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Subject: [stir] I-D Action: draft-ietf-stir-problem-statement-04.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Secure Telephone Identity Revisited Working Group of the IETF.

        Title           : Secure Telephone Identity Problem Statement and Requirements
        Authors         : Jon Peterson
                          Henning Schulzrinne
                          Hannes Tschofenig
	Filename        : draft-ietf-stir-problem-statement-04.txt
	Pages           : 25
	Date            : 2014-05-09

Abstract:
   Over the past decade, Voice over IP (VoIP) systems based on SIP have
   replaced many traditional telephony deployments.  Interworking VoIP
   systems with the traditional telephone network has reduced the
   overall security of calling party number and Caller ID assurances by
   granting attackers new and inexpensive tools to impersonate or
   obscure calling party numbers when orchestrating bulk commercial
   calling schemes, hacking voicemail boxes or even circumventing multi-
   factor authentication systems trusted by banks.  Despite previous
   attempts to provide a secure assurance of the origin of SIP
   communications, we still lack of effective standards for identifying
   the calling party in a VoIP session.  This document examines the
   reasons why providing identity for telephone numbers on the Internet
   has proven so difficult, and shows how changes in the last decade may
   provide us with new strategies for attaching a secure identity to SIP
   sessions.  It also gives high-level requirements for a solution in
   this space.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stir-problem-statement/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stir-problem-statement-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stir-problem-statement-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Fri May  9 12:11:56 2014
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Subject: [stir] I-D Action: draft-ietf-stir-problem-statement-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Secure Telephone Identity Revisited Working Group of the IETF.

        Title           : Secure Telephone Identity Problem Statement and Requirements
        Authors         : Jon Peterson
                          Henning Schulzrinne
                          Hannes Tschofenig
	Filename        : draft-ietf-stir-problem-statement-05.txt
	Pages           : 25
	Date            : 2014-05-09

Abstract:
   Over the past decade, Voice over IP (VoIP) systems based on SIP have
   replaced many traditional telephony deployments.  Interworking VoIP
   systems with the traditional telephone network has reduced the
   overall security of calling party number and Caller ID assurances by
   granting attackers new and inexpensive tools to impersonate or
   obscure calling party numbers when orchestrating bulk commercial
   calling schemes, hacking voicemail boxes or even circumventing multi-
   factor authentication systems trusted by banks.  Despite previous
   attempts to provide a secure assurance of the origin of SIP
   communications, we still lack of effective standards for identifying
   the calling party in a VoIP session.  This document examines the
   reasons why providing identity for telephone numbers on the Internet
   has proven so difficult, and shows how changes in the last decade may
   provide us with new strategies for attaching a secure identity to SIP
   sessions.  It also gives high-level requirements for a solution in
   this space.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stir-problem-statement/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stir-problem-statement-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stir-problem-statement-05


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Fri May  9 12:16:50 2014
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "stir@ietf.org" <stir@ietf.org>
Thread-Topic: [stir] I-D Action: draft-ietf-stir-problem-statement-04.txt
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Subject: Re: [stir] I-D Action: draft-ietf-stir-problem-statement-04.txt
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This version of the document reflects IESG review fixes. One resounding
comment from the IESG was that the document contained some high-level
requirements but did not mention requirements in its title or abstract;
that has been amended.

The most substantial change is the addition of section 6.6, "Concerns
about Pervasive Monitoring," to the list of current events that are
motivating our work.

There were also a number of smaller editorial changes, including some
last-minute nits that Sanjay found.

Comments still welcome, but this should be pretty much good to go.

Jon Peterson
Neustar, Inc.

On 5/9/14, 12:06 PM, "internet-drafts@ietf.org" <internet-drafts@ietf.org>
wrote:

>
>A New Internet-Draft is available from the on-line Internet-Drafts
>directories.
> This draft is a work item of the Secure Telephone Identity Revisited
>Working Group of the IETF.
>
>        Title           : Secure Telephone Identity Problem Statement and
>Requirements
>        Authors         : Jon Peterson
>                          Henning Schulzrinne
>                          Hannes Tschofenig
>	Filename        : draft-ietf-stir-problem-statement-04.txt
>	Pages           : 25
>	Date            : 2014-05-09
>
>Abstract:
>   Over the past decade, Voice over IP (VoIP) systems based on SIP have
>   replaced many traditional telephony deployments.  Interworking VoIP
>   systems with the traditional telephone network has reduced the
>   overall security of calling party number and Caller ID assurances by
>   granting attackers new and inexpensive tools to impersonate or
>   obscure calling party numbers when orchestrating bulk commercial
>   calling schemes, hacking voicemail boxes or even circumventing multi-
>   factor authentication systems trusted by banks.  Despite previous
>   attempts to provide a secure assurance of the origin of SIP
>   communications, we still lack of effective standards for identifying
>   the calling party in a VoIP session.  This document examines the
>   reasons why providing identity for telephone numbers on the Internet
>   has proven so difficult, and shows how changes in the last decade may
>   provide us with new strategies for attaching a secure identity to SIP
>   sessions.  It also gives high-level requirements for a solution in
>   this space.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-stir-problem-statement/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-stir-problem-statement-04
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stir-problem-statement-04
>
>
>Please note that it may take a couple of minutes from the time of
>submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>_______________________________________________
>stir mailing list
>stir@ietf.org
>https://www.ietf.org/mailman/listinfo/stir


From nobody Wed May 14 06:10:08 2014
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Subject: [stir] Document Action: 'Secure Telephone Identity Problem Statement and Requirements' to Informational RFC (draft-ietf-stir-problem-statement-05.txt)
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The IESG has approved the following document:
- 'Secure Telephone Identity Problem Statement and Requirements'
  (draft-ietf-stir-problem-statement-05.txt) as Informational RFC

This document is the product of the Secure Telephone Identity Revisited
Working Group.

The IESG contact persons are Richard Barnes and Alissa Cooper.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-stir-problem-statement/




Technical Summary

   Over the past decade, Voice over IP (VoIP) systems based on SIP have
   replaced many traditional telephony deployments.  Interworking VoIP
   systems with the traditional telephone network has reduced the
   overall security of calling party number and Caller ID assurances by
   granting attackers new and inexpensive tools to impersonate or
   obscure calling party numbers when orchestrating bulk commercial
   calling schemes, hacking voicemail boxes or even circumventing multi-
   factor authentication systems trusted by banks.  Despite previous
   attempts to provide a secure assurance of the origin of SIP
   communications, we still lack of effective standards for identifying
   the calling party in a VoIP session.  This document examines the
   reasons why providing identity for telephone numbers on the Internet
   has proven so difficult, and shows how changes in the last decade may
   provide us with new strategies for attaching a secure identity to SIP
   sessions.

Working Group Summary

  This document is a product of the STIR working group. 

Document Quality

  This document and its predecessors received significant review
  from during the working group formation stages to its current form.
  There is solid consensus that it reflects the problem the working
  group intends to address.

Personnel

  Robert Sparks is the document shepherd. 
  Richard Barnes is the Responsible AD.



From nobody Wed May 28 13:22:09 2014
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "stir@ietf.org" <stir@ietf.org>
Thread-Topic: A possible exploit for stir
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--_000_CFABBDA56CE9Abrianrosenneustarbiz_
Content-Type: text/plain; charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Suppose a caller with TN1 is on a call with TN2, and TN2 is served by evil.=
com.

If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign, an I=
NVITE From TN1, To TN3 and forward it to evil.

Evil can then call TN3, claim it=92s from TN1 and exploit.

As a practical matter, TN1=92s attempt to address a call to TN3@evil.com wo=
n=92t actually work.  The domain will get rewritten, and the call will be r=
outed by TN rules, and not domains.

But strictly speaking, an exploit like this is possible, and we should docu=
ment it.  Devices should not create random extra calls based on a REFER to =
domains they don=92t trust.

Paul Kyzivat pointed this possibility out in a discussion on how to handle =
a problem in another domain.

Brian

--_000_CFABBDA56CE9Abrianrosenneustarbiz_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <B65E00429973F54EA85A0F0CD07076FD@neustar.biz>
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Suppose a caller with TN1 is on a call with TN2, and TN2 is served by =
evil.com.</div>
<div><br>
</div>
<div>If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,=
 an INVITE From TN1, To TN3 and forward it to evil.</div>
<div><br>
</div>
<div>Evil can then call TN3, claim it=92s from TN1 and exploit.</div>
<div><br>
</div>
<div>As a practical matter, TN1=92s attempt to address a call to TN3@evil.c=
om won=92t actually work. &nbsp;The domain will get rewritten, and the call=
 will be routed by TN rules, and not domains.</div>
<div><br>
</div>
<div>But strictly speaking, an exploit like this is possible, and we should=
 document it. &nbsp;Devices should not create random extra calls based on a=
 REFER to domains they don=92t trust.</div>
<div><br>
</div>
<div>Paul Kyzivat pointed this possibility out in a discussion on how to ha=
ndle a problem in another domain.</div>
<div><br>
</div>
<div>Brian</div>
</body>
</html>

--_000_CFABBDA56CE9Abrianrosenneustarbiz_--


From nobody Wed May 28 13:57:41 2014
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Subject: Re: [stir] A possible exploit for stir
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On 5/28/14 4:21 PM, Rosen, Brian wrote:
> Suppose a caller with TN1 is on a call with TN2, and TN2 is served by
> evil.com.
>
> If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,
> an INVITE From TN1, To TN3 and forward it to evil.
>
> Evil can then call TN3, claim it’s from TN1 and exploit.
>
> As a practical matter, TN1’s attempt to address a call to TN3@evil.com
> won’t actually work.  The domain will get rewritten, and the call will
> be routed by TN rules, and not domains.

I don't know if that is true or not. There are *many* ways of handling 
transfer.

In the cases I'm familiar with, some server in the domain of the party 
initiating the transfer wants to remain in the signaling and media 
paths. There are multiple reasons for this. Taking responsibility for 
the cost of the new call is one. Keeping control for "lawful" intercept 
is another.

In those cases, if the domain in control doesn't have signing authority 
for the number of the remaining party in the call, then it won't be able 
to sign the new call that way. It can probably sign on behalf of the 
departing party (transferrer). In some cases that may be acceptable, or 
even a feature. In other cases it might be a change from current 
behavior, for good or bad.

And are we sure that there aren't any practical cases where REFER would 
work without intervention?

> But strictly speaking, an exploit like this is possible, and we should
> document it.  Devices should not create random extra calls based on a
> REFER to domains they don’t trust.
>
> Paul Kyzivat pointed this possibility out in a discussion on how to
> handle a problem in another domain.

Don't sell yourself short. You invented this exploit as a feature. I 
just recognized it as an exploit.

	Thanks,
	Paul


From nobody Wed May 28 16:38:26 2014
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "stir@ietf.org" <stir@ietf.org>
Thread-Topic: [stir] A possible exploit for stir
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Yes, there are many ways of implementing and handling transfer, so many
that it's difficult to get too specific about what the threat here might
be. The threats document already notes that "using transfer mechanisms
common in telephone systems, a callee can easily be connected to, or
conferenced with, telephone numbers other than the original calling number
once a call has been established." But evaluating the risks of this type
of transfer will be highly solution-specific, and for the moment we're
deferring solution-specific threats to the solution documents, rather than
the threat document.

Speculating along solution lines for a moment, this specific example begs
questions about replay protection, URL handling, REFER authorization and
many other issues. Any intermediary in a SIP call path can try to act as a
man-in-the-middle, and what this REFER really does is to bait a call to a
specific number so that an intermediary can man-in-the-middle it. We've
set the scope of STIR to explicitly not address man-in-the-middle attacks,
but perhaps the baiting dimension justifies addressing this.

Evil's ability to cut-and-paste the Identity token in the caller's INVITE
depends largely on the solution's scope of protection. It is possible to
create a scope of protection for a signature which is tied to the actual
source and sinks of media employed by the caller - this is the effect of
signing the SDP, for example. Of course the modification of those fields
by SBCs often renders this approach impractical. Maybe though for the case
where the caller receives a REFER like this, they would want to trigger
signing the SDP, and if the signature shows up broken, well, the callee
then would know not to trust the calling party number. Or maybe more
modestly, some other optional indication could be added to the
message/signature just to warn the recipient, "hey, this was a REFER, so
all bets are off."

To the URL handling question, if evil.com forwards a call To
sip:xyz@evil.com to gateway.com, what is the behavior we expect from
gateway.com? Is it different if "xyz" is a telephone number? Our
expectations is that requests will be routed differently to TNs, but this
is highly implementation-specific, despite our best efforts to standardize
behavior (see RFC3262 vs. RFC3824). But we may have an opportunity in
defining the STIR solution to propose authorization mechanisms for
gateway.com that could allow them to identify abuse more easily.
Ultimately, STIR is creating an Internet service that makes it possible to
figure out who has authority over telephone numbers: while we are
initially using that to answer the question of who should sign requests,
maybe it could also be leveraged to figure out that a call to TN3
shouldn't go to evil.com. That could be part of a REFER authorization
decision at the caller, even.

Jon Peterson
Neustar, Inc.

On 5/28/14, 1:57 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:

>On 5/28/14 4:21 PM, Rosen, Brian wrote:
>> Suppose a caller with TN1 is on a call with TN2, and TN2 is served by
>> evil.com.
>>
>> If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,
>> an INVITE From TN1, To TN3 and forward it to evil.
>>
>> Evil can then call TN3, claim it=B9s from TN1 and exploit.
>>
>> As a practical matter, TN1=B9s attempt to address a call to TN3@evil.com
>> won=B9t actually work.  The domain will get rewritten, and the call will
>> be routed by TN rules, and not domains.
>
>I don't know if that is true or not. There are *many* ways of handling
>transfer.
>
>In the cases I'm familiar with, some server in the domain of the party
>initiating the transfer wants to remain in the signaling and media
>paths. There are multiple reasons for this. Taking responsibility for
>the cost of the new call is one. Keeping control for "lawful" intercept
>is another.
>
>In those cases, if the domain in control doesn't have signing authority
>for the number of the remaining party in the call, then it won't be able
>to sign the new call that way. It can probably sign on behalf of the
>departing party (transferrer). In some cases that may be acceptable, or
>even a feature. In other cases it might be a change from current
>behavior, for good or bad.
>
>And are we sure that there aren't any practical cases where REFER would
>work without intervention?
>
>> But strictly speaking, an exploit like this is possible, and we should
>> document it.  Devices should not create random extra calls based on a
>> REFER to domains they don=B9t trust.
>>
>> Paul Kyzivat pointed this possibility out in a discussion on how to
>> handle a problem in another domain.
>
>Don't sell yourself short. You invented this exploit as a feature. I
>just recognized it as an exploit.
>
>	Thanks,
>	Paul
>
>_______________________________________________
>stir mailing list
>stir@ietf.org
>https://urldefense.proofpoint.com/v1/url?u=3Dhttps://www.ietf.org/mailman/=
li
>stinfo/stir&k=3DlQ50IrZ4n2wmPbDBDzKBYw%3D%3D%0A&r=3D8DMZR2OPVO0EflFSzDaVn8=
Q%2B
>31F1WQyjSQmLQXwHuHE%3D%0A&m=3DdSuvTgTyCXcnEPBvLhXaCRz2UjhIJMH4ckE9L3atntk%=
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>%0A&s=3D19d2d578965bf7e9937d402cde8b4502528251e7fd46ded654e7dd557fdf43b2


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From: "DOLLY, MARTIN C" <md3135@att.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "stir@ietf.org" <stir@ietf.org>
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References: <CFABBDA5.6CE9A%brian.rosen@neustar.biz> <53864DBE.4020301@alum.mit.edu> <CFABB3AE.101889%jon.peterson@neustar.biz>
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Subject: Re: [stir] A possible exploit for stir
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Typical use of REFER in managed deployments is between the UE/Enterprise an=
d the Carrier AS, and it does not transverse the NNI

-----Original Message-----
From: stir [mailto:stir-bounces@ietf.org] On Behalf Of Peterson, Jon
Sent: Wednesday, May 28, 2014 7:38 PM
To: Paul Kyzivat; stir@ietf.org
Subject: Re: [stir] A possible exploit for stir


Yes, there are many ways of implementing and handling transfer, so many
that it's difficult to get too specific about what the threat here might
be. The threats document already notes that "using transfer mechanisms
common in telephone systems, a callee can easily be connected to, or
conferenced with, telephone numbers other than the original calling number
once a call has been established." But evaluating the risks of this type
of transfer will be highly solution-specific, and for the moment we're
deferring solution-specific threats to the solution documents, rather than
the threat document.

Speculating along solution lines for a moment, this specific example begs
questions about replay protection, URL handling, REFER authorization and
many other issues. Any intermediary in a SIP call path can try to act as a
man-in-the-middle, and what this REFER really does is to bait a call to a
specific number so that an intermediary can man-in-the-middle it. We've
set the scope of STIR to explicitly not address man-in-the-middle attacks,
but perhaps the baiting dimension justifies addressing this.

Evil's ability to cut-and-paste the Identity token in the caller's INVITE
depends largely on the solution's scope of protection. It is possible to
create a scope of protection for a signature which is tied to the actual
source and sinks of media employed by the caller - this is the effect of
signing the SDP, for example. Of course the modification of those fields
by SBCs often renders this approach impractical. Maybe though for the case
where the caller receives a REFER like this, they would want to trigger
signing the SDP, and if the signature shows up broken, well, the callee
then would know not to trust the calling party number. Or maybe more
modestly, some other optional indication could be added to the
message/signature just to warn the recipient, "hey, this was a REFER, so
all bets are off."

To the URL handling question, if evil.com forwards a call To
sip:xyz@evil.com to gateway.com, what is the behavior we expect from
gateway.com? Is it different if "xyz" is a telephone number? Our
expectations is that requests will be routed differently to TNs, but this
is highly implementation-specific, despite our best efforts to standardize
behavior (see RFC3262 vs. RFC3824). But we may have an opportunity in
defining the STIR solution to propose authorization mechanisms for
gateway.com that could allow them to identify abuse more easily.
Ultimately, STIR is creating an Internet service that makes it possible to
figure out who has authority over telephone numbers: while we are
initially using that to answer the question of who should sign requests,
maybe it could also be leveraged to figure out that a call to TN3
shouldn't go to evil.com. That could be part of a REFER authorization
decision at the caller, even.

Jon Peterson
Neustar, Inc.

On 5/28/14, 1:57 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:

>On 5/28/14 4:21 PM, Rosen, Brian wrote:
>> Suppose a caller with TN1 is on a call with TN2, and TN2 is served by
>> evil.com.
>>
>> If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,
>> an INVITE From TN1, To TN3 and forward it to evil.
>>
>> Evil can then call TN3, claim it=B9s from TN1 and exploit.
>>
>> As a practical matter, TN1=B9s attempt to address a call to TN3@evil.com
>> won=B9t actually work.  The domain will get rewritten, and the call will
>> be routed by TN rules, and not domains.
>
>I don't know if that is true or not. There are *many* ways of handling
>transfer.
>
>In the cases I'm familiar with, some server in the domain of the party
>initiating the transfer wants to remain in the signaling and media
>paths. There are multiple reasons for this. Taking responsibility for
>the cost of the new call is one. Keeping control for "lawful" intercept
>is another.
>
>In those cases, if the domain in control doesn't have signing authority
>for the number of the remaining party in the call, then it won't be able
>to sign the new call that way. It can probably sign on behalf of the
>departing party (transferrer). In some cases that may be acceptable, or
>even a feature. In other cases it might be a change from current
>behavior, for good or bad.
>
>And are we sure that there aren't any practical cases where REFER would
>work without intervention?
>
>> But strictly speaking, an exploit like this is possible, and we should
>> document it.  Devices should not create random extra calls based on a
>> REFER to domains they don=B9t trust.
>>
>> Paul Kyzivat pointed this possibility out in a discussion on how to
>> handle a problem in another domain.
>
>Don't sell yourself short. You invented this exploit as a feature. I
>just recognized it as an exploit.
>
>	Thanks,
>	Paul
>
>_______________________________________________
>stir mailing list
>stir@ietf.org
>https://urldefense.proofpoint.com/v1/url?u=3Dhttps://www.ietf.org/mailman/=
li
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Subject: Re: [stir] A possible exploit for stir
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Martin,

On 5/28/14 9:04 PM, DOLLY, MARTIN C wrote:
> Typical use of REFER in managed deployments is between the UE/Enterprise and the Carrier AS, and it does not transverse the NNI

Can you explain in a little more detail?
Are you describing the same approach I mentioned earlier in this thread?
(Repeating that:)

> In the cases I'm familiar with, some server in the domain of the party
> initiating the transfer wants to remain in the signaling and media
> paths. There are multiple reasons for this. Taking responsibility for
> the cost of the new call is one. Keeping control for "lawful" intercept
> is another.
>
> In those cases, if the domain in control doesn't have signing authority
> for the number of the remaining party in the call, then it won't be
> able to sign the new call that way. It can probably sign on behalf of
> the departing party (transferrer). In some cases that may be
> acceptable, or even a feature. In other cases it might be a change from
> current behavior, for good or bad.

IMO this is a fine approach, if it provides the semantics that the 
transferring party wants. But it constrains what sort of signing is 
possible.

In particular, is it sufficient that the transferred-to party doesn't 
see the identity of the party to which he will be speaking?

	Thanks,
	Paul

> -----Original Message-----
> From: stir [mailto:stir-bounces@ietf.org] On Behalf Of Peterson, Jon
> Sent: Wednesday, May 28, 2014 7:38 PM
> To: Paul Kyzivat; stir@ietf.org
> Subject: Re: [stir] A possible exploit for stir
>
>
> Yes, there are many ways of implementing and handling transfer, so many
> that it's difficult to get too specific about what the threat here might
> be. The threats document already notes that "using transfer mechanisms
> common in telephone systems, a callee can easily be connected to, or
> conferenced with, telephone numbers other than the original calling number
> once a call has been established." But evaluating the risks of this type
> of transfer will be highly solution-specific, and for the moment we're
> deferring solution-specific threats to the solution documents, rather than
> the threat document.
>
> Speculating along solution lines for a moment, this specific example begs
> questions about replay protection, URL handling, REFER authorization and
> many other issues. Any intermediary in a SIP call path can try to act as a
> man-in-the-middle, and what this REFER really does is to bait a call to a
> specific number so that an intermediary can man-in-the-middle it. We've
> set the scope of STIR to explicitly not address man-in-the-middle attacks,
> but perhaps the baiting dimension justifies addressing this.
>
> Evil's ability to cut-and-paste the Identity token in the caller's INVITE
> depends largely on the solution's scope of protection. It is possible to
> create a scope of protection for a signature which is tied to the actual
> source and sinks of media employed by the caller - this is the effect of
> signing the SDP, for example. Of course the modification of those fields
> by SBCs often renders this approach impractical. Maybe though for the case
> where the caller receives a REFER like this, they would want to trigger
> signing the SDP, and if the signature shows up broken, well, the callee
> then would know not to trust the calling party number. Or maybe more
> modestly, some other optional indication could be added to the
> message/signature just to warn the recipient, "hey, this was a REFER, so
> all bets are off."
>
> To the URL handling question, if evil.com forwards a call To
> sip:xyz@evil.com to gateway.com, what is the behavior we expect from
> gateway.com? Is it different if "xyz" is a telephone number? Our
> expectations is that requests will be routed differently to TNs, but this
> is highly implementation-specific, despite our best efforts to standardize
> behavior (see RFC3262 vs. RFC3824). But we may have an opportunity in
> defining the STIR solution to propose authorization mechanisms for
> gateway.com that could allow them to identify abuse more easily.
> Ultimately, STIR is creating an Internet service that makes it possible to
> figure out who has authority over telephone numbers: while we are
> initially using that to answer the question of who should sign requests,
> maybe it could also be leveraged to figure out that a call to TN3
> shouldn't go to evil.com. That could be part of a REFER authorization
> decision at the caller, even.
>
> Jon Peterson
> Neustar, Inc.
>
> On 5/28/14, 1:57 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>
>> On 5/28/14 4:21 PM, Rosen, Brian wrote:
>>> Suppose a caller with TN1 is on a call with TN2, and TN2 is served by
>>> evil.com.
>>>
>>> If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,
>>> an INVITE From TN1, To TN3 and forward it to evil.
>>>
>>> Evil can then call TN3, claim it¹s from TN1 and exploit.
>>>
>>> As a practical matter, TN1¹s attempt to address a call to TN3@evil.com
>>> won¹t actually work.  The domain will get rewritten, and the call will
>>> be routed by TN rules, and not domains.
>>
>> I don't know if that is true or not. There are *many* ways of handling
>> transfer.
>>
>> In the cases I'm familiar with, some server in the domain of the party
>> initiating the transfer wants to remain in the signaling and media
>> paths. There are multiple reasons for this. Taking responsibility for
>> the cost of the new call is one. Keeping control for "lawful" intercept
>> is another.
>>
>> In those cases, if the domain in control doesn't have signing authority
>> for the number of the remaining party in the call, then it won't be able
>> to sign the new call that way. It can probably sign on behalf of the
>> departing party (transferrer). In some cases that may be acceptable, or
>> even a feature. In other cases it might be a change from current
>> behavior, for good or bad.
>>
>> And are we sure that there aren't any practical cases where REFER would
>> work without intervention?
>>
>>> But strictly speaking, an exploit like this is possible, and we should
>>> document it.  Devices should not create random extra calls based on a
>>> REFER to domains they don¹t trust.
>>>
>>> Paul Kyzivat pointed this possibility out in a discussion on how to
>>> handle a problem in another domain.
>>
>> Don't sell yourself short. You invented this exploit as a feature. I
>> just recognized it as an exploit.
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> stir mailing list
>> stir@ietf.org
>> https://urldefense.proofpoint.com/v1/url?u=https://www.ietf.org/mailman/li
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>
> _______________________________________________
> stir mailing list
> stir@ietf.org
> https://www.ietf.org/mailman/listinfo/stir
>


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Jon,

You identified the central security issue here.

One should always include the intended target of a signed "thing"
under the signature, to avoid re-direction attacks. This is a common
security design principle and needs to be part of the STIR design, as a 
matter
of good practice. (The source is usually identified anyway, because 
that's needed
to select the right credential to verify the sig.)

Sorry I didn't jump in immediately when Brian posted his message.

Steve


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Subject: Re: [stir] A possible exploit for stir
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That doesn=92t help.

A legitimate transfer has the same signaling, and would work.  If TN1 =
calls TN2 and TN2 desires to transfer the call to TN3, it sends a REFER =
to TN1 to do that.  TN1 responds with an INVITE to TN3.  Doesn=92t =
matter if the phone or some B2BUA handles the transfer, it has the same =
effect.

If evil negotiates the media in that replacement call to replicate what =
is already in place for TN1 to TN2, neither party will be aware of any =
change.
But then evil can create a new call from anyone to TN3, claim it=92s =
from TN1 and have the correct signature.

Brian

On May 28, 2014, at 9:04 PM, DOLLY, MARTIN C <md3135@att.com> wrote:

> Typical use of REFER in managed deployments is between the =
UE/Enterprise and the Carrier AS, and it does not transverse the NNI
>=20
> -----Original Message-----
> From: stir [mailto:stir-bounces@ietf.org] On Behalf Of Peterson, Jon
> Sent: Wednesday, May 28, 2014 7:38 PM
> To: Paul Kyzivat; stir@ietf.org
> Subject: Re: [stir] A possible exploit for stir
>=20
>=20
> Yes, there are many ways of implementing and handling transfer, so =
many
> that it's difficult to get too specific about what the threat here =
might
> be. The threats document already notes that "using transfer mechanisms
> common in telephone systems, a callee can easily be connected to, or
> conferenced with, telephone numbers other than the original calling =
number
> once a call has been established." But evaluating the risks of this =
type
> of transfer will be highly solution-specific, and for the moment we're
> deferring solution-specific threats to the solution documents, rather =
than
> the threat document.
>=20
> Speculating along solution lines for a moment, this specific example =
begs
> questions about replay protection, URL handling, REFER authorization =
and
> many other issues. Any intermediary in a SIP call path can try to act =
as a
> man-in-the-middle, and what this REFER really does is to bait a call =
to a
> specific number so that an intermediary can man-in-the-middle it. =
We've
> set the scope of STIR to explicitly not address man-in-the-middle =
attacks,
> but perhaps the baiting dimension justifies addressing this.
>=20
> Evil's ability to cut-and-paste the Identity token in the caller's =
INVITE
> depends largely on the solution's scope of protection. It is possible =
to
> create a scope of protection for a signature which is tied to the =
actual
> source and sinks of media employed by the caller - this is the effect =
of
> signing the SDP, for example. Of course the modification of those =
fields
> by SBCs often renders this approach impractical. Maybe though for the =
case
> where the caller receives a REFER like this, they would want to =
trigger
> signing the SDP, and if the signature shows up broken, well, the =
callee
> then would know not to trust the calling party number. Or maybe more
> modestly, some other optional indication could be added to the
> message/signature just to warn the recipient, "hey, this was a REFER, =
so
> all bets are off."
>=20
> To the URL handling question, if evil.com forwards a call To
> sip:xyz@evil.com to gateway.com, what is the behavior we expect from
> gateway.com? Is it different if "xyz" is a telephone number? Our
> expectations is that requests will be routed differently to TNs, but =
this
> is highly implementation-specific, despite our best efforts to =
standardize
> behavior (see RFC3262 vs. RFC3824). But we may have an opportunity in
> defining the STIR solution to propose authorization mechanisms for
> gateway.com that could allow them to identify abuse more easily.
> Ultimately, STIR is creating an Internet service that makes it =
possible to
> figure out who has authority over telephone numbers: while we are
> initially using that to answer the question of who should sign =
requests,
> maybe it could also be leveraged to figure out that a call to TN3
> shouldn't go to evil.com. That could be part of a REFER authorization
> decision at the caller, even.
>=20
> Jon Peterson
> Neustar, Inc.
>=20
> On 5/28/14, 1:57 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>=20
>> On 5/28/14 4:21 PM, Rosen, Brian wrote:
>>> Suppose a caller with TN1 is on a call with TN2, and TN2 is served =
by
>>> evil.com.
>>>=20
>>> If evil sends TN1 a REFER to TN3@evil.com, it should create, and =
sign,
>>> an INVITE =46rom TN1, To TN3 and forward it to evil.
>>>=20
>>> Evil can then call TN3, claim it=B9s from TN1 and exploit.
>>>=20
>>> As a practical matter, TN1=B9s attempt to address a call to =
TN3@evil.com
>>> won=B9t actually work.  The domain will get rewritten, and the call =
will
>>> be routed by TN rules, and not domains.
>>=20
>> I don't know if that is true or not. There are *many* ways of =
handling
>> transfer.
>>=20
>> In the cases I'm familiar with, some server in the domain of the =
party
>> initiating the transfer wants to remain in the signaling and media
>> paths. There are multiple reasons for this. Taking responsibility for
>> the cost of the new call is one. Keeping control for "lawful" =
intercept
>> is another.
>>=20
>> In those cases, if the domain in control doesn't have signing =
authority
>> for the number of the remaining party in the call, then it won't be =
able
>> to sign the new call that way. It can probably sign on behalf of the
>> departing party (transferrer). In some cases that may be acceptable, =
or
>> even a feature. In other cases it might be a change from current
>> behavior, for good or bad.
>>=20
>> And are we sure that there aren't any practical cases where REFER =
would
>> work without intervention?
>>=20
>>> But strictly speaking, an exploit like this is possible, and we =
should
>>> document it.  Devices should not create random extra calls based on =
a
>>> REFER to domains they don=B9t trust.
>>>=20
>>> Paul Kyzivat pointed this possibility out in a discussion on how to
>>> handle a problem in another domain.
>>=20
>> Don't sell yourself short. You invented this exploit as a feature. I
>> just recognized it as an exploit.
>>=20
>> 	Thanks,
>> 	Paul
>>=20
>> _______________________________________________
>> stir mailing list
>> stir@ietf.org
>> =
https://urldefense.proofpoint.com/v1/url?u=3Dhttps://www.ietf.org/mailman/=
li
>> =
stinfo/stir&k=3DlQ50IrZ4n2wmPbDBDzKBYw%3D%3D%0A&r=3D8DMZR2OPVO0EflFSzDaVn8=
Q%2B
>> =
31F1WQyjSQmLQXwHuHE%3D%0A&m=3DdSuvTgTyCXcnEPBvLhXaCRz2UjhIJMH4ckE9L3atntk%=
3D
>> =
%0A&s=3D19d2d578965bf7e9937d402cde8b4502528251e7fd46ded654e7dd557fdf43b2
>=20
> _______________________________________________
> stir mailing list
> stir@ietf.org
> https://www.ietf.org/mailman/listinfo/stir
>=20
> _______________________________________________
> stir mailing list
> stir@ietf.org
> https://www.ietf.org/mailman/listinfo/stir


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Subject: Re: [stir] A possible exploit for stir
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So, this problem is wrapped up in how domains and telephone numbers =
interact.  As a practical matter, we usually represent telephone numbers =
as sip:+12025551212@example.com;user=3Dphone.

In some sense, the domain of this SIP URI is usually ignored.  When the =
origination service provider sees that construction, it treats it as a =
phone number.  The practical matter for most domains is that if you =
managed to get a subscriber, or proxy, in example.com=92s domain, a uri =
of sip:+12025551212@evil.com;user=3Dphone, it is almost a certainty that =
somewhere in example.com that URI is going to be changed, or just =
ignored.  example.com will route the call by phone number, using number =
based routing decisions and not domain based decisions.  If the =
=93user=3Dphone=94 part wasn=92t there, at least in theory, the call =
would be routed to evil, because the origination domain would assume =
it=92s a normal sip URI with domain based routing that just happened to =
have a user with that string.  As a practical matter, I think that most =
service providers would reject such a call, or treat it as a telephone =
number and apply number based routing.

The way STIR works is that we canonicalize the TN.  We expressly ignore =
the domain, no matter what it is.  So =
sip:+12025551212@example.com;user=3Dphone, and =
sip:+12025551212@evil.com;user=3Dphone will generate the same digest, =
because we extract the telephone number and ignore the rest.  That is =
essential to deal with how number based routing really works.

The problem arises only if the origination domain will route the call =
using domain based routing to evil.  If it doesn=92t do that, and =
instead either rejects the call or ignores/rewrites the domain, then the =
exploit will fail.

If I were evil, I would try to REFER to sip:2025551212@evil.com.  That =
has the best chance of making it to evil.  If it did, it could rewrite =
it to sip:+12025551212@anyone.com;user=3Dphone, and have the signature =
work.

Brian

On May 29, 2014, at 11:27 AM, Stephen Kent <kent@bbn.com> wrote:

> Jon,
>=20
> You identified the central security issue here.
>=20
> One should always include the intended target of a signed "thing"
> under the signature, to avoid re-direction attacks. This is a common
> security design principle and needs to be part of the STIR design, as =
a matter
> of good practice. (The source is usually identified anyway, because =
that's needed
> to select the right credential to verify the sig.)
>=20
> Sorry I didn't jump in immediately when Brian posted his message.
>=20
> Steve
>=20
> _______________________________________________
> stir mailing list
> stir@ietf.org
> https://www.ietf.org/mailman/listinfo/stir


From nobody Thu May 29 11:25:10 2014
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Subject: Re: [stir] A possible exploit for stir
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On 5/29/14 12:36 PM, Brian Rosen wrote:
> So, this problem is wrapped up in how domains and telephone numbers interact.  As a practical matter, we usually represent telephone numbers as sip:+12025551212@example.com;user=phone.
>
> In some sense, the domain of this SIP URI is usually ignored.  When the origination service provider sees that construction, it treats it as a phone number.  The practical matter for most domains is that if you managed to get a subscriber, or proxy, in example.com’s domain, a uri of sip:+12025551212@evil.com;user=phone, it is almost a certainty that somewhere in example.com that URI is going to be changed, or just ignored.  example.com will route the call by phone number, using number based routing decisions and not domain based decisions.  If the “user=phone” part wasn’t there, at least in theory, the call would be routed to evil, because the origination domain would assume it’s a normal sip URI with domain based routing that just happened to have a user with that string.  As a practical matter, I think that most service providers would reject such a call, or treat it as a telephone number and apply number based routing.

This describes the perversion of SIP that has taken place.

According to 3261 a request URI of sip:+12025551212@evil.com;user=phone 
is to be routed to evil.com. The interpretation of the user part as a 
phone number may *then* be done, and user=phone may be taken into 
account when doing that.

Ignoring the domain is an abuse of sip. Unfortunately it is one that is 
so pervasive that it seems impossible to "fix".

This means that we are left to decide what must be supported, and what 
need not be supported, without any written rules for what is correct and 
what is not.

	Thanks,
	Paul

> The way STIR works is that we canonicalize the TN.  We expressly ignore the domain, no matter what it is.  So sip:+12025551212@example.com;user=phone, and sip:+12025551212@evil.com;user=phone will generate the same digest, because we extract the telephone number and ignore the rest.  That is essential to deal with how number based routing really works.
>
> The problem arises only if the origination domain will route the call using domain based routing to evil.  If it doesn’t do that, and instead either rejects the call or ignores/rewrites the domain, then the exploit will fail.
>
> If I were evil, I would try to REFER to sip:2025551212@evil.com.  That has the best chance of making it to evil.  If it did, it could rewrite it to sip:+12025551212@anyone.com;user=phone, and have the signature work.
>
> Brian
>
> On May 29, 2014, at 11:27 AM, Stephen Kent <kent@bbn.com> wrote:
>
>> Jon,
>>
>> You identified the central security issue here.
>>
>> One should always include the intended target of a signed "thing"
>> under the signature, to avoid re-direction attacks. This is a common
>> security design principle and needs to be part of the STIR design, as a matter
>> of good practice. (The source is usually identified anyway, because that's needed
>> to select the right credential to verify the sig.)
>>
>> Sorry I didn't jump in immediately when Brian posted his message.
>>
>> Steve
>>
>> _______________________________________________
>> stir mailing list
>> stir@ietf.org
>> https://www.ietf.org/mailman/listinfo/stir
>
> _______________________________________________
> stir mailing list
> stir@ietf.org
> https://www.ietf.org/mailman/listinfo/stir
>


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From: "DOLLY, MARTIN C" <md3135@att.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "Peterson, Jon" <jon.peterson@neustar.biz>, "stir@ietf.org" <stir@ietf.org>
Thread-Topic: [stir] A possible exploit for stir
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Subject: Re: [stir] A possible exploit for stir
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Paul

Yes. Sorry I missed it

Martin

-----Original Message-----
From: stir [mailto:stir-bounces@ietf.org] On Behalf Of Paul Kyzivat
Sent: Thursday, May 29, 2014 10:32 AM
To: DOLLY, MARTIN C; Peterson, Jon; stir@ietf.org
Subject: Re: [stir] A possible exploit for stir

Martin,

On 5/28/14 9:04 PM, DOLLY, MARTIN C wrote:
> Typical use of REFER in managed deployments is between the UE/Enterprise =
and the Carrier AS, and it does not transverse the NNI

Can you explain in a little more detail?
Are you describing the same approach I mentioned earlier in this thread?
(Repeating that:)

> In the cases I'm familiar with, some server in the domain of the party
> initiating the transfer wants to remain in the signaling and media
> paths. There are multiple reasons for this. Taking responsibility for
> the cost of the new call is one. Keeping control for "lawful" intercept
> is another.
>
> In those cases, if the domain in control doesn't have signing authority
> for the number of the remaining party in the call, then it won't be
> able to sign the new call that way. It can probably sign on behalf of
> the departing party (transferrer). In some cases that may be
> acceptable, or even a feature. In other cases it might be a change from
> current behavior, for good or bad.

IMO this is a fine approach, if it provides the semantics that the=20
transferring party wants. But it constrains what sort of signing is=20
possible.

In particular, is it sufficient that the transferred-to party doesn't=20
see the identity of the party to which he will be speaking?

	Thanks,
	Paul

> -----Original Message-----
> From: stir [mailto:stir-bounces@ietf.org] On Behalf Of Peterson, Jon
> Sent: Wednesday, May 28, 2014 7:38 PM
> To: Paul Kyzivat; stir@ietf.org
> Subject: Re: [stir] A possible exploit for stir
>
>
> Yes, there are many ways of implementing and handling transfer, so many
> that it's difficult to get too specific about what the threat here might
> be. The threats document already notes that "using transfer mechanisms
> common in telephone systems, a callee can easily be connected to, or
> conferenced with, telephone numbers other than the original calling numbe=
r
> once a call has been established." But evaluating the risks of this type
> of transfer will be highly solution-specific, and for the moment we're
> deferring solution-specific threats to the solution documents, rather tha=
n
> the threat document.
>
> Speculating along solution lines for a moment, this specific example begs
> questions about replay protection, URL handling, REFER authorization and
> many other issues. Any intermediary in a SIP call path can try to act as =
a
> man-in-the-middle, and what this REFER really does is to bait a call to a
> specific number so that an intermediary can man-in-the-middle it. We've
> set the scope of STIR to explicitly not address man-in-the-middle attacks=
,
> but perhaps the baiting dimension justifies addressing this.
>
> Evil's ability to cut-and-paste the Identity token in the caller's INVITE
> depends largely on the solution's scope of protection. It is possible to
> create a scope of protection for a signature which is tied to the actual
> source and sinks of media employed by the caller - this is the effect of
> signing the SDP, for example. Of course the modification of those fields
> by SBCs often renders this approach impractical. Maybe though for the cas=
e
> where the caller receives a REFER like this, they would want to trigger
> signing the SDP, and if the signature shows up broken, well, the callee
> then would know not to trust the calling party number. Or maybe more
> modestly, some other optional indication could be added to the
> message/signature just to warn the recipient, "hey, this was a REFER, so
> all bets are off."
>
> To the URL handling question, if evil.com forwards a call To
> sip:xyz@evil.com to gateway.com, what is the behavior we expect from
> gateway.com? Is it different if "xyz" is a telephone number? Our
> expectations is that requests will be routed differently to TNs, but this
> is highly implementation-specific, despite our best efforts to standardiz=
e
> behavior (see RFC3262 vs. RFC3824). But we may have an opportunity in
> defining the STIR solution to propose authorization mechanisms for
> gateway.com that could allow them to identify abuse more easily.
> Ultimately, STIR is creating an Internet service that makes it possible t=
o
> figure out who has authority over telephone numbers: while we are
> initially using that to answer the question of who should sign requests,
> maybe it could also be leveraged to figure out that a call to TN3
> shouldn't go to evil.com. That could be part of a REFER authorization
> decision at the caller, even.
>
> Jon Peterson
> Neustar, Inc.
>
> On 5/28/14, 1:57 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>
>> On 5/28/14 4:21 PM, Rosen, Brian wrote:
>>> Suppose a caller with TN1 is on a call with TN2, and TN2 is served by
>>> evil.com.
>>>
>>> If evil sends TN1 a REFER to TN3@evil.com, it should create, and sign,
>>> an INVITE From TN1, To TN3 and forward it to evil.
>>>
>>> Evil can then call TN3, claim it=B9s from TN1 and exploit.
>>>
>>> As a practical matter, TN1=B9s attempt to address a call to TN3@evil.co=
m
>>> won=B9t actually work.  The domain will get rewritten, and the call wil=
l
>>> be routed by TN rules, and not domains.
>>
>> I don't know if that is true or not. There are *many* ways of handling
>> transfer.
>>
>> In the cases I'm familiar with, some server in the domain of the party
>> initiating the transfer wants to remain in the signaling and media
>> paths. There are multiple reasons for this. Taking responsibility for
>> the cost of the new call is one. Keeping control for "lawful" intercept
>> is another.
>>
>> In those cases, if the domain in control doesn't have signing authority
>> for the number of the remaining party in the call, then it won't be able
>> to sign the new call that way. It can probably sign on behalf of the
>> departing party (transferrer). In some cases that may be acceptable, or
>> even a feature. In other cases it might be a change from current
>> behavior, for good or bad.
>>
>> And are we sure that there aren't any practical cases where REFER would
>> work without intervention?
>>
>>> But strictly speaking, an exploit like this is possible, and we should
>>> document it.  Devices should not create random extra calls based on a
>>> REFER to domains they don=B9t trust.
>>>
>>> Paul Kyzivat pointed this possibility out in a discussion on how to
>>> handle a problem in another domain.
>>
>> Don't sell yourself short. You invented this exploit as a feature. I
>> just recognized it as an exploit.
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> stir mailing list
>> stir@ietf.org
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n/li
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>
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>

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