprise doesn't trust the provider then ViPR in its current form
> doesn't seem to make sense.
>
> Muthu

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Mon Oct  3 10:24:49 2011
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Subject: [VIPR] A new proposal for SIP Intermediaries
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For the second time, the Design Team conference call got stuck on the subject of
SIP intermediaries in VIPR, and we were not able to make any progress on the
VIPR specifications themselves.  I was supposed to send another email to this
mailing-list to describe in more details the problems created by SIP
intermediaries but I fear that the next conference calls will still being spent
arguing about this.  So I would like to propose an alternate route to solve this
issue.

SIP intermediaries are not described at all in the current specifications, but I
think that everybody agree that we should say something about it, because it
will become a reality, even if we decide to make it a SHOULD NOT (which would
exclude a lot of potential participants from the VIPR federation).  On the other
hand, VIPR domains does not require SIP intermediaries to work.

So my proposal is:

1. Put everything related to SIP intermediaries in a separate I-D, so it does
not hold the main specifications.
2. Add in the main specifications the "hooks" required to be able to add support
for SIP intermediaries as an extension in the future (as opposed to have to work
on a new version of the VIPR RFCs to add SIP intermediaries, which will push
back the deployment of SIP intermediaries for another 3 years).
3. Add a new milestone for the SIP intermediaries (I would suggest April 2012[1])

Please let me know as soon as possible if you agree with this new plan, at least
on the two first points, so I can release the first version of the new
draft-petithuguenin-vipr-sip-intermediaries I-D, and so we can move one on the
other subjects that are actually blocking the work on -framework.

Thanks.


[1] And I would also suggest to move at the same time the "Submit Specification
of authorization tokens to mitigate SPAM for publication as Proposed Standard"
to December 2011, as it make sense to submit this at the same time than the
other core VIPR specifications.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Tue Oct  4 10:03:31 2011
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As discussed in the last conference call, I prepared a description of the
problem created by SIP intermediaries in a VIPR federation.  But as explained
yesterday, I decided to publish it as an Internet-Draft instead of in an email,
so we can separate this problem from the more pressing problems that we need to
solve to be able to publish a new version of the VIPR documents in time for the
IETF meeting in Taipei.

Thanks.

- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-vipr-sip-intermediaries-00.txt
Date: Tue, 04 Oct 2011 09:59:49 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

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

	Title           : Verification Involving PSTN Reachability (VIPR) enabled SIP
Intermediaries
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-petithuguenin-vipr-sip-intermediaries-00.txt
	Pages           : 8
	Date            : 2011-10-04

   This document presents some of the problems created by SIP
   intermediaries inside a VIPR federation.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-vipr-sip-intermediaries-00.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-petithuguenin-vipr-sip-intermediaries-00.txt


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From petithug@acm.org  Tue Oct  4 11:55:15 2011
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From: Marc Petit-Huguenin <petithug@acm.org>
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A new version that adds more details on the problems created by SIP intermediaries.

- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-vipr-sip-intermediaries-01.txt
Date: Tue, 04 Oct 2011 11:54:01 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

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

	Title           : Verification Involving PSTN Reachability (VIPR) enabled SIP
Intermediaries
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-petithuguenin-vipr-sip-intermediaries-01.txt
	Pages           : 11
	Date            : 2011-10-04

   This document presents some of the problems created by SIP
   intermediaries inside a VIPR federation.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-vipr-sip-intermediaries-01.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-petithuguenin-vipr-sip-intermediaries-01.txt

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A. Agenda

- - Move SIP intermediaries in a future draft (5 min)
- - RELOAD registration (5 min)
- - Replacement ticket by client certificate (10 min)
- - PVP privacy (10 min)
- - PVP method priority and registration (10 min)
- - PVP method entropy (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (5 min)

Note that the sum of the time allocated to each topic exceeds the time allocated
to the whole call.  The topics that cannot make it to the discussion will be
postponed to the next conference call if they are not subsequently decided in
the mailing-list.

Because the conference bridge as a limited capacity, an individual invitation to
the conference will be sent shortly after this email to each member of the
design team.  Please send me an email if you are not on this list and want to
receive an invitation.

Note that the design team conference call is not a way to short-circuit the
normal process, but a way to accelerate it by increasing face to face time.  All
decisions made must be confirmed on the mailing-list.

This conference call is subject to the rules of RFC 5378 and RFC 3979 (updated
by RFC 4879).


B. Handout (reading time: 10 min)

B.1. Move SIP intermediaries in a future draft

SIP intermediaries are not described at all in the current specifications, but
everybody agree that we should say something about it, because it will become a
reality, even if we decide to make it a SHOULD NOT (which would exclude a lot of
potential participants from the VIPR federation).  On the other hand, VIPR
domains does not require SIP intermediaries to work.

So the proposal is:

1. Put everything related to SIP intermediaries in a separate I-D, so it does
not hold the main specifications.
2. Add in the main specifications the "hooks" required to be able to add support
for SIP intermediaries as an extension in the future (as opposed to have to work
on a new version of the VIPR RFCs to add SIP intermediaries, which will push
back the deployment of SIP intermediaries for another 3 years).
3. Add a new milestone for the SIP intermediaries (April 2012 proposed)

B.2. RELOAD registration

The current specification states that a phone number must be registered 3 times,
which with the underlying replication done by the CHORD RELOAD algorithm, means
that the mapping between a PVP server and a phone number is stored 9 times.  The
question discussed in Quebec City was what to do when the 3 copies retrieved
from the overlay are not identical, and the proposal was to use the copy that
appears at least twice, so no peer can alone create a denial of service for a
specific number.  The discussion moved to the reasons for having 3 copies in the
first place.

The proposal is to drop the requirement for the 3 copies from VIPR, but to add
the requirement that the node requesting the phone number MUST also fetch (or
stat) the 2 additional replicas and use the value that matches at least one
other copy (using a majority vote prevents a rogue peer to block the access to
the phone numbers currently stored here).  If not all 3 copies are equals, then
an incident must be logged.  The PVP client must retrieve the 2 replicas in
addition to the original registration and use the voting algorithm to choose one
and log an incident if all 3 are not equal.

B.3. Ticket vs Client certificate

In the current specification, the PVP server builds and signs a VIPR ticket, and
sends it back to the PVP client, which passes it to the SIP element.  The SIP
element will insert this ticket in the next SIP method to the remote SIP
element, which will check the validity of the ticket and the domain of the TLS
client.

There is multiple problems with this process:

- - The verification of the ticket use HMAC-SHA1, and so the same password has to
be provisioned in both the SIP element (to verify the ticket) and the PVP server
(to generate the ticket).
- - The ticket can be used without TLS, and we all know that implementers think of
MUST as SHOULD and SHOULD as MAY.

So the proposal would be to replace the ticket by a client certificate, which
could work like this:

Instead of using a symmetrical key, the local PVP server generates an RSA key
pair, with the public key distributed to the SIP element(s).  After a successful
PVP connection, the remote PVP client sends a PKCS#10 certificate request,
containing the domain name and signed with a private key.  The local PVP server
generates a new certificate, sign it with its private key and send it back to
the remote PVP client, which pass it back to the remote SIP element.

The remote SIP element will use the certificate received to establish the TLS
connection for the SIP transactions to the local SIP element, which will use a
standard CA based certificate for its own domain.  The local SIP element will be
able to verify with the PVP server public key that it issued the certificate,
then will validate the standard parts of the certificate (expiration date and so
on).

There is multiple advantages to this proposal:

- - The private key is stored only in the PVP server, or better in a physical token.
- - VIPR does not work without TLS for the SIP connection.
- - There is plenty of commercial or free TLS and X.509 implementations that have
years of experience and testing, which is not the case for the ticket proposal.
- - Security agility is provided by TLS.
- - No additional SIP Header.
- - Because the ticket verification process is folded into the client certificate
verification, less processing resources are needed.

B.4. PVP Privacy

As reported by Michael Procter, there is ways with the current PVP methods to
discover valuable information about a competitor.  But first let's add some
common vocabulary related to PVP:

- - The PVP selector is a list of parameters that are sent by the PVP client side
to retrieve a set of call records in the PVP server side.  The resulting set can
be empty, in which case the validation fails, can contain one element, or can
contain multiple elements in which case the method description must also defines
what call record should be selected (e.g. in the case of method "a", the most
recent call record is selected).  One important point is that the selector is
always passed from the PVP client to the PVP server in clear, and so this is
where the privacy leakage happens.

- - The PVP parameter is a list of name/value pairs that are passed in clear
between the PVP client and PVP server to aid the verification.  Examples are the
rounding value and the vservice id for methods "a" and "b".

- - The PVP secret is the information that each side of the PVP transaction tries
to verify.  The algorithm used (SRP) uses a zero-knowledge password proof, so
neither side can deduce the secret if the verification fails.  In the case of
methods "a" and "b" the secret is the rounded start and end time for the call,
but there is a lot of other possible secrets that can be used in new methods
(DTMF, UUIE, fingerprint, SMS hash, etc...).

In the privacy issue, the problem is that callee number and caller numbers are
disclosed in the PVP selector for method "a" and the callee number is disclosed
in the PVP selector for method "b".

One proposed solution would be to hash these numbers, but finding the phone
numbers would still be trivial for a determined competitor.  Passing the hash
salt as a PVP parameter and using an adaptive hash method like bcrypt will force
the PVP server to rehash all the current terminating call record, but will force
a competitor to rehash probably half of the total E.164 space to find the
numbers.  This is still possible but can provide an adequate privacy protection
level.

Another possibility would be to invert the selector and the secret.  After all
the star/stop time of a call is not as sensitive as the caller/callee phone
numbers.  In this method the PVP selector would be the start/stop time and the
secret would be the caller/callee numbers.  The problem is that it will be more
difficult to design an algorithm to select a unique call record if the returned
set contains more than one.  One solution could be to add more data in the PVP
selector, for example by adding the call initiation time and the direction of
the termination.

We do not even need to decide - we can define these two algorithms as different
methods (in addition to "a" and "b") and let the VIPR domain choose (see PVP
registry proposal).

B.5. PVP Methods Registry

On the PVP subject, The VIPR specification currently defines two methods for the
PSTN validation:  Method "a", which is used when the Caller-ID is available and
method "b", which is used when it is not.  The PVP draft also permits to define
new validation methods, but it does not explain when and how to use this
extended methods.

This proposal is in two parts.  First we need to establish a IANA registry for
the PVP methods, and to assign a priority to each of these methods.  Let's say
for now that the priority is between -∞ (lowest priority) to +∞ (highest
priority) and that priorities should be assigned by steps of 100.  With this
scale we could assign priority 0 to method "b" and priority 100 to method "a".
Then we say in the spec that a VIPR server must try available methods starting
from the highest priority to the lowest priority, until one succeeds or all
fail.  This implements the same algorithm that is currently in -pvp ("a" first,
then "b").

Now the problem with this is that the number of methods will increase in the
future, but we may not want that the PVP client tries all the methods available
each time.  So the second part of this proposal is to have each VIPR domain
publishing the list of methods supported by each of its phone numbers in the
RELOAD distributed database, in the existing VIPR-REGISTRATION resource record.

With this information, the PVP client immediately knows what methods to try from
the VIPR-REGISTRATION record, and the order to try them from the IANA registry,
and can also accumulate statistics on the usage of unsupported methods.

The PVP server can use this mechanism to simplify the process.  For example if
it knows that it will never receive the Caller-ID, it just have to never
advertize the "a" method.  Another example is with analog FXO ports that cannot
be used with either method "a" or "b".  In this case the VIPR server can still
advertize a new method using DTMF or fingerprint.

Note that because the methods supported per phone number are stored in the
RELOAD distributed database, the SIP call agent cannot retrieve them to decide
what method to use when starting a call, as it would take too much time to
retrieve it (same reason the validation is not done in real-time).  But it can
query them in the background using the history of calls or just before
scheduling a re-validation.

I would also propose to move the description of the generic PVP algorithm to the
- -framework document, to add a section in -framework explaining how to register a
new method on a IANA registry and to keep the two existing methods ("a" and "b")
into the -pvp draft.

B.6. PVP Entropy

A second issue related to PVP is the management of entropy (Section 10.1 of the
- -pvp draft).  The idea is that the fact that one validation succeeds is not
always enough for a remote VIPR server to give back routes and ticket for this
destination.  For example a VIPR domain could decide that at least 3 different
calls validated with method "b" are needed to let a SIP call reach the endpoint.

This proposal describes a mechanism for PVP to implement this.  The idea is that
when a PVP validation succeeds, the PVP server will return a ticket but may not
return the list of routes if the entropy threshold is not reached.  The ticket
will contain an opaque value set by the PVP server that contain the level of
entropy that this caller has reached.  The PVP client stores the ticket but does
not notify the call agent as there is no routes available.  The next time the
PVP client has a successful validation with the same destination, it sends the
saved ticket in addition to the domain.  The PVP server then evaluates the
ticket, increases the entropy value stored and sends back a new ticket.  If the
threshold has been reached, then the PVP server sends back the routes at the
same time, routes that the PVP client can now send with the ticket to the SIP
element.

When renewing the routes/ticket, the PVP client also sends the existing ticket,
so the PVP server can decide to lower the threshold based on the entropy
collected the previous time and the date of the ticket.

This can be combined with the proposal for method priority.  If multiple methods
are available for a destination but the first that succeeds does not return the
routes then the PVP client can use the next available methods in the list to try
to increase the entropy and receive them.

A PVP server may even use different thresholds, depending on the domains to
validate.  This then becomes an extension to the white/black list, where a
blacklist is implemented as a default threshold of X and an infinite threshold
for the domains blacklisted, and where a whitelist is implemented as a default
infinite threshold and a threshold of X for the domains whitelisted.

B.7. Ticket renewal

Another thing that was discussed in Quebec City was the "First Call Problem",
i.e. the problem that it takes up to 48 hours to verify a call and been able to
use enhanced media.  I think the conclusion of the discussion was that it was
not too annoying for the first call, as it did not degrade the user experience.
On the other hand, the cryptographic ticket granted after the first has an
expiration date and so need to be renewed, and that will degrade the user
experience as it would mean that for each destination, the end-user will
periodically not be able to use enhanced media for the duration of a call.

There was multiple proposals at the microphone to solve this problem, but I
would like to detail the one I proposed.

The idea is, sometimes before the expiration of the ticket, to make in parallel
a PSTN call and a SIP call using the existing SIP route and ticket for the
enhanced media (let's say video for the remaining of this discussion).  There is
already an existing I-D that can be use for this, draft-ietf-mmusic-sdp-cs.  The
way it could work is that the call agent, after retrieving the SIP route and
ticket for a destination, will decide that it is time to re-validate the route,
based for example on the frequency of calls to this number and the remaining
validity of the ticket.  The SIP element then adds an additional m= line to the
SDP that contains a PSTN address.  The offered SDP then looks something like this:

v=0
o=- 2890844526 2890842807 IN IP4 10.47.16.5
s=
c=IN IP4 10.47.16.5
t=0 0
m=audio 49170 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN - -
a=setup:active
a=connection:new
a=cs-correlation: uuie callerid dtmf
m=video 51372 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The SIP element on the other side verifies the ticket then starts to process the
SDP.  If it does not support this feature then, as per the rules in RFC 3264, it
will reject the second m= line by using port 0 in the answer SDP (In this case
the end user will have to have PSTN only calls for the 48 hours after the
expiration of the ticket).  If the remote SIP element supports this extension
then it will send back an offer like this:

v=0
o=- 0 1 IN IP4 192.168.2.1
s=
c=IN IP4 192.168.2.1
t=0 0
m=audio 8000 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN E164 +14085551234
a=setup:passive
a=connection:new
a=cs-correlation:uuie:2890W284hAT452612908awudfjang908 dtmf:14D*3
m=video 9000 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The local SIP element checks that the c= line contains the right number, then
establishes the audio and video media over IP, as usual, but also starts a PSTN
call following the rules in draft-ietf-mmusic-sdp-cs (i.e. sending the
correlation value as needed).  The remote SIP element will receive the PSTN
call, and correlate it with the existing SIP connection.  Both parties must send
audio on both the IP and the PSTN side, but the receivers can choose to play the
audio coming from either the IP or PSTN connection (this is because we may want
to use in the future fingerprint methods for the validation).  The PSTN call is
ended at the same time than the IP connection if the methods used for validation
are based on the call duration (other methods may permit to end the call
before), so the VCRs are processed as for an initial call.  The local party
marks the route as been under re-validation, so to not use the renewal method
for the next 48 hours.  The VIPR server will send a notification before the end
of the 48 hours if there is still a PSTN route to this destination.

B.8. Infrastructure overlays

A new draft to discuss infrastructure overlays has been submitted to DISPATCH.

"[RELOAD] is a peer to peer protocol developed by the P2PSIP Working
 Group.  Each RELOAD instance has a unique name, which is used by the
 process in section 10.2 of this specification to find the
 configuration servers, enrollment servers and bootstrap servers
 needed to join the overlay.  The process assumes that the RELOAD
 instance name is a FQDN, and uses the process in [RFC2782] (SRV RR)
 to find the IP address of the HTTPS server that serves the
 configuration document for this overlay.

 This process is adequate when the management of the overlay does not
 need to be distinguished from the owner of the FQDN used as the
 instance name, which is the case most of the time.  But there is a
 special class of overlays that, by definition, requires to be unique
 on the Internet and for which having the possibility of create
 instances would defeat their very purpose.  This specification calls
 the kind of overlays that are not domain specific, but application
 specific "infrastructure overlays".

 [VIPR] is a technology that is being standardized in the VIPR Working
 Group and that aims to build bridges between SIP islands by
 automatically provision SIP routes after the "ownership" of a PSTN
 phone number has been verified by an actual PSTN phone call.  This
 technology uses an RELOAD overlay as a distributed database where
 mappings between phone numbers and servers responsible for the
 validation process are stored.  The promise of VIPR to bridge these
 SIP islands cannot be fulfilled if there is more than one distributed
 database storing these mappings."

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From mperumal@cisco.com  Fri Oct  7 04:51:03 2011
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Thread-Topic: [VIPR] A new proposal for SIP Intermediaries
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Marc Petit-Huguenin" <petithug@acm.org>, <vipr@ietf.org>
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Subject: Re: [VIPR] A new proposal for SIP Intermediaries
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I agree with this proposal.

Muthu

|-----Original Message-----
|From: vipr-bounces@ietf.org [mailto:vipr-bounces@ietf.org] On Behalf Of
Marc Petit-Huguenin
|Sent: Monday, October 03, 2011 10:58 PM
|To: vipr@ietf.org
|Subject: [VIPR] A new proposal for SIP Intermediaries
|
|-----BEGIN PGP SIGNED MESSAGE-----
|Hash: SHA1
|
|For the second time, the Design Team conference call got stuck on the
subject of
|SIP intermediaries in VIPR, and we were not able to make any progress
on the
|VIPR specifications themselves.  I was supposed to send another email
to this
|mailing-list to describe in more details the problems created by SIP
|intermediaries but I fear that the next conference calls will still
being spent
|arguing about this.  So I would like to propose an alternate route to
solve this
|issue.
|
|SIP intermediaries are not described at all in the current
specifications, but I
|think that everybody agree that we should say something about it,
because it
|will become a reality, even if we decide to make it a SHOULD NOT (which
would
|exclude a lot of potential participants from the VIPR federation).  On
the other
|hand, VIPR domains does not require SIP intermediaries to work.
|
|So my proposal is:
|
|1. Put everything related to SIP intermediaries in a separate I-D, so
it does
|not hold the main specifications.
|2. Add in the main specifications the "hooks" required to be able to
add support
|for SIP intermediaries as an extension in the future (as opposed to
have to work
|on a new version of the VIPR RFCs to add SIP intermediaries, which will
push
|back the deployment of SIP intermediaries for another 3 years).
|3. Add a new milestone for the SIP intermediaries (I would suggest
April 2012[1])
|
|Please let me know as soon as possible if you agree with this new plan,
at least
|on the two first points, so I can release the first version of the new
|draft-petithuguenin-vipr-sip-intermediaries I-D, and so we can move one
on the
|other subjects that are actually blocking the work on -framework.
|
|Thanks.
|
|
|[1] And I would also suggest to move at the same time the "Submit
Specification
|of authorization tokens to mitigate SPAM for publication as Proposed
Standard"
|to December 2011, as it make sense to submit this at the same time than
the
|other core VIPR specifications.
|
|- --
|Marc Petit-Huguenin
|Personal email: marc@petit-huguenin.org
|Professional email: petithug@acm.org
|Blog: http://blog.marc.petit-huguenin.org
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On 10/07/2011 04:49 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:
> Hi Mark,
> 
> |When an enterprise uses a provider, it is paying for |the service of routing
> the calls to the PSTN, so why |advertizing the number in the VIPR overlay
> could not |be part of this cost?
> 
> Sure, if it adds value to the enterprise and the provider can monetize it, I
> see no issue (it could be a provider hosted ViPR service, for e.g). At the
> first level, the enterprise might want to require that the provider published
> route be easily distinguishable from the enterprise published route, so that
> if both routes are available for someone calling into the enterprise they can
> chose one among them. I think in the majority of cases one would prefer a
> direct route, unless the provider offers differentiated services like
> audio/video transcoding, rate adaption, echo cancellation etc that a direct
> route doesn't have.

Yes, but the difficulty here is how to choose this route.  The proposal sent by
Hakim, i.e. sending the capabilities of the endpoint together with the SIP route
and ticket, would permit this, as only the VIPR domain the closest to the end
user would be able to send the real capabilities of the endpoint (extracted for
example from the SIP registrar).  The transcoding is interesting, as an
intermediary using it could eventually claim to better route than the direct
route.  Perhaps what we need is a new media feature tag that would indicate that
a capability set is "native" (i.e. built directly from the SIP registrar or
information coming from the endpoint), instead of been from a intermediate media
point, like transcoding.

> 
> Muthu
> 
> |-----Original Message----- |From: Marc Petit-Huguenin
> [mailto:petithug@acm.org] |Sent: Monday, October 03, 2011 9:56 PM |To: Muthu
> Arul Mozhi Perumal (mperumal) |Cc: vipr@ietf.org |Subject: Re: Agenda for
> 2011/09/23 | |-----BEGIN PGP SIGNED MESSAGE----- |Hash: SHA1 | |Hi Muthu, | 
> |On 09/28/2011 11:10 PM, Muthu Arul Mozhi Perumal (mperumal) wrote: |> A
> fundamental question: Why would a provider want to advertise routes for a |>
> number for which neither of the enterprises is willing to pay for? | |I am
> not sure why you are saying that.  When an enterprise uses a provider, it |is
> paying for the service of routing the calls to the PSTN, so why advertizing 
> |the number in the VIPR overlay could not be part of this cost? | |> Isn't
> ViPR based on the fact the provider is trusted for the first (PSTN) |> call?
> If the provider is fooling with the enterprise, won't the enterprise |>
> switch to a different provider? Or are we concerned about the security |>
> implications around routing subsequent calls through the provider? Aren't |>
> they applicable to the first call as well? |> |> If the enterprise doesn't
> trust the provider then ViPR in its current form |> doesn't seem to make
> sense. |> |> Muthu |

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From dean.willis@softarmor.com  Fri Oct  7 09:19:58 2011
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On 10/7/11 8:51 AM, Marc P
> Yes, but the difficulty here is how to choose this route.  The proposal sent by
> Hakim, i.e. sending the capabilities of the endpoint together with the SIP route
> and ticket, would permit this, as only the VIPR domain the closest to the end
> user would be able to send the real capabilities of the endpoint (extracted for
> example from the SIP registrar).  The transcoding is interesting, as an
> intermediary using it could eventually claim to better route than the direct
> route.  Perhaps what we need is a new media feature tag that would indicate that
> a capability set is "native" (i.e. built directly from the SIP registrar or
> information coming from the endpoint), instead of been from a intermediate media
> point, like transcoding.


If we create a situation where there is an incentive for an intermediary 
to mis-state a capability set in order to "capture" more 
revenue-generating traffic, then the intermediaries are going to lie 
like politicians seeking re-election.

I think this is a MUCH harder problem than we currently realize.

Even Hakim's proposal requires that route-validation also validate the 
offered capabilities, not just evaluate them and choose the richest set. 
This is hard; a motivated intermediary could probably convince a 
test-bot that it is the target instead of an intermediary.

--
Dean

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VIPR Design Team conference call on Friday October 7th, from 9:05am to 9:55am PST.

o Attendees:
  Mary Barnes
  Daryl Malas
  Hakim Mehmood
  Muthu Arul Mozhi Perumal
  Jon Peterson
  Marc Petit-Huguenin
  Michael Procter
  John Ward
  Dean Willis

o Agenda

- - Move SIP intermediaries in a future draft (5 min)
- - RELOAD registration (5 min)
- - Replacement ticket by client certificate (10 min)
- - PVP privacy (10 min)
- - PVP method priority and registration (10 min)
- - PVP method entropy (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (5 min)

o Action items

- - Treat SIP interworking as an extension.
- - Remove the text saying that the phone number must be stored 3 times at the
VIPR level.
- - Explain the problem and requirements for the ticket mechanism.

o Minutes

- - Move SIP intermediaries in a future draft

Marc: The proposal is to treat SIP intermediaries as an extension to VIPR, in a
separate draft, but to be sure that we have the required hooks in the VIPR
drafts to not have to modify VIPR later.  The last part of the proposal is to
add a milestone for this after finishing the work on the base VIPR.

Jon: The viability of this proposal would depends on what the hooks are.  Second
I would be careful to not put this in term of SIP intermediaries, a more generic
problem is inter-working. But I am really interested on what the hooks are.

Marc: I do not know what the hooks would be.  I just published what the problem
is, as I said I would do in the previous meeting. I will release another version
that will contain multiple solutions and we can see from here what the hooks
will be, if there is any needed.

Dean: If we are going to place a lot of requirements, it becomes a location
server.  It's a simple lookup operation.

Jon: Documenting the hooks in the intermediary draft is one thing but the hooks
will be in the base draft, we need to be clear on this.  We need to not scope
this work too narrowly.

Marc: Do we agree to work on this later?

Mary: Yes.

Dean:  I disagree slightly with Jon. One hop is straightforward, but more
raises a set of transitive trust issues.

Jon: If we have two overlays then we have a disaster.  Keep the milestone high
level.

- - RELOAD registration

Marc: The current spec is saying that each phone number is registered 3 times,
which with the normal RELOAD replication, means that it is stored 9 times.  So
the question was what to do when there is a difference between the copies
retrieved.  The proposal was to require to have at least two identical copies.
But the discussion moved to why do we require 3 copies in the first place.  So
the next proposal was to store the phone number one time, to verify that there
was 3 identical replicas, and to use the voting algorithm when retrieved.

Hakim: What does identical mean?

Marc: Which return the same Node-ID.  The idea is that a rogue peer should not
prevent PVP to work.  But the discussion was more about the requirement of
having 3 copies at the VIPR level.

Hakim:  Does it really mater? The data in the overlay is not trusted.

Marc: Do we want to say that all nodes need to register 3 times or keep it
simple, i.e. one time?  It does not make a difference when storing, but the
nodes that will fetch need to know the number of copies.  One way it can be
solved is to put a parameter in the configuration file that indicate the number
of copies at the VIPR level.

Jon: Did we ever find why there was 3 copies?

Hakim: That was to increase the security - I am not sure.

Michael: It is easy to retrieve the replica?

Marc: Yes, what you do is that you put the Node-ID of the successor in the Fetch
request, instead of using just the Resource-ID.

Michael:  If the first node is compromised, why should I trust it to give me the
successor?

Marc: An excellent question, but more of a RELOAD problem.

Michael: So RELOAD protects against network outages, no malicious nodes.  So to
go back to Jon's point, we will validate any request anyway because we do not
trust anything from RELOAD.

Hakim: Yes, the idea was to protect in case a copy is lost.

Jon: OK, so just need one copy.

Hakim: It does not matter for an application point of view.

Jon: It is related to the SIP intermediaries.

Hakim: No its a dictionary storage.  Also having multiple copies permits to
increase the trust in a copy.

Marc: So one proposal would be to add a variable ion the configuration file that
contain the number of copies at the VIPR level.

Michael: I am reluctant to add a configuration variable.  I do not think we need
to replicate at the VIPR level, because we will verify everything we get.

Marc:  OK so everybody agree to remove the duplication at the VIPR level.

- - Replacement ticket by client certificate

Marc: The first problem with the ticket is that the ticket is signed using a
symmetrical key, key that need to be shared between the PVP server and the call
agent.  The second problem is that the ticket is separated from the TLS
certificate, so you have to validate the TLS connection, then verify the ticket.
 Another problem is that there is not existing proven code for processing the
ticket.  So the idea is to replace the ticket by an X.509 certificate, which
already contains all the fields required by the ticket, can use asymmetrical
keys, and has existing proven libraries.

Jon: This is a significant change, and we would need more time.  We should not
diverge too much from what SIP is doing.  TLS is not use much so pushing all the
security features into TLS might not be a guarantee of widespread success.  How
do you push this certificate onto systems that want to use it, and how do you
capture the parameters of the call so that you verify that it is the right
person you are talking to.

Marc: To push the certificate, you just send it instead of the ticket.  As for
the significant change, I will publish code that shows that it is less work to
use a certificate than a ticket.

Hakim: If sharing the secret is a concern, we should consider diverging?

Marc: It is a concern for me.

Jon: It's unclear to me why you need to share the symmetrical key.  The proxy
could just pass the ticket to the element that can verify it.

Michael: A middle ground would be to use public keys to sign the ticket.

Marc: It's not like there is 100's of implementations that need to be modified.

Hakim: I agree with Jon that keeping the semantics of SIP is important.  If that
means changing whatever Cisco implemented, it is fine.

Marc: A certificate is not that different from a ticket using asymmetrical keys.

Hakim: If it falls within regular TLS, then OK.

Jon: If the symmetrical key is the problem, then we need to go back to the
requirements.  This proposal is worth considering, but there is a huge space of
solutions that can be considered.

Marc: One problem is that in the current version of VAP there is no possibility
to verify the ticket at another place than on the call agent.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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On 10/07/2011 09:23 AM, Dean Willis wrote:
> On 10/7/11 8:51 AM, Marc P
>> Yes, but the difficulty here is how to choose this route.  The proposal sent by
>> Hakim, i.e. sending the capabilities of the endpoint together with the SIP route
>> and ticket, would permit this, as only the VIPR domain the closest to the end
>> user would be able to send the real capabilities of the endpoint (extracted for
>> example from the SIP registrar).  The transcoding is interesting, as an
>> intermediary using it could eventually claim to better route than the direct
>> route.  Perhaps what we need is a new media feature tag that would indicate that
>> a capability set is "native" (i.e. built directly from the SIP registrar or
>> information coming from the endpoint), instead of been from a intermediate media
>> point, like transcoding.
> 
> 
> If we create a situation where there is an incentive for an intermediary to
> mis-state a capability set in order to "capture" more revenue-generating
> traffic, then the intermediaries are going to lie like politicians seeking
> re-election.
> 
> I think this is a MUCH harder problem than we currently realize.
> 
> Even Hakim's proposal requires that route-validation also validate the offered
> capabilities, not just evaluate them and choose the richest set. This is hard; a
> motivated intermediary could probably convince a test-bot that it is the target
> instead of an intermediary.

Perhaps one way to solve this would be to lower the priority of a route if a
subsequent SIP call made using it does not match the capabilities advertized.
It could also be done with an OPTIONS, but the response can also be faked.  It's
a little bit more difficult to fake the actual video (I suppose that wideband
audio can be faked by resampling the narrowband signal).

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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On 10/11/11 2:07 PM, Marc Petit-Huguenin wrote:

> Perhaps one way to solve this would be to lower the priority of a route if a
> subsequent SIP call made using it does not match the capabilities advertized.
> It could also be done with an OPTIONS, but the response can also be faked.  It's
> a little bit more difficult to fake the actual video (I suppose that wideband
> audio can be faked by resampling the narrowband signal).

My local broadcaster sends a lot of fake HDTV by just upsampling the SD 
signal.

The problem is that a transit provider might get "greedy" and advertise 
capabilities they can't deliver in order to get more traffic. But the 
ability of a route to support a capability says nothing about what 
capability the called party might allow on a given call, so a caller 
can't just deprioritize a route when a completed call receives less 
capability than the route advertised. Nor can the called party withdraw 
validation support for a route just because it receives low-capability 
calls over that route, because the callers themselves might have low 
capabilities (or desires).

Essentially, we're trying to detect a MITM "capability bid-down" attack. 
This might be a generic SIP problem, not just a ViPR problem, although 
we could perhaps use ViPR to help cure it.

If we had a way for the caller to securely express the capabilities that 
the route is offering, then the callee could compare those capabilities 
to what it thinks the route should be offering, and thereby filter out 
"bad" routes.

However, every attempt I make to solve this falls into "but the greedy 
intermediary could fake it" because they have access to the only actual 
authentication data in the system. It's a catch-22; by giving a transit 
provider access to your call verification data, you give them the keys 
to the city.

The only answer I have so far is that one could audit the routes 
pointing at oneself and get upset about bogus/misleading entries.

--
Dean

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During the last conference call, this proposal was criticized as been a
significant change from the way SIP is processing TLS requests.  My research
shows that it is not the case.

First of all, let's define exactly what SIP, without even talking about VIPR, is
doing regarding TLS.  The best place for this is RFC 5922.  We can skip the
client behavior discussion (section 7.3) because neither VIPR or my proposal
suggest any modification to this procedure.  Here's what RFC 5922 says about the
server behavior (aka client certificate):

"  When a server accepts a TLS connection, the server presents its own
   X.509 certificate to the client.  Servers that wish to authenticate
   the client will ask the client for a certificate.  If the client
   possesses a certificate, that certificate is presented to the server.
   If the client does not present a certificate, the client MUST NOT be
   considered authenticated.

   [...]

   When authenticating the client, the server MUST obtain the set of SIP
   domain identities from the client certificate as described in
   Section 7.1.  Because the server accepted the TLS connection
   passively, unlike a client, the server does not possess an AUS for
   comparison.  Nonetheless, server policies can use the set of SIP
   domain identities gathered from the certificate in Section 7.1 to
   make authorization decisions.

   For example, a very open policy could be to accept an X.509
   certificate and validate the certificate using the procedures in RFC
   5280 [6].  If the certificate is valid, the identity set is logged.

   Alternatively, the server could have a list of all SIP domains the
   server is allowed to accept connections from; when a client presents
   its certificate, for each identity in the client certificate, the
   server searches for the identity in the list of acceptable domains to
   decide whether or not to accept the connection.  Other policies that
   make finer distinctions are possible."

In other words, the SIP server (or proxy) does whatever it thinks is necessary
with the client certificate, and it does not seems that even following RFC 5280
is required (note in particular that nowhere it is said that the validation must
use a public trust anchor).

VIPR itself adds very little to the requirements of RFC 5922.  Here is what
- -sip-antispam says about TLS and the ticket:

"  The Border Element will receive the TLS ClientHello which begins the
   TLS handshake.  The Border Element will present its own configured
   cert.  Once TLS handshaking is complete, the Border Element notes the
   domain from the SubjectAltName on the other side of the TLS
   connection, and associates it with that connection.

   Next, the Border Element will receive an INVITE.  This INVITE will
   contain a ticket in the ViPR-Ticket header field value.  The Border
   Element extracts this header field.  This call flow assumes it is
   present.  The Border Element parses it, and obtains the epoch value
   encoded in the ticket.  This is matched against the current epoch
   value for the configured password.  If they match, processing
   continues.  The Border Element verifies the signature is correct.
   Next, it examines the start and stop time of the validity.  If the
   current time is between the start and stop times, the check is
   passed.  Next, the Border Element checks the granted-to domain in the
   ticket.  It compares that domain against the domain name in the
   SubjectAltName of the peer on the other side of the TLS connection,
   as noted above.  Next, it takes the Request-URI of the SIP INVITE.
   That will be of the form sip:+number@domain.  If it is not in that
   form, and if the number does not begin with a plus, the request is
   dropped.  The value, including the plus, is then compared to the
   number in the ticket.  If it is equal, the check has passed.  The
   Border Element leaves the header field in the request, but forwards
   to the Call Agent."

The only policy that is added to what RFC 5922 says is that the granted-to
domain in the ticket must be equal to the domain in the SubjectAltName of the
client certificate.  Here too, not a word about a public trust anchor.

Now back to my proposal.  The first thing is that the client certificate must be
validated using the CA of the VIPR server as trust anchor, which is already what
a SIP server could do.  It validates according to the rules in RFC 5280, which
is equivalent to verifying the signature of the ticket and the start/stop
validity.  The SubjectAltName in the client certificate (which is not even
required to be used in standard SIP) can be checked against the domain part of
the From header of the INVITE.  The phone number and the granting-domain (which
not even verified in VIPR) can be stored as an URI in the Issuer Alternative
Name extension in the client certificate and be validated against the
Request-URI of the INVITE.  The granting node can be also be stored as a reload
URI in an Issuer Alternative Name extension.

So in conclusion, my proposal does not change what a normal SIP stack is doing,
because the specification does not say what a SIP stack should do with a client
certificate.

(Note that to -sip-antispam should also be added some text saying that RFC 5923
must be used, because there is no way to create an additional TLS connection in
the other direction for a subsequent SIP transaction).

On 09/03/2011 12:06 PM, Marc Petit-Huguenin wrote:
> In the current specification, the PVP server builds and signs a VIPR ticket, and
> send it back to the PVP client, that passes it to the Call Agent/ Border Element
> (CA/BE).  The CA/BE will insert this ticket in the next SIP INVITE to the remote
> CA/BE, which will check the validity of the ticket and the domain of the TLS client.
> 
> There is multiple things that bother me with this process:
> 
> - The verification of the ticket use HMAC-SHA1, and so the same password has to
> be provisioned in both the CA/BE(s) (to verify the ticket) and the VIPR
> server(s) (to generate the ticket).
> - The ticket can be used without TLS (we all know that implementers think of
> MUST as SHOULD and SHOULD as MAY).
> 
> I would like to propose an alternative that is simpler to implement and, I
> think, more secure than the ticket.
> 
> Instead of using a symmetrical key, the local VIPR server generates an RSA key
> pair, with the public key distributed to the CA/BE(s).  After a successful PVP
> connection, the remote PVP client sends a PKCS#10 certificate request,
> containing the domain name and signed with a private key[1].  The local PVP
> server generates a new certificate[2], sign it with its private key and send it
> back to the remote PVP client, which pass it back to the remote CA/BE.
> 
> The remote CA/BE will use the certificate received to establish the TLS
> connection for the SIP transactions to the local CA/BE, which will use a
> standard CA based certificate for its domain.  The local CA/BE will be able to
> verify with the VIPR public key that it issued the certificate, then will
> validate the standard parts of the certificate (expiration date and so on).
> 
> There is multiple advantages to this proposal:
> 
> - The private key is stored only in the VIPR server, or better in a physical
> token.
> - VIPR does not work without TLS for the SIP connection.
> - There is plenty of commercial or free TLS and X.509 implementations that have
> years of experience and testing, which is not the case for the ticket proposal.
> - Security agility is provided by TLS.
> - No additional SIP Header.
> - Because the ticket verification process is folded into the client certificate
> verification, there is less resources needed.
> 
> 
> [1] This is the private key of the CA/BE, so the CA/BE needs to upload the
> PKCS#10 file in the VIPR server.  Depending on the architecture used, each CA/BE
> can use a different PKCS#10 or the same, which means that the PVP server must be
> able to process more than one PKCS#10 and send back more than one certificate on
> one PVP connection.
> 
> [2] The certificate sent back can directly contain the SIP routes, increasing
> the overall security.  One possible mapping from the ValExchange response to the
> certificate could be:
> 
> - Unique ID: serialNumber
> - Epoch: issuer
> - salt: N/A
> - integrity: signature
> - Validity: validity
> - Phone Number: subjectAltName uniformResourceIdentifier tel:+14085551234
> - Node-ID: subjectAltName uniformResourceIdentifier
> reload://0110123456789ABCDEF0123456789ABCDEF/
> - Granted domain: subjectAltName dNSName example.com
> - Granting domain: implicit in routes
> - Route(s): subjectAltName uniformResourceIdentifier
> sip:+14085551234@example.org:371;maddr=10.1.1.1
> 


- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From michael@voip.co.uk  Thu Oct 13 00:29:49 2011
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On 12 October 2011 23:53, Marc Petit-Huguenin <petithug@acm.org> wrote:
> During the last conference call, this proposal was criticized as been a
> significant change from the way SIP is processing TLS requests. =A0My res=
earch
> shows that it is not the case.
[...]
> So in conclusion, my proposal does not change what a normal SIP stack is =
doing,
> because the specification does not say what a SIP stack should do with a =
client
> certificate.

What you describe is that the server side of the connection will need
to validate a client certificate whatever happens, and changing the
ticket to a certificate does not substantively change the validation
process.  It does force the ticket validation to the edge of your
network though, which having the ticket in a header field doesn't.
Since ticket validation is a new process introduced with ViPR, this
may be a reasonable change.

However, what is overlooked in this analysis is that the client side
must present a different certificate on a per-call basis.  Most
implementations that I have seen tend to have a fairly statically
configured certificate (or several, but not lots) that is valid for
the duration of the process/between scheduled maintenance.  Having the
certificate change on a per-call basis is a change for these
implementations.  How significant a change depends on their
architecture, but it is a change nonetheless.

Why does this matter?  With the ViPR ticket in a header field, I can
route ViPR traffic out in much the same way as I route other SIP
traffic.  Granted it has an additional header, but that doesn't
substantively change the job my egress points perform.  With a ViPR
certificate instead of a ticket, my egress points need to be
sufficiently ViPR-aware to know to use a dynamically selected
certificate, and how to get hold of that certificate.  Not
insurmountable problems, but ones that are introduced by changing to a
certificate over a simple header field.

There may be other problems (Jon seemed to jump on the change quite
fast so he may have more reasons), but this is the one that struck me.
 I think this change warrants wider discussion before deciding whether
to adopt or not.

Regards,

Michael

From petithug@acm.org  Thu Oct 13 07:54:11 2011
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On 10/13/2011 12:29 AM, Michael Procter wrote:
> On 12 October 2011 23:53, Marc Petit-Huguenin <petithug@acm.org> wrote:
>> During the last conference call, this proposal was criticized as been a
>> significant change from the way SIP is processing TLS requests.  My research
>> shows that it is not the case.
> [...]
>> So in conclusion, my proposal does not change what a normal SIP stack is doing,
>> because the specification does not say what a SIP stack should do with a client
>> certificate.
> 
> What you describe is that the server side of the connection will need
> to validate a client certificate whatever happens, and changing the
> ticket to a certificate does not substantively change the validation
> process.  It does force the ticket validation to the edge of your
> network though, which having the ticket in a header field doesn't.
> Since ticket validation is a new process introduced with ViPR, this
> may be a reasonable change.
> 
> However, what is overlooked in this analysis is that the client side
> must present a different certificate on a per-call basis.  Most
> implementations that I have seen tend to have a fairly statically
> configured certificate (or several, but not lots) that is valid for
> the duration of the process/between scheduled maintenance.  Having the
> certificate change on a per-call basis is a change for these
> implementations.  How significant a change depends on their
> architecture, but it is a change nonetheless.

True.

> 
> Why does this matter?  With the ViPR ticket in a header field, I can
> route ViPR traffic out in much the same way as I route other SIP
> traffic.  Granted it has an additional header, but that doesn't
> substantively change the job my egress points perform.  With a ViPR
> certificate instead of a ticket, my egress points need to be
> sufficiently ViPR-aware to know to use a dynamically selected
> certificate, and how to get hold of that certificate.  Not
> insurmountable problems, but ones that are introduced by changing to a
> certificate over a simple header field.

Another argument to add to your point is that using a different certificate for
each different phone number reduces the possibility of session caching or
session resumption.

> 
> There may be other problems (Jon seemed to jump on the change quite
> fast so he may have more reasons), but this is the one that struck me.
>  I think this change warrants wider discussion before deciding whether
> to adopt or not.

OK, so I think that there is a way to keep the properties of the ticket (can be
processed separately from the TLS server processing, does not require to
dynamically choose the certificate on the client side), and still let "advanced"
implementations not use this.

The first step is to simply say that an X.509 certificate is a better
replacement for the ticket as defined currently, independently to its usage in TLS:

- - Existing and proven libraries.
- - Support for asymmetrical keys.
- - Security agility.

With this first step, we use the X.509 certificate just like the current ticket,
as a container to carry the parameters in the ticket.  This means that the
border element on the originating side establishes a TLS connection using a
static client certificate, and then send an INVITE with a vipr-ticket header
containing the X.509 certificate received from the PVP transaction and encoded
in base64.  The border element on the receiving side verifies the client
certificate as per RFC 5280, then verifies the certificate inside the
vipr-ticket (or pass it to another element to verify this second certificate).
At this point we have exactly the same process than before, expected for the
container used to carry the ticket values.

Now let's add two very small modifications on the server processing side:

- - The VIPR server CA is added to the list of public trust anchor that the TLS
server uses to validate the client certificate.
- - If there is no vipr-ticket in the INVITE, the client certificate received from
the other side is used as if it was received in a vipr-ticket header field.

With this, we now can have the TLS client choose to either work as a standard
SIP client, i.e. it uses its static client certificate for the TLS connection,
and put the certificate received from PVP in the vipr-ticket field, or work more
dynamically, by using directly the certificate received from PVP as the client
certificate for the TLS connection.  On the server side all the processing can
be done at one point, or it can have the ticket processing delegated to another
server.

This way we have the best of the two worlds.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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A. Agenda

- - Replacement ticket by client certificate (10 min)
- - PVP method priority and registration (10 min)
- - PVP privacy (10 min)
- - PVP method entropy (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (5 min)

Note that the sum of the time allocated to each topic exceeds the time allocated
to the whole call.  The topics that cannot make it to the discussion will be
postponed to the next conference call if they are not subsequently decided in
the mailing-list.

Because the conference bridge as a limited capacity, an individual invitation to
the conference will be sent shortly after this email to each member of the
design team.  Please send me an email if you are not on this list and want to
receive an invitation.

Note that the design team conference call is not a way to short-circuit the
normal process, but a way to accelerate it by increasing face to face time.  All
decisions made must be confirmed on the mailing-list.

This conference call is subject to the rules of RFC 5378 and RFC 3979 (updated
by RFC 4879).


B. Handout (reading time: 10 min)

B.1. Ticket vs Client certificate

In the current specification, the PVP server builds and signs a VIPR ticket, and
sends it back to the PVP client, which passes it to the SIP element.  The SIP
element will insert this ticket in the next SIP method to the remote SIP
element, which will check the validity of the ticket and the domain of the TLS
client.

The problem with this process is that the verification of the ticket use
HMAC-SHA1, and so the same password has to be provisioned in both the SIP
element (to verify the ticket) and the PVP server (to generate the ticket).
Also the ticket uses an adhoc format, for which no code currently exist, and
there is no provision for security agility.

So the proposal would be to replace the ticket by a client certificate, which
could work like this:

Instead of using a symmetrical key, the local PVP server generates an RSA key
pair, with the public key distributed to the SIP element(s).  After a successful
PVP connection, the remote PVP client sends a PKCS#10 certificate request,
containing the domain name and signed with a private key.  The local PVP server
generates a new certificate, sign it with its private key and send it back to
the remote PVP client, which pass it back to the remote SIP element.  The
SubjectAltName contains the granted-by field, and the phone number and the
granting-domain are stored as one URI in the Issuer Alternative Name extension,
and the granting node is also be stored as a reload URI in another Issuer
Alternative Name extension.

With this first step, we use the X.509 certificate just like the current ticket,
as a container to carry the parameters in the ticket.  This means that the
border element on the originating side establishes a TLS connection using a
static client certificate, and then send an INVITE with a Vipr-Ticket header
containing the X.509 certificate received from the PVP transaction and encoded
in base64.  The border element on the receiving side verifies the client
certificate as per RFC 5280 (using public trust anchors), then verifies the
certificate inside the Vipr-Ticket (or pass it to another element to verify this
second certificate), using RFC 5280 with the PVP CA certificate as trust anchor
and by using the same algorithm than currently. At this point we have exactly
the same process than before, expected for the container used to carry the
ticket values.

Now let's add two very small modifications on the server processing side:

- - The VIPR server CA is added to the list of public trust anchor that the TLS
server uses to validate the client certificate.
- - If there is no Vipr-Ticket in the INVITE, the client certificate received from
the other side is used as if it was received in a Vipr-Ticket header field.

With this, we now can have the TLS client choose to either work as a standard
SIP client, i.e. it uses its static client certificate for the TLS connection,
and put the certificate received from PVP in the Vipr-Ticket field, or work more
dynamically, by using directly the certificate received from PVP as the client
certificate for the TLS connection.  On the server side all the processing can
be done at one point, or it can have the ticket processing delegated to another
server.

B.2. PVP Methods Registry

On the PVP subject, The VIPR specification currently defines two methods for the
PSTN validation:  Method "a", which is used when the Caller-ID is available and
method "b", which is used when it is not.  The PVP draft also permits to define
new validation methods, but it does not explain when and how to use this
extended methods.

This proposal is in two parts.  First we need to establish a IANA registry for
the PVP methods, and to assign a priority to each of these methods.  Let's say
for now that the priority is between -∞ (lowest priority) to +∞ (highest
priority) and that priorities should be assigned by steps of 100.  With this
scale we could assign priority 0 to method "b" and priority 100 to method "a".
Then we say in the spec that a VIPR server must try available methods starting
from the highest priority to the lowest priority, until one succeeds or all
fail.  This implements the same algorithm that is currently in -pvp ("a" first,
then "b").

Now the problem with this is that the number of methods will increase in the
future, but we may not want that the PVP client tries all the methods available
each time.  So the second part of this proposal is to have each VIPR domain
publishing the list of methods supported by each of its phone numbers in the
RELOAD distributed database, in the existing VIPR-REGISTRATION resource record.

With this information, the PVP client immediately knows what methods to try from
the VIPR-REGISTRATION record, and the order to try them from the IANA registry,
and can also accumulate statistics on the usage of unsupported methods.

The PVP server can use this mechanism to simplify the process.  For example if
it knows that it will never receive the Caller-ID, it just have to never
advertize the "a" method.  Another example is with analog FXO ports that cannot
be used with either method "a" or "b".  In this case the VIPR server can still
advertize a new method using DTMF or fingerprint.

Note that because the methods supported per phone number are stored in the
RELOAD distributed database, the SIP call agent cannot retrieve them to decide
what method to use when starting a call, as it would take too much time to
retrieve it (same reason the validation is not done in real-time).  But it can
query them in the background using the history of calls or just before
scheduling a re-validation.

I would also propose to move the description of the generic PVP algorithm to the
- -framework document, to add a section in -framework explaining how to register a
new method on a IANA registry and to keep the two existing methods ("a" and "b")
into the -pvp draft.

B.3. PVP Privacy

As reported by Michael Procter, there is ways with the current PVP methods to
discover valuable information about a competitor.  But first let's add some
common vocabulary related to PVP:

- - The PVP selector is a list of parameters that are sent by the PVP client side
to retrieve a set of call records in the PVP server side.  The resulting set can
be empty, in which case the validation fails, can contain one element, or can
contain multiple elements in which case the method description must also defines
what call record should be selected (e.g. in the case of method "a", the most
recent call record is selected).  One important point is that the selector is
always passed from the PVP client to the PVP server in clear, and so this is
where the privacy leakage happens.

- - The PVP parameter is a list of name/value pairs that are passed in clear
between the PVP client and PVP server to aid the verification.  Examples are the
rounding value and the vservice id for methods "a" and "b".

- - The PVP secret is the information that each side of the PVP transaction tries
to verify.  The algorithm used (SRP) uses a zero-knowledge password proof, so
neither side can deduce the secret if the verification fails.  In the case of
methods "a" and "b" the secret is the rounded start and end time for the call,
but there is a lot of other possible secrets that can be used in new methods
(DTMF, UUIE, fingerprint, SMS hash, etc...).

In the privacy issue, the problem is that callee number and caller numbers are
disclosed in the PVP selector for method "a" and the callee number is disclosed
in the PVP selector for method "b".

One proposed solution would be to hash these numbers, but finding the phone
numbers would still be trivial for a determined competitor.  Passing the hash
salt as a PVP parameter and using an adaptive hash method like bcrypt will force
the PVP server to rehash all the current terminating call record, but will force
a competitor to rehash probably half of the total E.164 space to find the
numbers.  This is still possible but can provide an adequate privacy protection
level.

Another possibility would be to invert the selector and the secret.  After all
the star/stop time of a call is not as sensitive as the caller/callee phone
numbers.  In this method the PVP selector would be the start/stop time and the
secret would be the caller/callee numbers.  The problem is that it will be more
difficult to design an algorithm to select a unique call record if the returned
set contains more than one.  One solution could be to add more data in the PVP
selector, for example by adding the call initiation time and the direction of
the termination.

We do not even need to decide - we can define these two algorithms as different
methods (in addition to "a" and "b") and let the VIPR domain choose (see PVP
registry proposal).

B.4. PVP Entropy

Another issue related to PVP is the management of entropy (Section 10.1 of the
- -pvp draft).  The idea is that the fact that one validation succeeds is not
always enough for a remote VIPR server to give back routes and ticket for this
destination.  For example a VIPR domain could decide that at least 3 different
calls validated with method "b" are needed to let a SIP call reach the endpoint.

This proposal describes a mechanism for PVP to implement this.  The idea is that
when a PVP validation succeeds, the PVP server will return a ticket but may not
return the list of routes if the entropy threshold is not reached.  The ticket
will contain an opaque value set by the PVP server that contain the level of
entropy that this caller has reached.  The PVP client stores the ticket but does
not notify the call agent as there is no routes available.  The next time the
PVP client has a successful validation with the same destination, it sends the
saved ticket in addition to the domain.  The PVP server then evaluates the
ticket, increases the entropy value stored and sends back a new ticket.  If the
threshold has been reached, then the PVP server sends back the routes at the
same time, routes that the PVP client can now send with the ticket to the SIP
element.

When renewing the routes/ticket, the PVP client also sends the existing ticket,
so the PVP server can decide to lower the threshold based on the entropy
collected the previous time and the date of the ticket.

This can be combined with the proposal for method priority.  If multiple methods
are available for a destination but the first that succeeds does not return the
routes then the PVP client can use the next available methods in the list to try
to increase the entropy and receive them.

A PVP server may even use different thresholds, depending on the domains to
validate.  This then becomes an extension to the white/black list, where a
blacklist is implemented as a default threshold of X and an infinite threshold
for the domains blacklisted, and where a whitelist is implemented as a default
infinite threshold and a threshold of X for the domains whitelisted.

B.5. Ticket renewal

Another thing that was discussed in Quebec City was the "First Call Problem",
i.e. the problem that it takes up to 48 hours to verify a call and been able to
use enhanced media.  I think the conclusion of the discussion was that it was
not too annoying for the first call, as it did not degrade the user experience.
On the other hand, the cryptographic ticket granted after the first has an
expiration date and so need to be renewed, and that will degrade the user
experience as it would mean that for each destination, the end-user will
periodically not be able to use enhanced media for the duration of a call.

There was multiple proposals at the microphone to solve this problem, but I
would like to detail the one I proposed.

The idea is, sometimes before the expiration of the ticket, to make in parallel
a PSTN call and a SIP call using the existing SIP route and ticket for the
enhanced media (let's say video for the remaining of this discussion).  There is
already an existing I-D that can be use for this, draft-ietf-mmusic-sdp-cs.  The
way it could work is that the call agent, after retrieving the SIP route and
ticket for a destination, will decide that it is time to re-validate the route,
based for example on the frequency of calls to this number and the remaining
validity of the ticket.  The SIP element then adds an additional m= line to the
SDP that contains a PSTN address.  The offered SDP then looks something like this:

v=0
o=- 2890844526 2890842807 IN IP4 10.47.16.5
s=
c=IN IP4 10.47.16.5
t=0 0
m=audio 49170 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN - -
a=setup:active
a=connection:new
a=cs-correlation: uuie callerid dtmf
m=video 51372 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The SIP element on the other side verifies the ticket then starts to process the
SDP.  If it does not support this feature then, as per the rules in RFC 3264, it
will reject the second m= line by using port 0 in the answer SDP (In this case
the end user will have to have PSTN only calls for the 48 hours after the
expiration of the ticket).  If the remote SIP element supports this extension
then it will send back an offer like this:

v=0
o=- 0 1 IN IP4 192.168.2.1
s=
c=IN IP4 192.168.2.1
t=0 0
m=audio 8000 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN E164 +14085551234
a=setup:passive
a=connection:new
a=cs-correlation:uuie:2890W284hAT452612908awudfjang908 dtmf:14D*3
m=video 9000 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The local SIP element checks that the c= line contains the right number, then
establishes the audio and video media over IP, as usual, but also starts a PSTN
call following the rules in draft-ietf-mmusic-sdp-cs (i.e. sending the
correlation value as needed).  The remote SIP element will receive the PSTN
call, and correlate it with the existing SIP connection.  Both parties must send
audio on both the IP and the PSTN side, but the receivers can choose to play the
audio coming from either the IP or PSTN connection (this is because we may want
to use in the future fingerprint methods for the validation).  The PSTN call is
ended at the same time than the IP connection if the methods used for validation
are based on the call duration (other methods may permit to end the call
before), so the VCRs are processed as for an initial call.  The local party
marks the route as been under re-validation, so to not use the renewal method
for the next 48 hours.  The VIPR server will send a notification before the end
of the 48 hours if there is still a PSTN route to this destination.

B.6. Infrastructure overlays

A new draft to discuss infrastructure overlays has been submitted to DISPATCH.

"[RELOAD] is a peer to peer protocol developed by the P2PSIP Working
 Group.  Each RELOAD instance has a unique name, which is used by the
 process in section 10.2 of this specification to find the
 configuration servers, enrollment servers and bootstrap servers
 needed to join the overlay.  The process assumes that the RELOAD
 instance name is a FQDN, and uses the process in [RFC2782] (SRV RR)
 to find the IP address of the HTTPS server that serves the
 configuration document for this overlay.

 This process is adequate when the management of the overlay does not
 need to be distinguished from the owner of the FQDN used as the
 instance name, which is the case most of the time.  But there is a
 special class of overlays that, by definition, requires to be unique
 on the Internet and for which having the possibility of create
 instances would defeat their very purpose.  This specification calls
 the kind of overlays that are not domain specific, but application
 specific "infrastructure overlays".

 [VIPR] is a technology that is being standardized in the VIPR Working
 Group and that aims to build bridges between SIP islands by
 automatically provision SIP routes after the "ownership" of a PSTN
 phone number has been verified by an actual PSTN phone call.  This
 technology uses an RELOAD overlay as a distributed database where
 mappings between phone numbers and servers responsible for the
 validation process are stored.  The promise of VIPR to bridge these
 SIP islands cannot be fulfilled if there is more than one distributed
 database storing these mappings."

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Thu Oct 13 08:41:56 2011
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One very important point made in draft-jennings-vipr-overview is the
troubleshooting problem (section 2.4).  Because the provisioning of SIP routes
is automatic, there is no possibility to have a interop testing between two
domains before the establishment of the fist call.  This means two things:

- - We will need in the future to define a SIP profile that defines exactly what
SIP extensions and media are supported (I propose to call this profile
SIPbeyond).  I think that we have a common goal here with the RTCWEB WG, and
should probably work with them on this.

- - We need to provide help for debugging problems if things go wrong.

For the latter, the main issue is that it is not always possible to troubleshoot
a problem only by looking on the logs and traces on the sending side - some kind
of cooperation is required by the receiving side.  Having a way to retrieve the
logs and traces automatically would be a good step in this direction, but we now
have a security problem to take care of, as we certainly do not want anybody to
be able to retrieve this information.

One internet-draft on this topic was just published, and it could be a start on
this discussion:

http://www.ietf.org/id/draft-kaithal-dispatch-sip-log-information-00.txt

There is certainly other possibilities, and this email is meant as a way to
start this discussion.

Thanks.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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On the last design call, I expressed a number of reservations about this ch=
ange, but the main one was that it was insufficiently motivated. For my par=
t, anyway, I was curious to understand what the problem was with the existi=
ng mechanism that you believed this solved - once we understood the problem=
, we could understand the potential design space to solve it. I had thought=
 we'd left the call understanding that you were going to explain the proble=
m; instead, you're pretty much defending a solution here without reference =
to what the problem is.=20

I've taken a look at the sketch of your proposal below, but it's a bit too =
high level for me to really understand its properties. I will say that if y=
our argument is:

> my proposal does not change what a normal SIP stack is doing,
> because the specification does not say what a SIP stack should do with a =
client
> certificate.

I have to disagree. 5922 cites the relevant text in 3261:
      Each side of the connection SHOULD verify and inspect the
      certificate of the other, noting the domain name that appears in
      the certificate for comparison with the header fields of SIP
      messages.
Now by today's standards, that text is somewhat vague, but back when we wro=
te that the idea was basically that you compare the domain in the cert (typ=
ically the CN or SubjectAltName) used in TLS with the supposed identity of =
the sender of a request over that transport connection (typically given in =
the From header field value). That works when the cert has been issued by a=
n authority capable of ascertaining who owns domain names. 5922 introduces =
qualifications to this in order to support cases where a virtual hosting se=
rvice has a certificate with multiple names, for example, but it isn't real=
ly chipping away at the basic operation. ViPR today, as you cite, requires =
that same reference integrity between the granted-to field of the ticket an=
d the domain in the cert used in TLS. Again, the idea here is that the gran=
ted-to has to correspond with a cert issued by an authority who actually ve=
rifies ownership of domain names. Thus, purely on the basis of capturing a =
ticket with a granted-to of example.com, you cannot impersonate example.com=
 unless you also control a cert issued for that domain by a CA, and inciden=
tally a CA trusted by the target ViPR domain. If you are arguing for a case=
 where a ViPR domain that is not in a position to certify who owns what dom=
ain names is issuing certificates that are used for TLS, then you are chang=
ing how certificate usage in SIP is commonly understood.

>From what I can gather about your proposal, I suspect you underestimate the=
 complexity of acting as a CA, enrolling peers in the CA and enabling them =
to use those credentials properly - what it means for two entities that hav=
e no pre-existing relationship to suddenly treat each other as trust anchor=
s. I furthermore suspect that TLS operations would not be optimal in this s=
ort of model. Is the antispam property something that is best bound to the =
transport or to a single SIP request? The use cases, ultimately, answer tha=
t for us: consider two large ViPR domains that want to exchange traffic - s=
hould they have to establish a new TLS connection for each called number, o=
r should they be able to send signaling for many calls over a single TLS co=
nnection? Are there cases where the hop-by-hop property of TLS weakens the =
utility of the mechanism? Are there environments we want ViPR to be useful =
where TLS is not available?=20

Jon Peterson (as an individual)
NeuStar, Inc.

On Oct 12, 2011, at 3:53 PM, Marc Petit-Huguenin wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> During the last conference call, this proposal was criticized as been a s=
ignificant change from the way SIP is processing TLS requests.  My research
> shows that it is not the case.
>=20
> First of all, let's define exactly what SIP, without even talking about V=
IPR, is
> doing regarding TLS.  The best place for this is RFC 5922.  We can skip t=
he
> client behavior discussion (section 7.3) because neither VIPR or my propo=
sal
> suggest any modification to this procedure.  Here's what RFC 5922 says ab=
out the
> server behavior (aka client certificate):
>=20
> "  When a server accepts a TLS connection, the server presents its own
>   X.509 certificate to the client.  Servers that wish to authenticate
>   the client will ask the client for a certificate.  If the client
>   possesses a certificate, that certificate is presented to the server.
>   If the client does not present a certificate, the client MUST NOT be
>   considered authenticated.
>=20
>   [...]
>=20
>   When authenticating the client, the server MUST obtain the set of SIP
>   domain identities from the client certificate as described in
>   Section 7.1.  Because the server accepted the TLS connection
>   passively, unlike a client, the server does not possess an AUS for
>   comparison.  Nonetheless, server policies can use the set of SIP
>   domain identities gathered from the certificate in Section 7.1 to
>   make authorization decisions.
>=20
>   For example, a very open policy could be to accept an X.509
>   certificate and validate the certificate using the procedures in RFC
>   5280 [6].  If the certificate is valid, the identity set is logged.
>=20
>   Alternatively, the server could have a list of all SIP domains the
>   server is allowed to accept connections from; when a client presents
>   its certificate, for each identity in the client certificate, the
>   server searches for the identity in the list of acceptable domains to
>   decide whether or not to accept the connection.  Other policies that
>   make finer distinctions are possible."
>=20
> In other words, the SIP server (or proxy) does whatever it thinks is nece=
ssary
> with the client certificate, and it does not seems that even following RF=
C 5280
> is required (note in particular that nowhere it is said that the validati=
on must
> use a public trust anchor).
>=20
> VIPR itself adds very little to the requirements of RFC 5922.  Here is wh=
at
> - -sip-antispam says about TLS and the ticket:
>=20
> "  The Border Element will receive the TLS ClientHello which begins the
>   TLS handshake.  The Border Element will present its own configured
>   cert.  Once TLS handshaking is complete, the Border Element notes the
>   domain from the SubjectAltName on the other side of the TLS
>   connection, and associates it with that connection.
>=20
>   Next, the Border Element will receive an INVITE.  This INVITE will
>   contain a ticket in the ViPR-Ticket header field value.  The Border
>   Element extracts this header field.  This call flow assumes it is
>   present.  The Border Element parses it, and obtains the epoch value
>   encoded in the ticket.  This is matched against the current epoch
>   value for the configured password.  If they match, processing
>   continues.  The Border Element verifies the signature is correct.
>   Next, it examines the start and stop time of the validity.  If the
>   current time is between the start and stop times, the check is
>   passed.  Next, the Border Element checks the granted-to domain in the
>   ticket.  It compares that domain against the domain name in the
>   SubjectAltName of the peer on the other side of the TLS connection,
>   as noted above.  Next, it takes the Request-URI of the SIP INVITE.
>   That will be of the form sip:+number@domain.  If it is not in that
>   form, and if the number does not begin with a plus, the request is
>   dropped.  The value, including the plus, is then compared to the
>   number in the ticket.  If it is equal, the check has passed.  The
>   Border Element leaves the header field in the request, but forwards
>   to the Call Agent."
>=20
> The only policy that is added to what RFC 5922 says is that the granted-t=
o
> domain in the ticket must be equal to the domain in the SubjectAltName of=
 the
> client certificate.  Here too, not a word about a public trust anchor.
>=20
> Now back to my proposal.  The first thing is that the client certificate =
must be
> validated using the CA of the VIPR server as trust anchor, which is alrea=
dy what
> a SIP server could do.  It validates according to the rules in RFC 5280, =
which
> is equivalent to verifying the signature of the ticket and the start/stop
> validity.  The SubjectAltName in the client certificate (which is not eve=
n
> required to be used in standard SIP) can be checked against the domain pa=
rt of
> the From header of the INVITE.  The phone number and the granting-domain =
(which
> not even verified in VIPR) can be stored as an URI in the Issuer Alternat=
ive
> Name extension in the client certificate and be validated against the
> Request-URI of the INVITE.  The granting node can be also be stored as a =
reload
> URI in an Issuer Alternative Name extension.
>=20
> So in conclusion, my proposal does not change what a normal SIP stack is =
doing,
> because the specification does not say what a SIP stack should do with a =
client
> certificate.
>=20
> (Note that to -sip-antispam should also be added some text saying that RF=
C 5923
> must be used, because there is no way to create an additional TLS connect=
ion in
> the other direction for a subsequent SIP transaction).
>=20
> On 09/03/2011 12:06 PM, Marc Petit-Huguenin wrote:
>> In the current specification, the PVP server builds and signs a VIPR tic=
ket, and
>> send it back to the PVP client, that passes it to the Call Agent/ Border=
 Element
>> (CA/BE).  The CA/BE will insert this ticket in the next SIP INVITE to th=
e remote
>> CA/BE, which will check the validity of the ticket and the domain of the=
 TLS client.
>>=20
>> There is multiple things that bother me with this process:
>>=20
>> - The verification of the ticket use HMAC-SHA1, and so the same password=
 has to
>> be provisioned in both the CA/BE(s) (to verify the ticket) and the VIPR
>> server(s) (to generate the ticket).
>> - The ticket can be used without TLS (we all know that implementers thin=
k of
>> MUST as SHOULD and SHOULD as MAY).
>>=20
>> I would like to propose an alternative that is simpler to implement and,=
 I
>> think, more secure than the ticket.
>>=20
>> Instead of using a symmetrical key, the local VIPR server generates an R=
SA key
>> pair, with the public key distributed to the CA/BE(s).  After a successf=
ul PVP
>> connection, the remote PVP client sends a PKCS#10 certificate request,
>> containing the domain name and signed with a private key[1].  The local =
PVP
>> server generates a new certificate[2], sign it with its private key and =
send it
>> back to the remote PVP client, which pass it back to the remote CA/BE.
>>=20
>> The remote CA/BE will use the certificate received to establish the TLS
>> connection for the SIP transactions to the local CA/BE, which will use a
>> standard CA based certificate for its domain.  The local CA/BE will be a=
ble to
>> verify with the VIPR public key that it issued the certificate, then wil=
l
>> validate the standard parts of the certificate (expiration date and so o=
n).
>>=20
>> There is multiple advantages to this proposal:
>>=20
>> - The private key is stored only in the VIPR server, or better in a phys=
ical
>> token.
>> - VIPR does not work without TLS for the SIP connection.
>> - There is plenty of commercial or free TLS and X.509 implementations th=
at have
>> years of experience and testing, which is not the case for the ticket pr=
oposal.
>> - Security agility is provided by TLS.
>> - No additional SIP Header.
>> - Because the ticket verification process is folded into the client cert=
ificate
>> verification, there is less resources needed.
>>=20
>>=20
>> [1] This is the private key of the CA/BE, so the CA/BE needs to upload t=
he
>> PKCS#10 file in the VIPR server.  Depending on the architecture used, ea=
ch CA/BE
>> can use a different PKCS#10 or the same, which means that the PVP server=
 must be
>> able to process more than one PKCS#10 and send back more than one certif=
icate on
>> one PVP connection.
>>=20
>> [2] The certificate sent back can directly contain the SIP routes, incre=
asing
>> the overall security.  One possible mapping from the ValExchange respons=
e to the
>> certificate could be:
>>=20
>> - Unique ID: serialNumber
>> - Epoch: issuer
>> - salt: N/A
>> - integrity: signature
>> - Validity: validity
>> - Phone Number: subjectAltName uniformResourceIdentifier tel:+1408555123=
4
>> - Node-ID: subjectAltName uniformResourceIdentifier
>> reload://0110123456789ABCDEF0123456789ABCDEF/
>> - Granted domain: subjectAltName dNSName example.com
>> - Granting domain: implicit in routes
>> - Route(s): subjectAltName uniformResourceIdentifier
>> sip:+14085551234@example.org:371;maddr=3D10.1.1.1
>>=20
>=20
>=20
> - --=20
> Marc Petit-Huguenin
> Personal email: marc@petit-huguenin.org
> Professional email: petithug@acm.org
> Blog: http://blog.marc.petit-huguenin.org
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.11 (GNU/Linux)
>=20
> iEYEARECAAYFAk6WGmoACgkQ9RoMZyVa61dW3gCeMcKcsvqK27kk2ZW1mRMPYdep
> 3ZcAn3EIFv8hlBZop57YIDN7HCOUtixU
> =3Dv31B
> -----END PGP SIGNATURE-----
> _______________________________________________
> VIPR mailing list
> VIPR@ietf.org
> https://www.ietf.org/mailman/listinfo/vipr


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Subject: Re: [VIPR] An alternative to the VIPR ticket
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On 10/13/2011 10:25 AM, Peterson, Jon wrote:
> 
> On the last design call, I expressed a number of reservations about this
> change, but the main one was that it was insufficiently motivated. For my
> part, anyway, I was curious to understand what the problem was with the
> existing mechanism that you believed this solved - once we understood the
> problem, we could understand the potential design space to solve it. I had
> thought we'd left the call understanding that you were going to explain the
> problem; instead, you're pretty much defending a solution here without
> reference to what the problem is.

I thought I did this, but here is my list of problems associated with the ticket
as currently defined:

- - Use of a symmetrical key, which has to be shared between different boxes if
the generation of the ticket is done in a different box than the verification of
the ticket.
- - No security agility.
- - Code has to be written from scratch for both the generation and the
verification of the ticket.

> 
> I've taken a look at the sketch of your proposal below, but it's a bit too
> high level for me to really understand its properties. I will say that if
> your argument is:
> 
>> my proposal does not change what a normal SIP stack is doing, because the
>> specification does not say what a SIP stack should do with a client 
>> certificate.
> 
> I have to disagree. 5922 cites the relevant text in 3261: Each side of the
> connection SHOULD verify and inspect the certificate of the other, noting the
> domain name that appears in the certificate for comparison with the header
> fields of SIP messages.  Now by today's standards, that text is somewhat
> vague, but back when we wrote that the idea was basically that you compare
> the domain in the cert (typically the CN or SubjectAltName) used in TLS with
> the supposed identity of the sender of a request over that transport
> connection (typically given in the From header field value). That works when
> the cert has been issued by an authority capable of ascertaining who owns
> domain names. 5922 introduces qualifications to this in order to support
> cases where a virtual hosting service has a certificate with multiple names,
> for example, but it isn't really chipping away at the basic operation. 

Let me offer a more complete quote of RFC 5922:

" RFC 3261 [2], Section 26.3.2.2, only tells Proxy-B to "compare the
  domain asserted by the certificate with the 'domainname' portion
  of the From header field in the INVITE request".  The difficulty
  with that instruction is that the domainname in the From header
  field is not always that of the domain from which the request is
  received."

So here RFC 5922 acknowledges that doing this verification is not possible in
all cases.  The minimum should have been to do an RFC 5280 validation, but even
that is not required.

> ViPR
> today, as you cite, requires that same reference integrity between the
> granted-to field of the ticket and the domain in the cert used in TLS. Again,
> the idea here is that the granted-to has to correspond with a cert issued by
> an authority who actually verifies ownership of domain names. Thus, purely on
> the basis of capturing a ticket with a granted-to of example.com, you cannot
> impersonate example.com unless you also control a cert issued for that domain
> by a CA, and incidentally a CA trusted by the target ViPR domain. If you are
> arguing for a case where a ViPR domain that is not in a position to certify
> who owns what domain names is issuing certificates that are used for TLS,
> then you are changing how certificate usage in SIP is commonly understood.

Yes, you have a point here.  Verifying the granted-to domain at call time will
prevent someone to use a stolen ticket, and will prevent the granted-to domain
to sell their tickets (although a domain can sell its private key also with the
tickets).

So if this verification is really that important (still thinking about this), I
guess that using the ticket as the client certificate is a problem.  But in any
case I still think that using an X.509 certificate in the Vipr-Ticket field is a
better solution than using an adhoc ticket as described currently in VIPR.

> 
>> From what I can gather about your proposal, I suspect you underestimate the
>> complexity of acting as a CA, enrolling peers in the CA and enabling them
>> to use those credentials properly 

I do not underestimate anything.  On the other hand the CAs certainly
overestimate the value of their services, especially on the light of the recent
problems some of them experienced.  Running a CA is not difficult, and the best
proof is that RELOAD (draft-ietf-p2psip-base section 10.3) requires running it's
own CA in a way that is similar to what I propose.

>> - what it means for two entities that
>> have no pre-existing relationship to suddenly treat each other as trust
>> anchors. I furthermore suspect that TLS operations would not be optimal in
>> this sort of model. Is the antispam property something that is best bound
>> to the transport or to a single SIP request? The use cases, ultimately,
>> answer that for us: consider two large ViPR domains that want to exchange
>> traffic - should they have to establish a new TLS connection for each
>> called number, or should they be able to send signaling for many calls over
>> a single TLS connection? Are there cases where the hop-by-hop property of
>> TLS weakens the utility of the mechanism? Are there environments we want
>> ViPR to be useful where TLS is not available?

Here you are not talking about my proposal, but about VIPR itself, as it already
requires TLS.  That's a different problem.

> 
> Jon Peterson (as an individual) NeuStar, Inc.
> 
> On Oct 12, 2011, at 3:53 PM, Marc Petit-Huguenin wrote:
> 
> During the last conference call, this proposal was criticized as been a
> significant change from the way SIP is processing TLS requests.  My research 
> shows that it is not the case.
> 
> First of all, let's define exactly what SIP, without even talking about VIPR,
> is doing regarding TLS.  The best place for this is RFC 5922.  We can skip
> the client behavior discussion (section 7.3) because neither VIPR or my
> proposal suggest any modification to this procedure.  Here's what RFC 5922
> says about the server behavior (aka client certificate):
> 
> "  When a server accepts a TLS connection, the server presents its own X.509
> certificate to the client.  Servers that wish to authenticate the client will
> ask the client for a certificate.  If the client possesses a certificate,
> that certificate is presented to the server. If the client does not present a
> certificate, the client MUST NOT be considered authenticated.
> 
> [...]
> 
> When authenticating the client, the server MUST obtain the set of SIP domain
> identities from the client certificate as described in Section 7.1.  Because
> the server accepted the TLS connection passively, unlike a client, the server
> does not possess an AUS for comparison.  Nonetheless, server policies can use
> the set of SIP domain identities gathered from the certificate in Section 7.1
> to make authorization decisions.
> 
> For example, a very open policy could be to accept an X.509 certificate and
> validate the certificate using the procedures in RFC 5280 [6].  If the
> certificate is valid, the identity set is logged.
> 
> Alternatively, the server could have a list of all SIP domains the server is
> allowed to accept connections from; when a client presents its certificate,
> for each identity in the client certificate, the server searches for the
> identity in the list of acceptable domains to decide whether or not to accept
> the connection.  Other policies that make finer distinctions are possible."
> 
> In other words, the SIP server (or proxy) does whatever it thinks is
> necessary with the client certificate, and it does not seems that even
> following RFC 5280 is required (note in particular that nowhere it is said
> that the validation must use a public trust anchor).
> 
> VIPR itself adds very little to the requirements of RFC 5922.  Here is what 
> -sip-antispam says about TLS and the ticket:
> 
> "  The Border Element will receive the TLS ClientHello which begins the TLS
> handshake.  The Border Element will present its own configured cert.  Once
> TLS handshaking is complete, the Border Element notes the domain from the
> SubjectAltName on the other side of the TLS connection, and associates it
> with that connection.
> 
> Next, the Border Element will receive an INVITE.  This INVITE will contain a
> ticket in the ViPR-Ticket header field value.  The Border Element extracts
> this header field.  This call flow assumes it is present.  The Border Element
> parses it, and obtains the epoch value encoded in the ticket.  This is
> matched against the current epoch value for the configured password.  If they
> match, processing continues.  The Border Element verifies the signature is
> correct. Next, it examines the start and stop time of the validity.  If the 
> current time is between the start and stop times, the check is passed.  Next,
> the Border Element checks the granted-to domain in the ticket.  It compares
> that domain against the domain name in the SubjectAltName of the peer on the
> other side of the TLS connection, as noted above.  Next, it takes the
> Request-URI of the SIP INVITE. That will be of the form sip:+number@domain.
> If it is not in that form, and if the number does not begin with a plus, the
> request is dropped.  The value, including the plus, is then compared to the 
> number in the ticket.  If it is equal, the check has passed.  The Border
> Element leaves the header field in the request, but forwards to the Call
> Agent."
> 
> The only policy that is added to what RFC 5922 says is that the granted-to 
> domain in the ticket must be equal to the domain in the SubjectAltName of
> the client certificate.  Here too, not a word about a public trust anchor.
> 
> Now back to my proposal.  The first thing is that the client certificate must
> be validated using the CA of the VIPR server as trust anchor, which is
> already what a SIP server could do.  It validates according to the rules in
> RFC 5280, which is equivalent to verifying the signature of the ticket and
> the start/stop validity.  The SubjectAltName in the client certificate (which
> is not even required to be used in standard SIP) can be checked against the
> domain part of the From header of the INVITE.  The phone number and the
> granting-domain (which not even verified in VIPR) can be stored as an URI in
> the Issuer Alternative Name extension in the client certificate and be
> validated against the Request-URI of the INVITE.  The granting node can be
> also be stored as a reload URI in an Issuer Alternative Name extension.
> 
> So in conclusion, my proposal does not change what a normal SIP stack is
> doing, because the specification does not say what a SIP stack should do with
> a client certificate.
> 
> (Note that to -sip-antispam should also be added some text saying that RFC
> 5923 must be used, because there is no way to create an additional TLS
> connection in the other direction for a subsequent SIP transaction).
> 
> On 09/03/2011 12:06 PM, Marc Petit-Huguenin wrote:
>>>> In the current specification, the PVP server builds and signs a VIPR
>>>> ticket, and send it back to the PVP client, that passes it to the Call
>>>> Agent/ Border Element (CA/BE).  The CA/BE will insert this ticket in
>>>> the next SIP INVITE to the remote CA/BE, which will check the validity
>>>> of the ticket and the domain of the TLS client.
>>>> 
>>>> There is multiple things that bother me with this process:
>>>> 
>>>> - The verification of the ticket use HMAC-SHA1, and so the same
>>>> password has to be provisioned in both the CA/BE(s) (to verify the
>>>> ticket) and the VIPR server(s) (to generate the ticket). - The ticket
>>>> can be used without TLS (we all know that implementers think of MUST as
>>>> SHOULD and SHOULD as MAY).
>>>> 
>>>> I would like to propose an alternative that is simpler to implement
>>>> and, I think, more secure than the ticket.
>>>> 
>>>> Instead of using a symmetrical key, the local VIPR server generates an
>>>> RSA key pair, with the public key distributed to the CA/BE(s).  After a
>>>> successful PVP connection, the remote PVP client sends a PKCS#10
>>>> certificate request, containing the domain name and signed with a
>>>> private key[1].  The local PVP server generates a new certificate[2],
>>>> sign it with its private key and send it back to the remote PVP client,
>>>> which pass it back to the remote CA/BE.
>>>> 
>>>> The remote CA/BE will use the certificate received to establish the
>>>> TLS connection for the SIP transactions to the local CA/BE, which will
>>>> use a standard CA based certificate for its domain.  The local CA/BE
>>>> will be able to verify with the VIPR public key that it issued the
>>>> certificate, then will validate the standard parts of the certificate
>>>> (expiration date and so on).
>>>> 
>>>> There is multiple advantages to this proposal:
>>>> 
>>>> - The private key is stored only in the VIPR server, or better in a
>>>> physical token. - VIPR does not work without TLS for the SIP
>>>> connection. - There is plenty of commercial or free TLS and X.509
>>>> implementations that have years of experience and testing, which is not
>>>> the case for the ticket proposal. - Security agility is provided by
>>>> TLS. - No additional SIP Header. - Because the ticket verification
>>>> process is folded into the client certificate verification, there is
>>>> less resources needed.
>>>> 
>>>> 
>>>> [1] This is the private key of the CA/BE, so the CA/BE needs to upload
>>>> the PKCS#10 file in the VIPR server.  Depending on the architecture
>>>> used, each CA/BE can use a different PKCS#10 or the same, which means
>>>> that the PVP server must be able to process more than one PKCS#10 and
>>>> send back more than one certificate on one PVP connection.
>>>> 
>>>> [2] The certificate sent back can directly contain the SIP routes,
>>>> increasing the overall security.  One possible mapping from the
>>>> ValExchange response to the certificate could be:
>>>> 
>>>> - Unique ID: serialNumber - Epoch: issuer - salt: N/A - integrity:
>>>> signature - Validity: validity - Phone Number: subjectAltName
>>>> uniformResourceIdentifier tel:+14085551234 - Node-ID: subjectAltName
>>>> uniformResourceIdentifier 
>>>> reload://0110123456789ABCDEF0123456789ABCDEF/ - Granted domain:
>>>> subjectAltName dNSName example.com - Granting domain: implicit in
>>>> routes - Route(s): subjectAltName uniformResourceIdentifier 
>>>> sip:+14085551234@example.org:371;maddr=10.1.1.1
>>>> 
> 
> 
_______________________________________________
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https://www.ietf.org/mailman/listinfo/vipr

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Marc Petit-Huguenin <petithug@acm.org>
Date: Thu, 13 Oct 2011 15:55:35 -0400
Thread-Topic: [VIPR] An alternative to the VIPR ticket
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Subject: Re: [VIPR] An alternative to the VIPR ticket
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I thought I did this, but here is my list of problems associated with the t=
icket
as currently defined:

- - Use of a symmetrical key, which has to be shared between different boxe=
s if
the generation of the ticket is done in a different box than the verificati=
on of
the ticket.

This was the only problem you brought up during the call. I would say howev=
er that if you're running a CA, then you need to share a great deal more th=
an a symmetrical key to validate signed messages. Unless the thing that iss=
ued the cert is colocated with the box that acts as a TLS server, you're no=
t going to be able to avoid some communication between those boxes.

As I said on the call, there are a number of ways to address that need for =
communication among distributed components that do not involve replicating =
secrets, and those are part of the design space for addressing implementati=
ons that want to distribute these components.

- - No security agility.

I assume you mean algorithm agility. TLS does have a great deal of latitude=
 here, it's true, including the ability to set a null cipher suite. You're =
going to end up having to narrowly profile TLS for the application if you w=
ant to achieve any measure of interoperability or security. SIP, for exampl=
e, is pretty specific about what algorithms you use with TLS. If you want t=
o change the recommended algorithm, you will still be iterating your specif=
ications and rolling out new versions.

- - Code has to be written from scratch for both the generation and the
verification of the ticket.

I'm dubious that any off-the-shelf CA library will enable you to put attrib=
utes like granted-by and so on into certificates without putting any signif=
icant burden on implementers, let alone parsing and verifying those attribu=
tes as a step in validating a TLS connection.


I've taken a look at the sketch of your proposal below, but it's a bit too
high level for me to really understand its properties. I will say that if
your argument is:

my proposal does not change what a normal SIP stack is doing, because the
specification does not say what a SIP stack should do with a client
certificate.

I have to disagree. 5922 cites the relevant text in 3261: Each side of the
connection SHOULD verify and inspect the certificate of the other, noting t=
he
domain name that appears in the certificate for comparison with the header
fields of SIP messages.  Now by today's standards, that text is somewhat
vague, but back when we wrote that the idea was basically that you compare
the domain in the cert (typically the CN or SubjectAltName) used in TLS wit=
h
the supposed identity of the sender of a request over that transport
connection (typically given in the From header field value). That works whe=
n
the cert has been issued by an authority capable of ascertaining who owns
domain names. 5922 introduces qualifications to this in order to support
cases where a virtual hosting service has a certificate with multiple names=
,
for example, but it isn't really chipping away at the basic operation.

Let me offer a more complete quote of RFC 5922:

" RFC 3261 [2], Section 26.3.2.2, only tells Proxy-B to "compare the
 domain asserted by the certificate with the 'domainname' portion
 of the From header field in the INVITE request".  The difficulty
 with that instruction is that the domainname in the From header
 field is not always that of the domain from which the request is
 received."

So here RFC 5922 acknowledges that doing this verification is not possible =
in
all cases.  The minimum should have been to do an RFC 5280 validation, but =
even
that is not required.

RFC3261 still naively assumed a SIP trapezoid. ViPR does as well, but much =
less naively. The notice given in here in RFC5922 is about cases where an i=
ntermediary domain prevents the establishment of a direct TLS connection. I=
f we have an intrusive intermediary in ViPR, then yes, you'll have a failur=
e establishing reference integrity. However, I'd say that the whole, entire=
 point of doing ViPR, of using RELOAD as a foundation, was to establish dir=
ect connections between ViPR domains without relying on any intermediary in=
frastructure. ViPR is by design a dis-intermediation mechanism. (This is in=
 danger of being the intermediary discussion again - hopefully this isn't o=
ne of the "hooks" that we're fishing for.)

Anyway, 5922 is speaking to a subset of SIP TLS behavior - behavior, incide=
ntally, that has been controversial in all of the protocols that rely on TL=
S, like virtual hosting cases and proxying cases. We should be using that a=
s a departure point to argue that reference integrity isn't a desirable or =
necessary property, especially not in ViPR. That is not the intention behin=
d RFC5922.


ViPR
today, as you cite, requires that same reference integrity between the
granted-to field of the ticket and the domain in the cert used in TLS. Agai=
n,
the idea here is that the granted-to has to correspond with a cert issued b=
y
an authority who actually verifies ownership of domain names. Thus, purely =
on
the basis of capturing a ticket with a granted-to of example.com<http://exa=
mple.com>, you cannot
impersonate example.com<http://example.com> unless you also control a cert =
issued for that domain
by a CA, and incidentally a CA trusted by the target ViPR domain. If you ar=
e
arguing for a case where a ViPR domain that is not in a position to certify
who owns what domain names is issuing certificates that are used for TLS,
then you are changing how certificate usage in SIP is commonly understood.

Yes, you have a point here.  Verifying the granted-to domain at call time w=
ill
prevent someone to use a stolen ticket, and will prevent the granted-to dom=
ain
to sell their tickets (although a domain can sell its private key also with=
 the
tickets).

So if this verification is really that important (still thinking about this=
), I
guess that using the ticket as the client certificate is a problem.  But in=
 any
case I still think that using an X.509 certificate in the Vipr-Ticket field=
 is a
better solution than using an adhoc ticket as described currently in VIPR.

Again, I think this would need to be far more thoroughly specified for us t=
o be able to evaluate the idea. I gather you mean that you could just put t=
he signature generated by the cert into the ticket's header field value. Th=
at is a much less radical proposal than doing this at the transport layer. =
We'd then need to evaluate, again, how those certs get created, installed a=
nd validated, and whether or not this truly solves your problem with distri=
buted components - or if indeed there are other ways entirely to approach t=
he distributed components problem that changing the way the ticket is secur=
ed, and which ways would require the least work.



>From what I can gather about your proposal, I suspect you underestimate the
complexity of acting as a CA, enrolling peers in the CA and enabling them
to use those credentials properly

I do not underestimate anything.  On the other hand the CAs certainly
overestimate the value of their services, especially on the light of the re=
cent
problems some of them experienced.  Running a CA is not difficult, and the =
best
proof is that RELOAD (draft-ietf-p2psip-base section 10.3) requires running=
 it's
own CA in a way that is similar to what I propose.


I rather think the recent CA problems are an illustration of how easy it is=
 to do things wrong. I look forward to reading your CPS.

Honestly, this would be a lot easier to swallow if you proposed just to gen=
erate a public/private key pair without all the X.509 baggage. I really don=
't see how a certificate adds any value, except in so far TLS requires them=
, but if you're willing to move this to a header field, X.509 would be the =
first thing I'd drop.

- what it means for two entities that
have no pre-existing relationship to suddenly treat each other as trust
anchors. I furthermore suspect that TLS operations would not be optimal in
this sort of model. Is the antispam property something that is best bound
to the transport or to a single SIP request? The use cases, ultimately,
answer that for us: consider two large ViPR domains that want to exchange
traffic - should they have to establish a new TLS connection for each
called number, or should they be able to send signaling for many calls over
a single TLS connection? Are there cases where the hop-by-hop property of
TLS weakens the utility of the mechanism? Are there environments we want
ViPR to be useful where TLS is not available?

Here you are not talking about my proposal, but about VIPR itself, as it al=
ready
requires TLS.  That's a different problem.


If ViPR Domain A nails up a TLS connection to ViPR Domain B, in the origina=
l ViPR plan as I understand it, the (public) cert presented by A in TLS is =
compared to the granted-to of tickets by B when A sends requests over that =
connection. A could send thousands of different requests, from thousands of=
 different callers, to thousands of different destinations, over the same T=
LS channel, because the channel is bound to the domain and not to any speci=
fic ticket. Unless I totally misunderstand your proposal, you are changing =
that, right? You want to instantiate the antispam function at the establish=
ment of a TLS connection. The issue is that in SIP, call signaling isn't ne=
cessary bound to the establishment of an underlying transport.

Agreed, however, that ViPR does basically require TLS today, so if TLS isn'=
t available, those parts of the spec as written today would't work, so that=
 isn't a problem with your proposal as such - though perhaps your (original=
) proposal would make adjusting the spec to fix that impossible.

Jon Peterson (as an individual)
NeuStar, Inc.


Jon Peterson (as an individual) NeuStar, Inc.

On Oct 12, 2011, at 3:53 PM, Marc Petit-Huguenin wrote:

During the last conference call, this proposal was criticized as been a
significant change from the way SIP is processing TLS requests.  My researc=
h
shows that it is not the case.

First of all, let's define exactly what SIP, without even talking about VIP=
R,
is doing regarding TLS.  The best place for this is RFC 5922.  We can skip
the client behavior discussion (section 7.3) because neither VIPR or my
proposal suggest any modification to this procedure.  Here's what RFC 5922
says about the server behavior (aka client certificate):

"  When a server accepts a TLS connection, the server presents its own X.50=
9
certificate to the client.  Servers that wish to authenticate the client wi=
ll
ask the client for a certificate.  If the client possesses a certificate,
that certificate is presented to the server. If the client does not present=
 a
certificate, the client MUST NOT be considered authenticated.

[...]

When authenticating the client, the server MUST obtain the set of SIP domai=
n
identities from the client certificate as described in Section 7.1.  Becaus=
e
the server accepted the TLS connection passively, unlike a client, the serv=
er
does not possess an AUS for comparison.  Nonetheless, server policies can u=
se
the set of SIP domain identities gathered from the certificate in Section 7=
.1
to make authorization decisions.

For example, a very open policy could be to accept an X.509 certificate and
validate the certificate using the procedures in RFC 5280 [6].  If the
certificate is valid, the identity set is logged.

Alternatively, the server could have a list of all SIP domains the server i=
s
allowed to accept connections from; when a client presents its certificate,
for each identity in the client certificate, the server searches for the
identity in the list of acceptable domains to decide whether or not to acce=
pt
the connection.  Other policies that make finer distinctions are possible."

In other words, the SIP server (or proxy) does whatever it thinks is
necessary with the client certificate, and it does not seems that even
following RFC 5280 is required (note in particular that nowhere it is said
that the validation must use a public trust anchor).

VIPR itself adds very little to the requirements of RFC 5922.  Here is what
-sip-antispam says about TLS and the ticket:

"  The Border Element will receive the TLS ClientHello which begins the TLS
handshake.  The Border Element will present its own configured cert.  Once
TLS handshaking is complete, the Border Element notes the domain from the
SubjectAltName on the other side of the TLS connection, and associates it
with that connection.

Next, the Border Element will receive an INVITE.  This INVITE will contain =
a
ticket in the ViPR-Ticket header field value.  The Border Element extracts
this header field.  This call flow assumes it is present.  The Border Eleme=
nt
parses it, and obtains the epoch value encoded in the ticket.  This is
matched against the current epoch value for the configured password.  If th=
ey
match, processing continues.  The Border Element verifies the signature is
correct. Next, it examines the start and stop time of the validity.  If the
current time is between the start and stop times, the check is passed.  Nex=
t,
the Border Element checks the granted-to domain in the ticket.  It compares
that domain against the domain name in the SubjectAltName of the peer on th=
e
other side of the TLS connection, as noted above.  Next, it takes the
Request-URI of the SIP INVITE. That will be of the form sip:+number@domain.
If it is not in that form, and if the number does not begin with a plus, th=
e
request is dropped.  The value, including the plus, is then compared to the
number in the ticket.  If it is equal, the check has passed.  The Border
Element leaves the header field in the request, but forwards to the Call
Agent."

The only policy that is added to what RFC 5922 says is that the granted-to
domain in the ticket must be equal to the domain in the SubjectAltName of
the client certificate.  Here too, not a word about a public trust anchor.

Now back to my proposal.  The first thing is that the client certificate mu=
st
be validated using the CA of the VIPR server as trust anchor, which is
already what a SIP server could do.  It validates according to the rules in
RFC 5280, which is equivalent to verifying the signature of the ticket and
the start/stop validity.  The SubjectAltName in the client certificate (whi=
ch
is not even required to be used in standard SIP) can be checked against the
domain part of the From header of the INVITE.  The phone number and the
granting-domain (which not even verified in VIPR) can be stored as an URI i=
n
the Issuer Alternative Name extension in the client certificate and be
validated against the Request-URI of the INVITE.  The granting node can be
also be stored as a reload URI in an Issuer Alternative Name extension.

So in conclusion, my proposal does not change what a normal SIP stack is
doing, because the specification does not say what a SIP stack should do wi=
th
a client certificate.

(Note that to -sip-antispam should also be added some text saying that RFC
5923 must be used, because there is no way to create an additional TLS
connection in the other direction for a subsequent SIP transaction).

On 09/03/2011 12:06 PM, Marc Petit-Huguenin wrote:
In the current specification, the PVP server builds and signs a VIPR
ticket, and send it back to the PVP client, that passes it to the Call
Agent/ Border Element (CA/BE).  The CA/BE will insert this ticket in
the next SIP INVITE to the remote CA/BE, which will check the validity
of the ticket and the domain of the TLS client.

There is multiple things that bother me with this process:

- The verification of the ticket use HMAC-SHA1, and so the same
password has to be provisioned in both the CA/BE(s) (to verify the
ticket) and the VIPR server(s) (to generate the ticket). - The ticket
can be used without TLS (we all know that implementers think of MUST as
SHOULD and SHOULD as MAY).

I would like to propose an alternative that is simpler to implement
and, I think, more secure than the ticket.

Instead of using a symmetrical key, the local VIPR server generates an
RSA key pair, with the public key distributed to the CA/BE(s).  After a
successful PVP connection, the remote PVP client sends a PKCS#10
certificate request, containing the domain name and signed with a
private key[1].  The local PVP server generates a new certificate[2],
sign it with its private key and send it back to the remote PVP client,
which pass it back to the remote CA/BE.

The remote CA/BE will use the certificate received to establish the
TLS connection for the SIP transactions to the local CA/BE, which will
use a standard CA based certificate for its domain.  The local CA/BE
will be able to verify with the VIPR public key that it issued the
certificate, then will validate the standard parts of the certificate
(expiration date and so on).

There is multiple advantages to this proposal:

- The private key is stored only in the VIPR server, or better in a
physical token. - VIPR does not work without TLS for the SIP
connection. - There is plenty of commercial or free TLS and X.509
implementations that have years of experience and testing, which is not
the case for the ticket proposal. - Security agility is provided by
TLS. - No additional SIP Header. - Because the ticket verification
process is folded into the client certificate verification, there is
less resources needed.


[1] This is the private key of the CA/BE, so the CA/BE needs to upload
the PKCS#10 file in the VIPR server.  Depending on the architecture
used, each CA/BE can use a different PKCS#10 or the same, which means
that the PVP server must be able to process more than one PKCS#10 and
send back more than one certificate on one PVP connection.

[2] The certificate sent back can directly contain the SIP routes,
increasing the overall security.  One possible mapping from the
ValExchange response to the certificate could be:

- Unique ID: serialNumber - Epoch: issuer - salt: N/A - integrity:
signature - Validity: validity - Phone Number: subjectAltName
uniformResourceIdentifier tel:+14085551234 - Node-ID: subjectAltName
uniformResourceIdentifier
reload://0110123456789ABCDEF0123456789ABCDEF/ - Granted domain:
subjectAltName dNSName example.com<http://example.com> - Granting domain: i=
mplicit in
routes - Route(s): subjectAltName uniformResourceIdentifier
sip:+14085551234@example.org:371;maddr=3D10.1.1.1



_______________________________________________
VIPR mailing list
VIPR@ietf.org<mailto:VIPR@ietf.org>
https://www.ietf.org/mailman/listinfo/vipr

- --
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode:=
 space; -webkit-line-break: after-white-space; "><div><br></div><div><block=
quote type=3D"cite"><div>I thought I did this, but here is my list of probl=
ems associated with the ticket<br>as currently defined:<br><br>- - Use of a=
 symmetrical key, which has to be shared between different boxes if<br>the =
generation of the ticket is done in a different box than the verification o=
f<br>the ticket.<br></div></blockquote><div><br></div><div>This was the onl=
y problem you brought up during the call. I would say however that if you'r=
e running a CA, then you need to share a great deal more than a symmetrical=
 key to validate signed messages. Unless the thing that issued the cert is =
colocated with the box that acts as a TLS server, you're not going to be ab=
le to avoid some communication between those boxes.</div><div><br></div><di=
v>As I said on the call, there are a number of ways to address that need fo=
r communication among distributed components that do not involve replicatin=
g secrets, and those are part of the design space for addressing implementa=
tions that want to distribute these components.</div><br><blockquote type=
=3D"cite"><div>- - No security agility.<br></div></blockquote><div><br></di=
v>I assume you mean algorithm agility. TLS does have a great deal of latitu=
de here, it's true, including the ability to set a null cipher suite. You'r=
e going to end up having to narrowly profile TLS for the application if you=
 want to achieve any measure of interoperability or security. SIP, for exam=
ple, is pretty specific about what algorithms you use with TLS. If you want=
 to change the recommended algorithm, you will still be iterating your spec=
ifications and rolling out new versions.<br><br><blockquote type=3D"cite"><=
div>- - Code has to be written from scratch for both the generation and the=
<br>verification of the ticket.<br></div></blockquote><div><br></div><div>I=
'm dubious that any off-the-shelf CA library will enable you to put attribu=
tes like granted-by and so on into certificates without putting any signifi=
cant burden on implementers, let alone parsing and verifying those attribut=
es as a step in validating a TLS connection.</div><br><blockquote type=3D"c=
ite"><div><blockquote type=3D"cite"><br></blockquote><blockquote type=3D"ci=
te">I've taken a look at the sketch of your proposal below, but it's a bit =
too<br></blockquote><blockquote type=3D"cite">high level for me to really u=
nderstand its properties. I will say that if<br></blockquote><blockquote ty=
pe=3D"cite">your argument is:<br></blockquote><blockquote type=3D"cite"><br=
></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">my propos=
al does not change what a normal SIP stack is doing, because the<br></block=
quote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">spec=
ification does not say what a SIP stack should do with a client<br></blockq=
uote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">certi=
ficate.<br></blockquote></blockquote><blockquote type=3D"cite"><br></blockq=
uote><blockquote type=3D"cite">I have to disagree. 5922 cites the relevant =
text in 3261: Each side of the<br></blockquote><blockquote type=3D"cite">co=
nnection SHOULD verify and inspect the certificate of the other, noting the=
<br></blockquote><blockquote type=3D"cite">domain name that appears in the =
certificate for comparison with the header<br></blockquote><blockquote type=
=3D"cite">fields of SIP messages. &nbsp;Now by today's standards, that text=
 is somewhat<br></blockquote><blockquote type=3D"cite">vague, but back when=
 we wrote that the idea was basically that you compare<br></blockquote><blo=
ckquote type=3D"cite">the domain in the cert (typically the CN or SubjectAl=
tName) used in TLS with<br></blockquote><blockquote type=3D"cite">the suppo=
sed identity of the sender of a request over that transport<br></blockquote=
><blockquote type=3D"cite">connection (typically given in the From header f=
ield value). That works when<br></blockquote><blockquote type=3D"cite">the =
cert has been issued by an authority capable of ascertaining who owns<br></=
blockquote><blockquote type=3D"cite">domain names. 5922 introduces qualific=
ations to this in order to support<br></blockquote><blockquote type=3D"cite=
">cases where a virtual hosting service has a certificate with multiple nam=
es,<br></blockquote><blockquote type=3D"cite">for example, but it isn't rea=
lly chipping away at the basic operation.<br></blockquote><br>Let me offer =
a more complete quote of RFC 5922:<br><br>" RFC 3261 [2], Section 26.3.2.2,=
 only tells Proxy-B to "compare the<br> &nbsp;domain asserted by the certif=
icate with the 'domainname' portion<br> &nbsp;of the From header field in t=
he INVITE request". &nbsp;The difficulty<br> &nbsp;with that instruction is=
 that the domainname in the From header<br> &nbsp;field is not always that =
of the domain from which the request is<br> &nbsp;received."<br><br>So here=
 RFC 5922 acknowledges that doing this verification is not possible in<br>a=
ll cases. &nbsp;The minimum should have been to do an RFC 5280 validation, =
but even<br>that is not required.<br></div></blockquote><div><br></div><div=
>RFC3261 still naively assumed a SIP trapezoid. ViPR does as well, but much=
 less naively. The notice given in here in RFC5922 is about cases where an =
intermediary domain prevents the establishment of a direct TLS connection. =
If we have an intrusive intermediary in ViPR, then yes, you'll have a failu=
re establishing reference integrity. However, I'd say that the whole, entir=
e point of doing ViPR, of using RELOAD as a foundation, was to establish di=
rect connections between ViPR domains without relying on any intermediary i=
nfrastructure. ViPR is by design a dis-intermediation mechanism. (This is i=
n danger of being the intermediary discussion again - hopefully this isn't =
one of the "hooks" that we're fishing for.)</div><div><br></div><div>Anyway=
, 5922 is speaking to a subset of SIP TLS behavior - behavior, incidentally=
, that has been controversial in all of the protocols that rely on TLS, lik=
e virtual hosting cases and proxying cases. We should be using that as a de=
parture point to argue that reference integrity isn't a desirable or necess=
ary property, especially not in ViPR. That is not the intention behind RFC5=
922.</div><br><blockquote type=3D"cite"><div><br><blockquote type=3D"cite">=
ViPR<br></blockquote><blockquote type=3D"cite">today, as you cite, requires=
 that same reference integrity between the<br></blockquote><blockquote type=
=3D"cite">granted-to field of the ticket and the domain in the cert used in=
 TLS. Again,<br></blockquote><blockquote type=3D"cite">the idea here is tha=
t the granted-to has to correspond with a cert issued by<br></blockquote><b=
lockquote type=3D"cite">an authority who actually verifies ownership of dom=
ain names. Thus, purely on<br></blockquote><blockquote type=3D"cite">the ba=
sis of capturing a ticket with a granted-to of <a href=3D"http://example.co=
m">example.com</a>, you cannot<br></blockquote><blockquote type=3D"cite">im=
personate <a href=3D"http://example.com">example.com</a> unless you also co=
ntrol a cert issued for that domain<br></blockquote><blockquote type=3D"cit=
e">by a CA, and incidentally a CA trusted by the target ViPR domain. If you=
 are<br></blockquote><blockquote type=3D"cite">arguing for a case where a V=
iPR domain that is not in a position to certify<br></blockquote><blockquote=
 type=3D"cite">who owns what domain names is issuing certificates that are =
used for TLS,<br></blockquote><blockquote type=3D"cite">then you are changi=
ng how certificate usage in SIP is commonly understood.<br></blockquote><br=
>Yes, you have a point here. &nbsp;Verifying the granted-to domain at call =
time will<br>prevent someone to use a stolen ticket, and will prevent the g=
ranted-to domain<br>to sell their tickets (although a domain can sell its p=
rivate key also with the<br>tickets).<br><br>So if this verification is rea=
lly that important (still thinking about this), I<br>guess that using the t=
icket as the client certificate is a problem. &nbsp;But in any<br>case I st=
ill think that using an X.509 certificate in the Vipr-Ticket field is a<br>=
better solution than using an adhoc ticket as described currently in VIPR.<=
br></div></blockquote><div><br></div><div>Again, I think this would need to=
 be far more thoroughly specified for us to be able to evaluate the idea. I=
 gather you mean that you could just put the signature generated by the cer=
t into the ticket's header field value. That is a much less radical proposa=
l than doing this at the transport layer. We'd then need to evaluate, again=
, how those certs get created, installed and validated, and whether or not =
this truly solves your problem with distributed components - or if indeed t=
here are other ways entirely to approach the distributed components problem=
 that changing the way the ticket is secured, and which ways would require =
the least work.</div><br><blockquote type=3D"cite"><div><br><blockquote typ=
e=3D"cite"><br></blockquote><blockquote type=3D"cite"><blockquote type=3D"c=
ite">From what I can gather about your proposal, I suspect you underestimat=
e the<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote ty=
pe=3D"cite">complexity of acting as a CA, enrolling peers in the CA and ena=
bling them<br></blockquote></blockquote><blockquote type=3D"cite"><blockquo=
te type=3D"cite">to use those credentials properly<br></blockquote></blockq=
uote><br>I do not underestimate anything. &nbsp;On the other hand the CAs c=
ertainly<br>overestimate the value of their services, especially on the lig=
ht of the recent<br>problems some of them experienced. &nbsp;Running a CA i=
s not difficult, and the best<br>proof is that RELOAD (draft-ietf-p2psip-ba=
se section 10.3) requires running it's<br>own CA in a way that is similar t=
o what I propose.<br><br></div></blockquote><div><br></div><div>I rather th=
ink the recent CA problems are an illustration of how easy it is to do thin=
gs wrong. I look forward to reading your CPS.</div><div><br></div><div>Hone=
stly, this would be a lot easier to swallow if you proposed just to generat=
e a public/private key pair without all the X.509 baggage. I really don't s=
ee how a certificate adds any value, except in so far TLS requires them, bu=
t if you're willing to move this to a header field, X.509 would be the firs=
t thing I'd drop.</div><br><blockquote type=3D"cite"><div><blockquote type=
=3D"cite"><blockquote type=3D"cite">- what it means for two entities that<b=
r></blockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"c=
ite">have no pre-existing relationship to suddenly treat each other as trus=
t<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote type=
=3D"cite">anchors. I furthermore suspect that TLS operations would not be o=
ptimal in<br></blockquote></blockquote><blockquote type=3D"cite"><blockquot=
e type=3D"cite">this sort of model. Is the antispam property something that=
 is best bound<br></blockquote></blockquote><blockquote type=3D"cite"><bloc=
kquote type=3D"cite">to the transport or to a single SIP request? The use c=
ases, ultimately,<br></blockquote></blockquote><blockquote type=3D"cite"><b=
lockquote type=3D"cite">answer that for us: consider two large ViPR domains=
 that want to exchange<br></blockquote></blockquote><blockquote type=3D"cit=
e"><blockquote type=3D"cite">traffic - should they have to establish a new =
TLS connection for each<br></blockquote></blockquote><blockquote type=3D"ci=
te"><blockquote type=3D"cite">called number, or should they be able to send=
 signaling for many calls over<br></blockquote></blockquote><blockquote typ=
e=3D"cite"><blockquote type=3D"cite">a single TLS connection? Are there cas=
es where the hop-by-hop property of<br></blockquote></blockquote><blockquot=
e type=3D"cite"><blockquote type=3D"cite">TLS weakens the utility of the me=
chanism? Are there environments we want<br></blockquote></blockquote><block=
quote type=3D"cite"><blockquote type=3D"cite">ViPR to be useful where TLS i=
s not available?<br></blockquote></blockquote><br>Here you are not talking =
about my proposal, but about VIPR itself, as it already<br>requires TLS. &n=
bsp;That's a different problem.<br><font class=3D"Apple-style-span" color=
=3D"#000000"><font class=3D"Apple-style-span" color=3D"#144FAE"><br></font>=
</font></div></blockquote><div><br></div><div>If ViPR Domain A nails up a T=
LS connection to ViPR Domain B, in the original ViPR plan as I understand i=
t, the (public) cert presented by A in TLS is compared to the granted-to of=
 tickets by B when A sends requests over that connection. A could send thou=
sands of different requests, from thousands of different callers, to thousa=
nds of different destinations, over the same TLS channel, because the chann=
el is bound to the domain and not to any specific ticket. Unless I totally =
misunderstand your proposal, you are changing that, right? You want to inst=
antiate the antispam function at the establishment of a TLS connection. The=
 issue is that in SIP, call signaling isn't necessary bound to the establis=
hment of an underlying transport.</div><div><br></div><div>Agreed, however,=
 that ViPR does basically require TLS today, so if TLS isn't available, tho=
se parts of the spec as written today would't work, so that isn't a problem=
 with your proposal as such - though perhaps your (original) proposal would=
 make adjusting the spec to fix that impossible.</div><div><br></div><div>J=
on Peterson (as an individual)</div><div>NeuStar, Inc.</div><br><blockquote=
 type=3D"cite"><div><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">Jon Peterson (as an individual) NeuStar, Inc.<br></blockquote=
><blockquote type=3D"cite"><br></blockquote><blockquote type=3D"cite">On Oc=
t 12, 2011, at 3:53 PM, Marc Petit-Huguenin wrote:<br></blockquote><blockqu=
ote type=3D"cite"><br></blockquote><blockquote type=3D"cite">During the las=
t conference call, this proposal was criticized as been a<br></blockquote><=
blockquote type=3D"cite">significant change from the way SIP is processing =
TLS requests. &nbsp;My research<br></blockquote><blockquote type=3D"cite">s=
hows that it is not the case.<br></blockquote><blockquote type=3D"cite"><br=
></blockquote><blockquote type=3D"cite">First of all, let's define exactly =
what SIP, without even talking about VIPR,<br></blockquote><blockquote type=
=3D"cite">is doing regarding TLS. &nbsp;The best place for this is RFC 5922=
. &nbsp;We can skip<br></blockquote><blockquote type=3D"cite">the client be=
havior discussion (section 7.3) because neither VIPR or my<br></blockquote>=
<blockquote type=3D"cite">proposal suggest any modification to this procedu=
re. &nbsp;Here's what RFC 5922<br></blockquote><blockquote type=3D"cite">sa=
ys about the server behavior (aka client certificate):<br></blockquote><blo=
ckquote type=3D"cite"><br></blockquote><blockquote type=3D"cite">" &nbsp;Wh=
en a server accepts a TLS connection, the server presents its own X.509<br>=
</blockquote><blockquote type=3D"cite">certificate to the client. &nbsp;Ser=
vers that wish to authenticate the client will<br></blockquote><blockquote =
type=3D"cite">ask the client for a certificate. &nbsp;If the client possess=
es a certificate,<br></blockquote><blockquote type=3D"cite">that certificat=
e is presented to the server. If the client does not present a<br></blockqu=
ote><blockquote type=3D"cite">certificate, the client MUST NOT be considere=
d authenticated.<br></blockquote><blockquote type=3D"cite"><br></blockquote=
><blockquote type=3D"cite">[...]<br></blockquote><blockquote type=3D"cite">=
<br></blockquote><blockquote type=3D"cite">When authenticating the client, =
the server MUST obtain the set of SIP domain<br></blockquote><blockquote ty=
pe=3D"cite">identities from the client certificate as described in Section =
7.1. &nbsp;Because<br></blockquote><blockquote type=3D"cite">the server acc=
epted the TLS connection passively, unlike a client, the server<br></blockq=
uote><blockquote type=3D"cite">does not possess an AUS for comparison. &nbs=
p;Nonetheless, server policies can use<br></blockquote><blockquote type=3D"=
cite">the set of SIP domain identities gathered from the certificate in Sec=
tion 7.1<br></blockquote><blockquote type=3D"cite">to make authorization de=
cisions.<br></blockquote><blockquote type=3D"cite"><br></blockquote><blockq=
uote type=3D"cite">For example, a very open policy could be to accept an X.=
509 certificate and<br></blockquote><blockquote type=3D"cite">validate the =
certificate using the procedures in RFC 5280 [6]. &nbsp;If the<br></blockqu=
ote><blockquote type=3D"cite">certificate is valid, the identity set is log=
ged.<br></blockquote><blockquote type=3D"cite"><br></blockquote><blockquote=
 type=3D"cite">Alternatively, the server could have a list of all SIP domai=
ns the server is<br></blockquote><blockquote type=3D"cite">allowed to accep=
t connections from; when a client presents its certificate,<br></blockquote=
><blockquote type=3D"cite">for each identity in the client certificate, the=
 server searches for the<br></blockquote><blockquote type=3D"cite">identity=
 in the list of acceptable domains to decide whether or not to accept<br></=
blockquote><blockquote type=3D"cite">the connection. &nbsp;Other policies t=
hat make finer distinctions are possible."<br></blockquote><blockquote type=
=3D"cite"><br></blockquote><blockquote type=3D"cite">In other words, the SI=
P server (or proxy) does whatever it thinks is<br></blockquote><blockquote =
type=3D"cite">necessary with the client certificate, and it does not seems =
that even<br></blockquote><blockquote type=3D"cite">following RFC 5280 is r=
equired (note in particular that nowhere it is said<br></blockquote><blockq=
uote type=3D"cite">that the validation must use a public trust anchor).<br>=
</blockquote><blockquote type=3D"cite"><br></blockquote><blockquote type=3D=
"cite">VIPR itself adds very little to the requirements of RFC 5922. &nbsp;=
Here is what<br></blockquote><blockquote type=3D"cite">-sip-antispam says a=
bout TLS and the ticket:<br></blockquote><blockquote type=3D"cite"><br></bl=
ockquote><blockquote type=3D"cite">" &nbsp;The Border Element will receive =
the TLS ClientHello which begins the TLS<br></blockquote><blockquote type=
=3D"cite">handshake. &nbsp;The Border Element will present its own configur=
ed cert. &nbsp;Once<br></blockquote><blockquote type=3D"cite">TLS handshaki=
ng is complete, the Border Element notes the domain from the<br></blockquot=
e><blockquote type=3D"cite">SubjectAltName on the other side of the TLS con=
nection, and associates it<br></blockquote><blockquote type=3D"cite">with t=
hat connection.<br></blockquote><blockquote type=3D"cite"><br></blockquote>=
<blockquote type=3D"cite">Next, the Border Element will receive an INVITE. =
&nbsp;This INVITE will contain a<br></blockquote><blockquote type=3D"cite">=
ticket in the ViPR-Ticket header field value. &nbsp;The Border Element extr=
acts<br></blockquote><blockquote type=3D"cite">this header field. &nbsp;Thi=
s call flow assumes it is present. &nbsp;The Border Element<br></blockquote=
><blockquote type=3D"cite">parses it, and obtains the epoch value encoded i=
n the ticket. &nbsp;This is<br></blockquote><blockquote type=3D"cite">match=
ed against the current epoch value for the configured password. &nbsp;If th=
ey<br></blockquote><blockquote type=3D"cite">match, processing continues. &=
nbsp;The Border Element verifies the signature is<br></blockquote><blockquo=
te type=3D"cite">correct. Next, it examines the start and stop time of the =
validity. &nbsp;If the<br></blockquote><blockquote type=3D"cite">current ti=
me is between the start and stop times, the check is passed. &nbsp;Next,<br=
></blockquote><blockquote type=3D"cite">the Border Element checks the grant=
ed-to domain in the ticket. &nbsp;It compares<br></blockquote><blockquote t=
ype=3D"cite">that domain against the domain name in the SubjectAltName of t=
he peer on the<br></blockquote><blockquote type=3D"cite">other side of the =
TLS connection, as noted above. &nbsp;Next, it takes the<br></blockquote><b=
lockquote type=3D"cite">Request-URI of the SIP INVITE. That will be of the =
form <a href=3D"sip:+number@domain.">sip:+number@domain.</a><br></blockquot=
e><blockquote type=3D"cite">If it is not in that form, and if the number do=
es not begin with a plus, the<br></blockquote><blockquote type=3D"cite">req=
uest is dropped. &nbsp;The value, including the plus, is then compared to t=
he<br></blockquote><blockquote type=3D"cite">number in the ticket. &nbsp;If=
 it is equal, the check has passed. &nbsp;The Border<br></blockquote><block=
quote type=3D"cite">Element leaves the header field in the request, but for=
wards to the Call<br></blockquote><blockquote type=3D"cite">Agent."<br></bl=
ockquote><blockquote type=3D"cite"><br></blockquote><blockquote type=3D"cit=
e">The only policy that is added to what RFC 5922 says is that the granted-=
to<br></blockquote><blockquote type=3D"cite">domain in the ticket must be e=
qual to the domain in the SubjectAltName of<br></blockquote><blockquote typ=
e=3D"cite">the client certificate. &nbsp;Here too, not a word about a publi=
c trust anchor.<br></blockquote><blockquote type=3D"cite"><br></blockquote>=
<blockquote type=3D"cite">Now back to my proposal. &nbsp;The first thing is=
 that the client certificate must<br></blockquote><blockquote type=3D"cite"=
>be validated using the CA of the VIPR server as trust anchor, which is<br>=
</blockquote><blockquote type=3D"cite">already what a SIP server could do. =
&nbsp;It validates according to the rules in<br></blockquote><blockquote ty=
pe=3D"cite">RFC 5280, which is equivalent to verifying the signature of the=
 ticket and<br></blockquote><blockquote type=3D"cite">the start/stop validi=
ty. &nbsp;The SubjectAltName in the client certificate (which<br></blockquo=
te><blockquote type=3D"cite">is not even required to be used in standard SI=
P) can be checked against the<br></blockquote><blockquote type=3D"cite">dom=
ain part of the From header of the INVITE. &nbsp;The phone number and the<b=
r></blockquote><blockquote type=3D"cite">granting-domain (which not even ve=
rified in VIPR) can be stored as an URI in<br></blockquote><blockquote type=
=3D"cite">the Issuer Alternative Name extension in the client certificate a=
nd be<br></blockquote><blockquote type=3D"cite">validated against the Reque=
st-URI of the INVITE. &nbsp;The granting node can be<br></blockquote><block=
quote type=3D"cite">also be stored as a reload URI in an Issuer Alternative=
 Name extension.<br></blockquote><blockquote type=3D"cite"><br></blockquote=
><blockquote type=3D"cite">So in conclusion, my proposal does not change wh=
at a normal SIP stack is<br></blockquote><blockquote type=3D"cite">doing, b=
ecause the specification does not say what a SIP stack should do with<br></=
blockquote><blockquote type=3D"cite">a client certificate.<br></blockquote>=
<blockquote type=3D"cite"><br></blockquote><blockquote type=3D"cite">(Note =
that to -sip-antispam should also be added some text saying that RFC<br></b=
lockquote><blockquote type=3D"cite">5923 must be used, because there is no =
way to create an additional TLS<br></blockquote><blockquote type=3D"cite">c=
onnection in the other direction for a subsequent SIP transaction).<br></bl=
ockquote><blockquote type=3D"cite"><br></blockquote><blockquote type=3D"cit=
e">On 09/03/2011 12:06 PM, Marc Petit-Huguenin wrote:<br></blockquote><bloc=
kquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><b=
lockquote type=3D"cite">In the current specification, the PVP server builds=
 and signs a VIPR<br></blockquote></blockquote></blockquote></blockquote><b=
lockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"=
><blockquote type=3D"cite">ticket, and send it back to the PVP client, that=
 passes it to the Call<br></blockquote></blockquote></blockquote></blockquo=
te><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"=
cite"><blockquote type=3D"cite">Agent/ Border Element (CA/BE). &nbsp;The CA=
/BE will insert this ticket in<br></blockquote></blockquote></blockquote></=
blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote =
type=3D"cite"><blockquote type=3D"cite">the next SIP INVITE to the remote C=
A/BE, which will check the validity<br></blockquote></blockquote></blockquo=
te></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockq=
uote type=3D"cite"><blockquote type=3D"cite">of the ticket and the domain o=
f the TLS client.<br></blockquote></blockquote></blockquote></blockquote><b=
lockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"=
><blockquote type=3D"cite"><br></blockquote></blockquote></blockquote></blo=
ckquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote typ=
e=3D"cite"><blockquote type=3D"cite">There is multiple things that bother m=
e with this process:<br></blockquote></blockquote></blockquote></blockquote=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te"><blockquote type=3D"cite"><br></blockquote></blockquote></blockquote></=
blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote =
type=3D"cite"><blockquote type=3D"cite">- The verification of the ticket us=
e HMAC-SHA1, and so the same<br></blockquote></blockquote></blockquote></bl=
ockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote ty=
pe=3D"cite"><blockquote type=3D"cite">password has to be provisioned in bot=
h the CA/BE(s) (to verify the<br></blockquote></blockquote></blockquote></b=
lockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote t=
ype=3D"cite"><blockquote type=3D"cite">ticket) and the VIPR server(s) (to g=
enerate the ticket). - The ticket<br></blockquote></blockquote></blockquote=
></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquo=
te type=3D"cite"><blockquote type=3D"cite">can be used without TLS (we all =
know that implementers think of MUST as<br></blockquote></blockquote></bloc=
kquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite"><blockquote type=3D"cite">SHOULD and SHOULD as MAY).=
<br></blockquote></blockquote></blockquote></blockquote><blockquote type=3D=
"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite"><br></blockquote></blockquote></blockquote></blockquote><blockquo=
te type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><block=
quote type=3D"cite">I would like to propose an alternative that is simpler =
to implement<br></blockquote></blockquote></blockquote></blockquote><blockq=
uote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blo=
ckquote type=3D"cite">and, I think, more secure than the ticket.<br></block=
quote></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><=
br></blockquote></blockquote></blockquote></blockquote><blockquote type=3D"=
cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite">Instead of using a symmetrical key, the local VIPR server generat=
es an<br></blockquote></blockquote></blockquote></blockquote><blockquote ty=
pe=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote=
 type=3D"cite">RSA key pair, with the public key distributed to the CA/BE(s=
). &nbsp;After a<br></blockquote></blockquote></blockquote></blockquote><bl=
ockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">=
<blockquote type=3D"cite">successful PVP connection, the remote PVP client =
sends a PKCS#10<br></blockquote></blockquote></blockquote></blockquote><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><=
blockquote type=3D"cite">certificate request, containing the domain name an=
d signed with a<br></blockquote></blockquote></blockquote></blockquote><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><=
blockquote type=3D"cite">private key[1]. &nbsp;The local PVP server generat=
es a new certificate[2],<br></blockquote></blockquote></blockquote></blockq=
uote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite"><blockquote type=3D"cite">sign it with its private key and send i=
t back to the remote PVP client,<br></blockquote></blockquote></blockquote>=
</blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquot=
e type=3D"cite"><blockquote type=3D"cite">which pass it back to the remote =
CA/BE.<br></blockquote></blockquote></blockquote></blockquote><blockquote t=
ype=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquot=
e type=3D"cite"><br></blockquote></blockquote></blockquote></blockquote><bl=
ockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">=
<blockquote type=3D"cite">The remote CA/BE will use the certificate receive=
d to establish the<br></blockquote></blockquote></blockquote></blockquote><=
blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite=
"><blockquote type=3D"cite">TLS connection for the SIP transactions to the =
local CA/BE, which will<br></blockquote></blockquote></blockquote></blockqu=
ote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D=
"cite"><blockquote type=3D"cite">use a standard CA based certificate for it=
s domain. &nbsp;The local CA/BE<br></blockquote></blockquote></blockquote><=
/blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote=
 type=3D"cite"><blockquote type=3D"cite">will be able to verify with the VI=
PR public key that it issued the<br></blockquote></blockquote></blockquote>=
</blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquot=
e type=3D"cite"><blockquote type=3D"cite">certificate, then will validate t=
he standard parts of the certificate<br></blockquote></blockquote></blockqu=
ote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><block=
quote type=3D"cite"><blockquote type=3D"cite">(expiration date and so on).<=
br></blockquote></blockquote></blockquote></blockquote><blockquote type=3D"=
cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite"><br></blockquote></blockquote></blockquote></blockquote><blockquo=
te type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><block=
quote type=3D"cite">There is multiple advantages to this proposal:<br></blo=
ckquote></blockquote></blockquote></blockquote><blockquote type=3D"cite"><b=
lockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"=
><br></blockquote></blockquote></blockquote></blockquote><blockquote type=
=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote t=
ype=3D"cite">- The private key is stored only in the VIPR server, or better=
 in a<br></blockquote></blockquote></blockquote></blockquote><blockquote ty=
pe=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote=
 type=3D"cite">physical token. - VIPR does not work without TLS for the SIP=
<br></blockquote></blockquote></blockquote></blockquote><blockquote type=3D=
"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite">connection. - There is plenty of commercial or free TLS and X.509=
<br></blockquote></blockquote></blockquote></blockquote><blockquote type=3D=
"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=
=3D"cite">implementations that have years of experience and testing, which =
is not<br></blockquote></blockquote></blockquote></blockquote><blockquote t=
ype=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquot=
e type=3D"cite">the case for the ticket proposal. - Security agility is pro=
vided by<br></blockquote></blockquote></blockquote></blockquote><blockquote=
 type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockqu=
ote type=3D"cite">TLS. - No additional SIP Header. - Because the ticket ver=
ification<br></blockquote></blockquote></blockquote></blockquote><blockquot=
e type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockq=
uote type=3D"cite">process is folded into the client certificate verificati=
on, there is<br></blockquote></blockquote></blockquote></blockquote><blockq=
uote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blo=
ckquote type=3D"cite">less resources needed.<br></blockquote></blockquote><=
/blockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite=
"><blockquote type=3D"cite"><blockquote type=3D"cite"><br></blockquote></bl=
ockquote></blockquote></blockquote><blockquote type=3D"cite"><blockquote ty=
pe=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><br></block=
quote></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">[=
1] This is the private key of the CA/BE, so the CA/BE needs to upload<br></=
blockquote></blockquote></blockquote></blockquote><blockquote type=3D"cite"=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te">the PKCS#10 file in the VIPR server. &nbsp;Depending on the architectur=
e<br></blockquote></blockquote></blockquote></blockquote><blockquote type=
=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote t=
ype=3D"cite">used, each CA/BE can use a different PKCS#10 or the same, whic=
h means<br></blockquote></blockquote></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquo=
te type=3D"cite">that the PVP server must be able to process more than one =
PKCS#10 and<br></blockquote></blockquote></blockquote></blockquote><blockqu=
ote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><bloc=
kquote type=3D"cite">send back more than one certificate on one PVP connect=
ion.<br></blockquote></blockquote></blockquote></blockquote><blockquote typ=
e=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote =
type=3D"cite"><br></blockquote></blockquote></blockquote></blockquote><bloc=
kquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><b=
lockquote type=3D"cite">[2] The certificate sent back can directly contain =
the SIP routes,<br></blockquote></blockquote></blockquote></blockquote><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><=
blockquote type=3D"cite">increasing the overall security. &nbsp;One possibl=
e mapping from the<br></blockquote></blockquote></blockquote></blockquote><=
blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite=
"><blockquote type=3D"cite">ValExchange response to the certificate could b=
e:<br></blockquote></blockquote></blockquote></blockquote><blockquote type=
=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote t=
ype=3D"cite"><br></blockquote></blockquote></blockquote></blockquote><block=
quote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite">- Unique ID: serialNumber - Epoch: issuer - salt: N/=
A - integrity:<br></blockquote></blockquote></blockquote></blockquote><bloc=
kquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><b=
lockquote type=3D"cite">signature - Validity: validity - Phone Number: subj=
ectAltName<br></blockquote></blockquote></blockquote></blockquote><blockquo=
te type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><block=
quote type=3D"cite">uniformResourceIdentifier tel:+14085551234 - Node-ID: s=
ubjectAltName<br></blockquote></blockquote></blockquote></blockquote><block=
quote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite">uniformResourceIdentifier<br></blockquote></blockquo=
te></blockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"=
cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><a href=3D"reload=
://0110123456789ABCDEF0123456789ABCDEF/">reload://0110123456789ABCDEF012345=
6789ABCDEF/</a> - Granted domain:<br></blockquote></blockquote></blockquote=
></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquo=
te type=3D"cite"><blockquote type=3D"cite">subjectAltName dNSName <a href=
=3D"http://example.com">example.com</a> - Granting domain: implicit in<br><=
/blockquote></blockquote></blockquote></blockquote><blockquote type=3D"cite=
"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"c=
ite">routes - Route(s): subjectAltName uniformResourceIdentifier<br></block=
quote></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blo=
ckquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><=
a href=3D"sip:+14085551234@example.org:371;maddr=3D10.1.1.1">sip:+140855512=
34@example.org:371;maddr=3D10.1.1.1</a><br></blockquote></blockquote></bloc=
kquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite"><blockquote type=3D"cite"><br></blockquote></blockqu=
ote></blockquote></blockquote><blockquote type=3D"cite"><br></blockquote><b=
lockquote type=3D"cite"><br></blockquote>__________________________________=
_____________<br>VIPR mailing list<br><a href=3D"mailto:VIPR@ietf.org">VIPR=
@ietf.org</a><br>https://www.ietf.org/mailman/listinfo/vipr<br><br>- --<br>=
Marc Petit-Huguenin<br>Personal email: marc@petit-huguenin.org<br>Professio=
nal email: petithug@acm.org<br>Blog: http://blog.marc.petit-huguenin.org<br=
>-----BEGIN PGP SIGNATURE-----<br>Version: GnuPG v1.4.11 (GNU/Linux)<br><br=
>iEYEARECAAYFAk6XMLUACgkQ9RoMZyVa61dhCgCghMV7hIJ8Gcc1GZjGEiDflw44<br>kwkAn3=
zHPotOdEQRb4VQort1a8aYb/f4<br>=3DSlq9<br>-----END PGP SIGNATURE-----<br></d=
iv></blockquote></div><br></body></html>=

--_000_5143068F634B4578AA6D4EF902471B30neustarbiz_--

From dean.willis@softarmor.com  Thu Oct 13 15:23:25 2011
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Subject: Re: [VIPR] An alternative to the VIPR ticket
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On 10/13/11 12:25 PM, Peterson, Jon wrote:

> I furthermore suspect
> that TLS operations would not be optimal in this sort of model. Is
> the antispam property something that is best bound to the transport
> or to a single SIP request? The use cases, ultimately, answer that
> for us: consider two large ViPR domains that want to exchange traffic
> - should they have to establish a new TLS connection for each called
> number, or should they be able to send signaling for many calls over
> a single TLS connection? Are there cases where the hop-by-hop
> property of TLS weakens the utility of the mechanism? Are there
> environments we want ViPR to be useful where TLS is not available?

Perhaps the ViPR domain intermediaries shouldn't be cryptographically 
involved in TLS at all.

See: http://tools.ietf.org/html/draft-gurbani-sip-sipsec-01


--
Dean

From petithug@acm.org  Mon Oct 17 08:46:17 2011
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

VIPR Design Team conference call on Friday October 14th, from 9:05am to 9:50am PST.

o Attendees:
  Daryl Malas
  Hakim Mehmood
  Muthu Arul Mozhi Perumal
  Jon Peterson
  Marc Petit-Huguenin
  Michael Procter
  Dean Willis

o Regrets:
  Mary Barnes
  John Ward

o Agenda

- - Replacement ticket by client certificate (10 min)
- - PVP method priority and registration (10 min)
- - PVP privacy (10 min)
- - PVP method entropy (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (5 min)

o Action items

- - Add template for IANA registration for PVP methods, with priority.
- - Work on new method inverting the selector with the secret.
- - Send an email in the list detailing the entropy solutions (client-side vs
server-side).

o Minutes

- - Replacement ticket by client certificate

Marc: The goal of this meeting is to discuss items that are not discussed in the
mailing-list, and as there is good discussions on the mailing-list about this
item, I propose to skip it for this conference call.

- - PVP method priority and registration

Marc: There is currently two PVP methods defined in the draft: method "a' which
use the caller and callee number, and method "b" which uses the callee number
and the time in the middle of the call.  But there is more methods that can be
used (fingerprint, DTMF, SMS, etc...).  The first part of the proposal is to
establish a IANA registry for new methods and to assign a priority to each
method.  The second part of the proposal is to advertize the methods supported
by the PVP server in the RELOAD database, to accelerate the validation process.
 The third part of the proposal is to move the PVP algorithm into -framework and
to keep only the description of the methods in -pvp.

Hakim: So the proposal is to put the priority of the methods for PVP.

Marc: I was proposing to put only the list of methods supported.  I am not sure
it is interesting to put also the priority in the RELOAD database.

Hakim: OK, its just a list of methods supported for this particular resource.

Marc: Yes, and this should not be an issue for security.

Jon: It would matter if one method was weaker than another and a domain was
claiming it prefers it.

Marc: That's a good reason to not put the priority in the RELOAD database.

Jon: From an administrative point view, you do not have to move out the current
methods to provide a template.  Also how an implementation knows if Caller-ID is
going to work or not?  That maybe specific to the method, so can you really say
at the registry level that a method will work or not?

Marc: If a domain will never ever receive the Caller-ID, then it should not
advertize method "a".  But if there is a chance that method "a" succeed, then it
should advertize.

Jon: So anything that could work, you will ask to try it.

Marc: Yes, but the advantage is that methods we are absolutely sure that will
not work, like SMS on a non mobile connection, can be excluded.  If you have no
access to the audio stream, then a method using fingerprint can also be excluded.

Jon: OK, there is language that can capture that.

Marc: And what about priority in the IANA registry?

Hakim: That's fine.

Jon: The more methods we have, the more an attacker will be able to choose the
weakest.  So I think that what it will come down to what we think our
requirements a method has to meet to validate.  Until we understand that, it
will be tough to say that all these methods are secure.

Marc: In the framework document, I will list exactly what should be in the
document defining a method.

- - PVP privacy

The problem, reported by Michael Procter, is that the Caller-ID is send in clear
in method "a".  I proposed three additional definitions, to make the discussion
on this topic easier.  The PVP selector, which is sent in clear, is used to
select the call record on the server side. In method "a" the selector is the
called number and the Caller-ID, in method "b", it is the called number and the
time in the middle of the call.  The PVP parameters are a set of parameters that
are also sent in clear, for example the rounding value for "a" and "b".  The PVP
secret is what is verified, and cannot be discovered if the verification fails.
 For the two methods, it is the beginning and end of the audio call. The privacy
problem here is that someone can register a phone number and knows who is
calling this phone number.  There was a discussion on the mailing between
Michael and Cullen Jennings, where the idea was to hash the phone numbers. But
perhaps that instead of fixing methods "a" and "b" we should just create new
methods.  So a first new method is to use the same than "a", but hashing with
something like bcrypt.  A second new method would be to invert the selector and
the secret.

Hakim: If I understand correctly, the problem here is that someone can learn who
calls.

Marc: Yes

Jon: Hashing is a speed bump, but I suspect not worth enough to do it.  The idea
of inverting the secret with the selector is interesting.  I agree that the call
start and stop is not sufficiently unique and it has the property that the more
people are using VIPR the more likely there will be collision.  But it can be
combined with part of the numbers, i.e. area code or area code and prefixes that
are longer.

Hakim: Method "a" is more secure than method "b", but method "b" is not exposing
the Caller-ID at all, so domains that have concerns over privacy simply do not
advertize method "a".

Marc: The problem is that method "b" probably does not have enough entropy.

Hakim: Yes, it does circumvent to some extend the security concerns.

Marc: So do we want to fix the two methods, or keep them as they are and define
new methods?

Hakim: Invent new methods.

Jon: The problem is that if we invent new things that are less secure, then
people will use them.

Hakim: I do not disagree with that.

Marc: True, but "b" is already not very secure.  We cannot make something less
secure than "b", and we can simply say that we cannot have a method less secure
than "b".

Jon: Let's figure how secure "b" is first.  If we find that "b" is unacceptable,
then we may not want to do that.

Michael: We need to understand what the tradeoff are between have enough entropy
for a reliable call identification compared to leaking too much information.

Marc: I am personally on working on new methods and put explanations on the
security section of the current methods saying, do not use this.

- - PVP method entropy

Marc: Method "b" provides less entropy than method "a", which itself provides
less entropy than the future methods.  The idea is to accumulate entropy, so a
domain can require more than one validation before sending back a route and a
ticket.  Instead of sending back a route and ticket, the server sends back a
temporary ticket that contains the entropy accumulated, which is sent during the
next attempt at validation.  It fits well with what we talked before, priority,
and the fact that some methods have less security than other.  We can put in the
IANA registry the entropy that a specific method would accumulate.

Jon: I do not understand the way entropy is calculated but someone should figure
out if that actually works.

Marc: But is it a direction we want to go to?

Jon: The goal of VIPR is to reduce as much as possible the number of calls to
the PSTN, so something that would require more of those is undesirable.  We need
to evaluate what we get in return.  Is there alternative that accomplish the
same thing?  We can use probe calls during off hours.  If we decide to do this,
we really need to decide what the goals are.

Marc: The goal is to make method "b" better than what it is, because it is will
the best that some domains will be able to do.  I not would like to exclude
method "b" because it would exclude a lot of domains from VIPR.

Jon: Do we have a sense of how many people are excluded by this today?

Marc: This is all the cases where we do not have the Caller-ID.

Jon: How many things that we think will participate in VIPR have PRI?

Hakim: The problem is not necessarily not having access to Caller-ID, but how
enterprise domains send Caller-IDs that are the Caller-Id of the enterprise
versus the calling entity.

Jon: But for VIPR purpose it still works.

Hakim: No it does not because it would create an issue when you match the ticket.

Jon: What I am asking is about the 80/20 rule.  Does it happen 20% of the time,
1% of the time or 0.1% of the time?

Hakim: It's in the 20%.

Dean: The majority of PBXs send their trunk number, so we are talking about how
many PBXs versus SOHO devices.

Jon: If it's 20% we need to do something, if it's 1% we should do something, if
it's less than 1%, I do not care.

Hakim: For sure it's not 1%, but somewhere close.

Michael: For people that are making international calls, like videocalls to
grandma, then using the CLI will fail more often than it succeeds.

Daryl: Good point.

Michael: If you are talking of calls between enterprises, then yes Caller-Id
will work most of the time.

Jon: I am not really sure that this is for grandmas, because things for grandmas
do not have this property that VIPR has.  But point taken about the
international call issues.

Marc: One thing I would like to add is that we are not reducing the user
experience, because the 3 calls are at the beginning.  So doing this is
acceptable.  What is unacceptable would be to switch back to PSTN after having
video.

Jon: I do think that it is a problem.  It's not unacceptable, but it's
undesirable. It's not just video, it's wideband audio, it's presence, and it's
our job here to find the quickest way to enable this.  And if the way is to have
3 PSTN calls, it's definitively undesirable.

Hakim: I agree.  It's the #1 complaint I see today.  That it takes 24 hours, all
these things.

Marc:  I agree that we need to design methods that does this faster. But we
still need to have a bottom line which is method "b" because this is how we will
have many people in the overlay.  We should not require an SS7 connection to
participate in VIPR.

Marc: We need to decide if we want to talk about entropy or not at all.

Hakim: Is it possible to keep the guidance here, and let the implementers decide?

Marc: We still need a way to manage the entropy.  It can be a PVP server side
thing, where the server keeps a log of all the calls.  Or it can be done on the
PVP client side, in which case we have to define what is exchanged.

Hakim:  My preference would be server-side.

Marc: My preference is client side.

Jon: We need to think more about what we need to accomplish.

Dean: Post the question on the mailing-list.

Marc:  OK, I'll do that.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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Subject: [VIPR] Using only method "b" for privacy
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During the conference call last Friday, we had a discussion on which the
conclusion was that privacy conscious VIPR domains could simply require the use
of method "b" for validation, as this method does not send the Caller-ID in the
PVP login, and so does not disclose the phone number of the callers to VIPR
domains that registers the phone number of one of their competitors.

Unfortunately, I do not think that doing this would work.  Here's a diagram
explaining that:

DomainA      RELOAD   DomainB     DomainC
   |            |   Store|           |
1  |            |<-------|     Store |
2  |            |<-------------------|
   :            :        :           :
   | SETUP      |        |           |
3  |~~~~~~~~~~~~~~~~~~~~~|           |
   |            |        |           |
4  |<~~~~~~~~~~~~~~~~~~~~|           |
   :            :        :           :
   | Fetch      |        |           |
5  |----------->|        |           |
   | PVP        |        |           |
6  |::::::::::::::::::::::::::::::::X|
   | PVP        |        |           |
7  |::::::::::::::::::::>|           |
   |            |        |           |

In 1, the legitimate "owner" of the phone number registers a mapping between
this number and its PVP server in the distributed RELOAD database.  In 2, a
competitor registers its own PVP server for the same phone number.  In 3 and 4
we have the PSTN call itself, that generates the originating VCR in domain A and
the terminating VCR in domain B.  Shortly after this, 5 shows that DomainA
retrieved the mappings for the called phone number, and so retrieved the address
of the PVP servers from DomainB and DomainC, without knowing which one is the
"owner".  It starts a PVP transaction with both of them, with the transaction to
DomainC failing because DomainC does not know the PVP secret, and the one with
DomainB succeeding.

Now even if DomainB is privacy conscious and refuses to process PVP transactions
using methods "a", that does not prevent DomainC to accept transactions using
methods "a", and so permitting it to learn the caller number.

Even using the registration mechanism, where the Store also contains the list of
methods supported by a specific endpoint, does not fix this problem.

The problem here is that DomainA needs to be privacy conscious on behalf of the
remote domains, and never use method "a".

One simple solution is to never even define method "a", so it can never be used.
 That is a change from what was discussed during the conference call, which was
to keep method "a" and "b" as they are, and to define better methods.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Tue Oct 18 08:49:39 2011
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Subject: Re: [VIPR] PVP entropy
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The discussion on the subject of PVP entropy on the last conference call did not
reach consensus.  The -pvp draft already contains a placeholder for discussing
entropy (section 10.1), so we need to do something about this.  The various
proposals were:

1. We do not need to manage entropy because we do not want to have the user
making multiple PSTN calls before receiving a SIP route.
2. We need to manage entropy, but it is best managed completely in the PVP server.
3. We need to manage entropy, but it is best managed in both client and server
by using something like a cookie.

My proposal is 3, as detailed below.  I think that 1 would exclude too many
participants in VIPR, so I oppose it.

On 09/04/2011 01:53 PM, Marc Petit-Huguenin wrote:
> Section 10.1 of the -pvp draft is an empty section that is supposed to describe
> how to aggregate entropy.  The idea is that the fact that one validation
> succeeds could not be enough for a remote VIPR server to give back routes and
> ticket for this destination.  For example a site could decide that 3 different
> calls validated with method "b" are needed to let a SIP call reach the endpoint.
> 
> This proposal describes a mechanism for PVP to implement this.  The idea is that
> when a PVP validation succeeds, the PVP server will return a ticket[1] but may
> not return the list of routes if the entropy threshold is not reached.  The
> ticket will contain an opaque value set by the PVP server that contain the level
> of entropy that this caller has reached.  The PVP client stores the ticket but
> does not notify the call agent as there is no routes available.  The next time
> the PVP client has a successful validation with the same destination, it sends
> the saved ticket in addition to the domain.  The PVP server then evaluates the
> ticket, increases the entropy value stored and send back a new ticket back.  If
> the threshold has been reached, then the PVP server sends back the routes at the
> same time, routes that the PVP client can now send with the ticket to the call
> agent.
> 
> When renewing the routes/ticket, the PVP client also sends the existing ticket,
> so the PVP server can decide to lower the threshold based on the entropy
> collected the previous time and the date of the ticket.
> 
> This can be combined with my proposal for method priority.  If multiple methods
> are available for a destination but the first that succeeds does not return the
> routes then the PVP client can use the next available methods in the list to try
> to increase the entropy and receive them.
> 
> A PVP server may even use different thresholds, depending on the domains to
> validate.  This become then an extension to the white/black list, where a
> blacklist is implemented as a default threshold of X and a infinite threshold
> for the domains blacklisted, and where a whitelist is implemented as a default
> infinite threshold and a threshold of X for the domains whitelisted.
> 
> 
> [1] Or a certificate if my other proposal is accepted.
> 

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Tue Oct 18 12:46:35 2011
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Subject: [VIPR] Hashing the Caller-ID to increase privacy in PVP
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During the last conference call, the proposal for using hash to protect the
privacy of the caller in VIPR was summarily dismissed.  I would like to give it
another shot.

The idea is that when using method "a", the Caller-ID is hashed using the bcrypt
hash algorithm[1], and the resulting hash is passed in the login of the SRP-TLS
transaction.  On the PVP server side the code retrieves the list of all calls
made to the called number (which is in clear), extracts the salt from the hash,
rehashes the Caller-ID of the calls selected, and selects the most recent call
that have a Caller-ID that matches the hash received.

Now a VIPR domain that would want to now the Caller-ID of the callers of a
specific phone number will have to hash a good part of the E.164 space to have a
chance to find the Caller-ID.  If the target is a 1800 number, so the attacker
can restrict the search space to the +1 prefix, it will need to hash in average
half this space, which is something like 1.3 billion numbers.  The beauty of the
bcrypt algorithm is that the number of rekeying rounds is configurable, and so
it can be increased as the hardware becomes faster.  For example on my computer,
it takes 1.3 ms to hash a US number when using 2^4 rounds, but 100 ms when using
2^10 rounds.  That means that it will take 4 years to hash half of the USA phone
numbers on one processor (adding more processors decreases the hash time
linearly).  The number of rounds could be a parameter of the VIPR overlay.

Here's what the login could look like:

a:vs=7f5a8630b6365bf2;op=$2a$10$uhNBlMT5O063n5/YMlg3Y.xTmZuHMeAFbSxhwSacX87aujFxPvoxG;tp=+16504614667;r=1000;

Exercise for the skeptical:  Make a phone call to the originating phone number
in this example...


[1] http://www.usenix.org/event/usenix99/provos/provos.pdf

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
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Been sick since yesterday, I am not able to work on the agenda and handout for
tomorrow's meeting.

So unless someone want to take over, I propose to cancel this call, and to
resume the normal conference call next week.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From michael@voip.co.uk  Fri Oct 21 10:58:04 2011
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From: Michael Procter <michael@voip.co.uk>
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Subject: [VIPR] Privacy concerns
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I've just submitted a draft[0] which summarises most of the privacy
concerns that have been raised in recent times, along with the
suggested mitigations from the list.  This is clearly not exhaustive,
but I was losing track of which concerns we had addressed and which
ones were still on the table.  I thought a draft would be a good place
to keep track of these issues for the time being.

Comments, suggestions, text happily accepted.

Regards,

Michael

[0] https://datatracker.ietf.org/doc/draft-procter-vipr-privacy-concerns/

From petithug@acm.org  Sat Oct 22 10:52:28 2011
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Hi Michael,

On 10/21/2011 10:57 AM, Michael Procter wrote:
> I've just submitted a draft[0] which summarises most of the privacy
> concerns that have been raised in recent times, along with the
> suggested mitigations from the list.  This is clearly not exhaustive,
> but I was losing track of which concerns we had addressed and which
> ones were still on the table.  I thought a draft would be a good place
> to keep track of these issues for the time being.
> 
> Comments, suggestions, text happily accepted.
> 
> Regards,
> 
> Michael
> 
> [0] https://datatracker.ietf.org/doc/draft-procter-vipr-privacy-concerns/

in Section 1.1.1:

Note that I refined and tested this proposal since I wrote the email that is the
source for this section.  My new proposal[1] is to use bcrypt for the Caller-ID
alone, so rehashing on the PVP server side is limited to the calls made to the
terminating phone number in the last 48 hours.  The salt and the number of
rounds is part of the hash string, so the only configuration parameter would be
to have the minimal number of rounds under which a PVP server refuses a hash.

The main problem is that the phone numbers that are the most interesting to
attack are also the one that generates the most calls per 48 hours interval
(e.g. 1-800 numbers).  This means that the number of rounds must be balanced
between the desire to made the hash practically unbreakable, and the time to
rehash the phone numbers for the last 48 hours.  Again we are in a situation
where a VIPR domain registering a phone number may want that PVP clients use
different algorithms depending on the phone number.  So instead of having only
the minimal number of rounds in the configuration file, we can have both the
minimal number of rounds and the default number of rounds in the configuration,
and let the VIPR domain change the default value for each registered number
(down to the minimal value) inside the RELOAD registration.

I used a 4 cores computer to crack a phone number hashed with 2^4 rounds, and it
took 106 minutes (410 minutes of CPU) to crack it, *knowing* the area code.
There is ~300 area codes in the USA, so if all area codes were tried in
parallel, it would take 22 days.  And it is for one phone call, and with only
2^4 rounds.   With 2^10 rounds, it would take 6 years to crack this phone number
with the same hardware.

The bcrypt paper claims that "its inner loop relies exclusively on operations
that are efficient on general-purpose CPUs, leaving little opportunity for
specialized hardware to achieve dramatic improvements", so using GPU would not
help much.  But if we start with the smaller cost that one's would not pay for a
phone number, then evaluate the CPU power this cost would buy on Amazon EC2[2],
and then calculate the number of rounds that would use this CPU power, then we
could assign a default minimal number of rounds in the global VIPR configuration
file, and increase it as the cost of Amazon EC2 decreases.


[1] http://www.ietf.org/mail-archive/web/vipr/current/msg00088.html
[2]
http://stacksmashing.net/2010/11/15/cracking-in-the-cloud-amazons-new-ec2-gpu-instances/

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Sun Oct 23 09:32:49 2011
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As instructed by our AD[1] after the meeting in Quebec City, I moved this draft
back to the VIPR WG.

The main change in this version is that the quota does not apply to replicas
(this change was made after discussion with the developers of the only known
deployed implementation of this algorithm).  The reason is that because there
can be huge difference in the fraction of the overlay owned by each peer,
entries are likely to be lost during replication.  Because the storing peer of
the replication verifies that the source of the replication is valid, this
should not create the possibility to store more entries than permitted by the
quota mechanism (see section 6.4.1.1 of draft-ietf-p2psip-base).

Changelog:

o  The quota does not apply to replicas.
o  Draft moved to VIPR WG.
o  Nits.


[1] http://www.ietf.org/mail-archive/web/vipr/current/msg00040.html

- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-vipr-proportional-quota-00.txt
Date: Sun, 23 Oct 2011 09:12:05 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

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

	Title           : Proportional Quota in REsource LOcation And Discovery (RELOAD)
	Author(s)       : Marc Petit-Huguenin
                          Jonathan Rosenberg
                          Cullen Jennings
	Filename        : draft-petithuguenin-vipr-proportional-quota-00.txt
	Pages           : 7
	Date            : 2011-10-23

   This document defines an extension to RELOAD [I-D.ietf-p2psip-base]
   that limits the number of a specific kind element that can be stored
   by one RELOAD peer.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-vipr-proportional-quota-00.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-petithuguenin-vipr-proportional-quota-00.txt

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A. Agenda

- - Replacement ticket by client certificate (10 min)
- - PVP privacy (10 min)
- - PVP method entropy (10 min)
- - Ticket renewal (10 min)
- - Infrastructure overlays (10 min)
- - VIPR troubleshooting (10 min)

Note that the sum of the time allocated to each topic may exceeds the time
allocated to the whole call.  The topics that cannot make it to the discussion
will be postponed to the next conference call if they are not subsequently
decided in the mailing-list.

Because the conference bridge as a limited capacity, an individual invitation to
the conference will be sent shortly after this email to each member of the
design team.  Please send me an email if you are not on this list and want to
receive an invitation.

Note that the design team conference call is not a way to short-circuit the
normal process, but a way to accelerate it by increasing face to face time.  All
decisions made must be confirmed on the mailing-list.

This conference call is subject to the rules of RFC 5378 and RFC 3979 (updated
by RFC 4879).


B. Handout (reading time: 10 min)

B.1 Replacement ticket by client certificate

The discussion did not progress since the previous call. Jon position is that
there is ways to have distributed components that does not require replicating
secrets, that TLS brings algorithm agility but we will still have to define a
profile, and that using X.509 certificates does not add any value.

My position is that an X.509 certificate with symmetrical keys is better than an
adhoc cryptographic ticket.

B.2. PVP Privacy

Michael Procter published an Internet-Draft summarizing all the privacy issues.
 Two things were discussed on the mailing-list:

First that restricting a domain to a particular list of methods does not
prevents other domain to use less private methods.  They need to be fixed or
being removed.

Second I proposed to hash the Caller-ID in method "a" in a way that make it
difficult to retrieve, by using bcrypt.

B.3. PVP Entropy

The discussion at the end of the previous conference call stalled on what to do
about entropy.  There is 3 possibilities:

1. We do not need to manage entropy because we do not want to have the user
making multiple PSTN calls before receiving a SIP route.
2. We need to manage entropy, but it is best managed completely in the PVP server.
3. We need to manage entropy, but it is best managed in both client and server
by using something like a cookie.

B.4. Ticket renewal

Another thing that was discussed in Quebec City was the "First Call Problem",
i.e. the problem that it takes up to 48 hours to verify a call and been able to
use enhanced media.  I think the conclusion of the discussion was that it was
not too annoying for the first call, as it did not degrade the user experience.
On the other hand, the cryptographic ticket granted after the first has an
expiration date and so need to be renewed, and that will degrade the user
experience as it would mean that for each destination, the end-user will
periodically not be able to use enhanced media for the duration of a call.

There was multiple proposals at the microphone to solve this problem, but I
would like to detail the one I proposed.

The idea is, sometimes before the expiration of the ticket, to make in parallel
a PSTN call and a SIP call using the existing SIP route and ticket for the
enhanced media (let's say video for the remaining of this discussion).  There is
already an existing I-D that can be use for this, draft-ietf-mmusic-sdp-cs.  The
way it could work is that the call agent, after retrieving the SIP route and
ticket for a destination, will decide that it is time to re-validate the route,
based for example on the frequency of calls to this number and the remaining
validity of the ticket.  The SIP element then adds an additional m= line to the
SDP that contains a PSTN address.  The offered SDP then looks something like this:

v=0
o=- 2890844526 2890842807 IN IP4 10.47.16.5
s=
c=IN IP4 10.47.16.5
t=0 0
m=audio 49170 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN - -
a=setup:active
a=connection:new
a=cs-correlation: uuie callerid dtmf
m=video 51372 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The SIP element on the other side verifies the ticket then starts to process the
SDP.  If it does not support this feature then, as per the rules in RFC 3264, it
will reject the second m= line by using port 0 in the answer SDP (In this case
the end user will have to have PSTN only calls for the 48 hours after the
expiration of the ticket).  If the remote SIP element supports this extension
then it will send back an offer like this:

v=0
o=- 0 1 IN IP4 192.168.2.1
s=
c=IN IP4 192.168.2.1
t=0 0
m=audio 8000 RTP/AVP 0
m=audio 9 PSTN -
c=PSTN E164 +14085551234
a=setup:passive
a=connection:new
a=cs-correlation:uuie:2890W284hAT452612908awudfjang908 dtmf:14D*3
m=video 9000 RTP/AVP 99
a=rtpmap:99 h263-1998/90000

The local SIP element checks that the c= line contains the right number, then
establishes the audio and video media over IP, as usual, but also starts a PSTN
call following the rules in draft-ietf-mmusic-sdp-cs (i.e. sending the
correlation value as needed).  The remote SIP element will receive the PSTN
call, and correlate it with the existing SIP connection.  Both parties must send
audio on both the IP and the PSTN side, but the receivers can choose to play the
audio coming from either the IP or PSTN connection (this is because we may want
to use in the future fingerprint methods for the validation).  The PSTN call is
ended at the same time than the IP connection if the methods used for validation
are based on the call duration (other methods may permit to end the call
before), so the VCRs are processed as for an initial call.  The local party
marks the route as been under re-validation, so to not use the renewal method
for the next 48 hours.  The VIPR server will send a notification before the end
of the 48 hours if there is still a PSTN route to this destination.

B.5. Infrastructure overlays

A new draft to discuss infrastructure overlays has been submitted to DISPATCH.

"[RELOAD] is a peer to peer protocol developed by the P2PSIP Working
 Group.  Each RELOAD instance has a unique name, which is used by the
 process in section 10.2 of this specification to find the
 configuration servers, enrollment servers and bootstrap servers
 needed to join the overlay.  The process assumes that the RELOAD
 instance name is a FQDN, and uses the process in [RFC2782] (SRV RR)
 to find the IP address of the HTTPS server that serves the
 configuration document for this overlay.

 This process is adequate when the management of the overlay does not
 need to be distinguished from the owner of the FQDN used as the
 instance name, which is the case most of the time.  But there is a
 special class of overlays that, by definition, requires to be unique
 on the Internet and for which having the possibility of create
 instances would defeat their very purpose.  This specification calls
 the kind of overlays that are not domain specific, but application
 specific "infrastructure overlays".

 [VIPR] is a technology that is being standardized in the VIPR Working
 Group and that aims to build bridges between SIP islands by
 automatically provision SIP routes after the "ownership" of a PSTN
 phone number has been verified by an actual PSTN phone call.  This
 technology uses an RELOAD overlay as a distributed database where
 mappings between phone numbers and servers responsible for the
 validation process are stored.  The promise of VIPR to bridge these
 SIP islands cannot be fulfilled if there is more than one distributed
 database storing these mappings."

B.6. VIPR troubleshooting

One very important point made in draft-jennings-vipr-overview is the
troubleshooting problem (section 2.4).  Because the provisioning of SIP routes
is automatic, there is no possibility to have a interop testing between two
domains before the establishment of the fist call.  This means two things:

- - We will need in the future to define a SIP profile that defines exactly what
SIP extensions and media are supported (I propose to call this profile
SIPbeyond).  I think that we have a common goal here with the RTCWEB WG, and
should probably work with them on this.

- - We need to provide help for debugging problems if things go wrong.

For the latter, the main issue is that it is not always possible to troubleshoot
a problem only by looking on the logs and traces on the sending side - some kind
of cooperation is required by the receiving side.  Having a way to retrieve the
logs and traces automatically would be a good step in this direction, but we now
have a security problem to take care of, as we certainly do not want anybody to
be able to retrieve this information.


Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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Section 1.2.1 of draft-procter-vipr-privacy-concerns proposes to use TURN to
anonymise the source of a PVP validation.  This proposal is probably incomplete,
but the main problem is that we still do not know what transports will be used
on the RELOAD overlay used by VIPR.  First here's a list of what we currently
know about this overlay:

- - The algorithm is RELOAD-CHORD
- - The Node-ID size is 16 bytes (128 bits).
- - Clients are not authorized because of the quota algorithm.
- - Self-signed certificates are not authorized.
- - There will be one or more root-certs.
- - There will be one or more bootstrap peers (unicast or anycast - no multicast).
- - Clients are not authorized because of the quota algorithm.
- - Shared secret are not used.
- - The overlay link protocol is TLS.
- - There will be a list of kind and configuration signers.
- - The urn:ietf:params:xml:ns:p2p:config-chord, urn:ietf:params:xml:ns:p2p:quota
and urn:ietf:params:xml:ns:p2p:vipr extensions are mandatory.
- - The Kind VIPR-REGISTRATION will be authorized.

What we do not know yet:

- - Do we need to authorize the CERTIFICATE-BY-NODE and CERTIFICATE-BY-USER Kinds?
- - Will TURN server deployed in the overlay?
- - Will ICE overlay link protocols be used?

The important question for this topic is the support of ICE in the RELOAD overlay:

Pros:  With ICE support the RELOAD nodes, the PVP client and the PVP server can
be installed behind a NAT/Firewall without having to put them on the DMZ or to
open pinholes (the bootstrap servers will still need to be installed in the DMZ
or on the Internet).  TURN servers are required for ICE support, so they would
also be available for anonymising the connections.

Cons:  ICE is complex to implement and debug.

If we decide to exclude ICE, then the TURN servers are no longer available by
default, and the idea of using them for anonymising the connections is moot.

An additional consideration is that the RELOAD transports are currently limited
to UDP and TCP, neither of them being really well suited to carry RELOAD or PVP
messages.  If we decide to exclude ICE support, I will then propose to use
(D)TLS over SCTP instead (and write the I-D for this).

So the initial question is:  Do we need ICE or not?

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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Exceptional circumstances prevented me to complete the minutes yesterday (a car
hit a power pole, so no power and no access to the audio files Sunday).  In a
normal time I would have been able to finish this by noon, but today is also the
cut-off for submitting I-Ds.  I have another deadline tomorrow, so the minutes
will not be published until Wednesday.  Sorry for that.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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So much efforts and so little progresses...

Changelogs:

draft-petithuguenin-vipr-reload-usage-03

   o  The E.164 number is no longer stored 3 times.
   o  Moved the <store-rate-limit> parameter under the <configuration>
      element.
   o  The VIPR registration now also contains the list of supported PVP
      methods.

draft-petithuguenin-vipr-pvp-02

   o  The Calling number is method "a" is now hashed with the bcrypt
      hash.
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A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : The Public Switched Telephone Network (PSTN) Validation Protocol (PVP)
	Author(s)       : Marc Petit-Huguenin
                          Jonathan Rosenberg
                          Cullen Jennings
	Filename        : draft-petithuguenin-vipr-pvp-02.txt
	Pages           : 30
	Date            : 2011-10-31

   One of the main challenges in inter-domain federation of Session
   Initiation Protocol (SIP) calls is that many domains continue to
   utilize phone numbers, and not email-style SIP URI.  Consequently, a
   mechanism is needed that enables secure mappings from phone numbers
   to domains.  The main technical challenge in doing this securely is
   to verify that the domain in question truly is the &quot;owner&quot; of the
   phone number.  This specification defines the PSTN Validation
   Protocol (PVP), which can be used by a domain to verify this
   ownership by means of a forward routability check in the PSTN.


A URL for this Internet-Draft is:
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Subject: I-D Action: draft-petithuguenin-vipr-reload-usage-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : A Usage of Resource Location and Discovery (RELOAD) for Public Switched Telephone Network (PSTN) Verification
	Author(s)       : Marc Petit-Huguenin
                          Jonathan Rosenberg
                          Cullen Jennings
	Filename        : draft-petithuguenin-vipr-reload-usage-03.txt
	Pages           : 11
	Date            : 2011-10-31

   Verification Involving PSTN Reachability (VIPR) is a technique for
   inter-domain SIP federation.  VIPR makes use of the RELOAD protocol
   to store unverified mappings from phone numbers to RELOAD nodes, with
   whom a validation process can be run.  This document defines the
   usage of RELOAD for this purpose.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-vipr-reload-usage-03.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-petithuguenin-vipr-reload-usage-03.txt
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