Network Working Group J. Peterson
Internet-Draft NeuStar, Inc.
Intended status: Informational H. Schulzrinne
Expires: November 10, 2014 Columbia University
H. Tschofenig
Nokia Siemens Networks
May 9, 2014
Secure Telephone Identity Problem Statement and Requirements
draft-ietf-stir-problem-statement-04.txt
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.
Status of This Memo
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provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 10, 2014.
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Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 6
4.1. VoIP-to-VoIP Call . . . . . . . . . . . . . . . . . . . . 6
4.2. IP-PSTN-IP Call . . . . . . . . . . . . . . . . . . . . . 7
4.3. PSTN-to-VoIP Call . . . . . . . . . . . . . . . . . . . . 8
4.4. VoIP-to-PSTN Call . . . . . . . . . . . . . . . . . . . . 9
4.5. PSTN-VoIP-PSTN Call . . . . . . . . . . . . . . . . . . . 10
4.6. PSTN-to-PSTN Call . . . . . . . . . . . . . . . . . . . . 11
5. Limitations of Current Solutions . . . . . . . . . . . . . . 11
5.1. P-Asserted-Identity . . . . . . . . . . . . . . . . . . . 12
5.2. SIP Identity . . . . . . . . . . . . . . . . . . . . . . 14
5.3. VIPR . . . . . . . . . . . . . . . . . . . . . . . . . . 17
6. Environmental Changes . . . . . . . . . . . . . . . . . . . . 19
6.1. Shift to Mobile Communication . . . . . . . . . . . . . . 19
6.2. Failure of Public ENUM . . . . . . . . . . . . . . . . . 19
6.3. Public Key Infrastructure Developments . . . . . . . . . 20
6.4. Prevalence of B2BUA Deployments . . . . . . . . . . . . . 20
6.5. Stickiness of Deployed Infrastructure . . . . . . . . . . 20
6.6. Concerns about Pervasive Monitoring . . . . . . . . . . . 21
6.7. Relationship with Number Assignment and Management . . . 21
7. Basic Requirements . . . . . . . . . . . . . . . . . . . . . 21
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 22
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
10. Security Considerations . . . . . . . . . . . . . . . . . . . 23
11. Informative References . . . . . . . . . . . . . . . . . . . 23
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 25
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1. Introduction
In many communication architectures that allow users to communicate
with other users, the need arises for identifying the originating
party that initiates a call or a messaging interaction. The desire
for identifying communication parties in end-to-end communication
attempt derives from the need to implement authorization policies (to
grant or reject call attempts) but has also been utilized for
charging. While there are a number of ways to enable identification
this functionality has been provided by the Session Initiation
Protocol (SIP) [RFC3261] by using two main types of approaches,
namely using P-Asserted-Identity (PAI) [RFC3325] and SIP Identity
[RFC4474], which are described in more detail in Section 5. The goal
of these mechanisms is to validate that originator of a call is
authorized to claim an originating identifier. Protocols, like XMPP,
use mechanisms that are conceptually similar to those offered by SIP.
Although solutions have been standardized, it turns out that the
current deployment situation is unsatisfactory and, even worse, there
is little indication that it will be improved in the future. In
[I-D.cooper-iab-secure-origin] we illustrate what challenges arise.
In particular, interworking with different communication
architectures (e.g., SIP, PSTN, XMPP, RTCWeb) or other forms of
mediation breaks the end-to-end semantic of the communication
interaction and destroys any identification capabilities.
Furthermore, the use of different identifiers (e.g., E.164 numbers
vs. SIP URIs) creates challenges for determining who is able to claim
"ownership" for a specific identifier; although domain-based
identifiers (sip:user@example.com) might use certificate or DNS-
related approaches to determine who is able to claim "ownership" of
the URI, telephone numbers do not yet have any similar mechanism
defined.
After the publication of the PAI and SIP Identity specifications
various further attempts have been made to tackle the topic but
unfortunately with little success. The complexity resides in the
deployment situation and the long list of (often conflicting)
requirements. A number of years have passed since the last attempts
were made to improve the situation and we therefore believe it is
time to give it another try. With this document we would like to
start to develop a common understanding of the problem statement as
well as basic requirements to develop a vision on how to advance the
state of the art and to initiate technical work to enable secure call
origin identification.
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2. Problem Statement
In the classical public-switched telephone network, there were a
limited number of carriers, all of whom trusted each other to provide
accurate caller origination information, in an evnironment without
any cryptographic validation. In some cases, national
telecommunication regulation codified these obligations. This model
worked as long as the number of entities was relatively small, easily
identified (e.g., in the manner carriers are certified int he US) and
subject to effective legal sanctions in case of misbehavior.
However, for some time, these assumptions have no longer held true.
For example, entities that are not traditional telecommunication
carriers, possibly located outside the country whose country code
they are using, can act as voice service providers. While in the
past, there was a clear distinction between customers and service
providers, VoIP service providers can now easily act as customers,
originating and transit providers. The problem is moreover not
limited to voice communications, as growth in text messaging has made
it another vector for bulk unsolicited commercial messaging relying
on impersonation of a source telephone number (sometimes a short
code). For telephony, Caller ID spoofing has become common, with a
small subset of entities either ignoring abuse of their services or
willingly serving to enable fraud and other illegal behavior.
For example, recently, enterprises and public safety organizations
[TDOS] have been subjected to telephony denial-of-service attacks.
In this case, an individual claiming to represent a collections
company for payday loans starts the extortion scheme with a phone
call to an organization. Failing to get payment from an individual
or organization, the criminal organization launches a barrage of
phone calls, with spoofed numbers, preventing the targeted
organization from receiving legitimate phone calls. Other boiler-
room organizations use number spoofing to place illegal "robocalls"
(automated telemarketing, see, for example, the US Federal
Communications Commission webpage [robocall-fcc] on this topic).
Robocalls are a problem that has been recognized already by various
regulators; for example, the US Federal Trade Commission (FTC)
recently organized a robocall competition to solicit ideas for
creating solutions that will block illegal robocalls
[robocall-competition]. Criminals may also use number spoofing to
impersonate banks or bank customers to gain access to information or
financial accounts.
In general, number spoofing is used in two ways, impersonation and
anonymization. For impersonation, the attacker pretends to be a
specific individual. Impersonation can be used for pretexting, where
the attacker obtains information about the individual impersonated,
activates credit cards or for harassment, e.g., by causing utility
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services to be disconnected, take-out food to be delivered, or by
causing police to respond to a non-existing hostage situation
("swatting", see [swatting]). Some voicemail systems can be set up
so that they grant access to stored messages without a password,
relying solely on the caller identity. As an example, the News
International phone-hacking scandal [news-hack] has also gained a lot
of press attention where employees of the newspaper were accused of
engaging in phone hacking by utilizing Caller ID spoofing to get
access to a voicemail. For numbers where the caller has suppressed
textual caller identification, number spoofing can be used to
retrieve this information, stored in the so-called Calling Name
(CNAM) database. For anonymization, the caller does not necessarily
care whether the number is in service, or who it is assigned to, and
may switch rapidly and possibly randomly between numbers.
Anonymization facilitates automated illegal telemarketing or
telephony denial-of-service attacks, as described above, as it makes
it difficult to identify perpetators and craft policies to block
them. It also makes tracing such calls much more labor-intensive, as
each call has to be identified in each transit carrier hop-by-hop,
based on destination number and time of call.
It is insufficient to simply outlaw all spoofing of originating
telephone numbers, because the entities spoofing numbers are already
committing other crimes and thus unlikely to be deterred by legal
sanctions. Secure origin identification should prevent impersonation
and, to a lesser extent, anonymization. However, if numbers are easy
and cheap to obtain, and if the organizations assigning identifiers
cannot or will not establish the true corporate or individual
identity of the entity requesting such identifiers, robocallers will
still be able to switch between many different identities.
The problem space is further complicated by a number of use cases
where entities in the telephone network legitimately send calls on
behalf of others, including "Find-Me/Follow-Me" services.
Ultimately, any SIP entity can receive an INVITE and forward it to
any other entity, and the recipient of a forwarded message has little
means to ascertain which recipient a call should legitimately target
(see [I-D.peterson-sipping-retarget]. Also, in some cases, third
parties may need to temporarily use the identity of another
individual or organization, with full consent of the "owner" of the
identifier. For example:
The doctor's office: Physicians calling their patients using their
cell phones would like to replace their mobile phone number with
the number of their office to avoid being called back by patients
on their personal phone.
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Call centers: Call centers operate on behalf of companies and the
called party expects to see the Caller ID of the company, not the
call center.
3. Terminology
The following terms are defined in this document:
In-band Identity Conveyance: In-band conveyance is the presence of
call origin identification information conveyed within the control
plane protocol(s) setting up a call. Any in-band solution must
accommodate prevalence of in-band intermdiaries such as B2BUAs.
Out-of-Band Identity Verification: Out-of-band verification
determines whether the telephone number used by the calling party
actually exists, whether the calling entity is entitled to use the
number and whether a call has recently been made from this phone
number. This approach is needed because the in-band technique
does not work in all cases, as when certain intermediaries are
involved or due to interworking with PSTN networks.
Authority Delegation Infrastructure: This functionality defines how
existing authority over telephone numbers are used in number
portability and delegation cases. It also describes how the
existing numbering infrastructure is re-used to maintain the
lifecycle of number assignments.
Canonical Telephone Number: In order for either in-band conveyance
or out-of-band verification to work, entities in this architecture
must be able to canonicalize telephone numbers to arrive at a
common syntactical form.
4. Use Cases
In order to explain the requirements and other design assumptions we
will explain some of the scenarios that need to be supported by any
solution. To reduce clutter, the figures do not show call routing
elements, such as SIP proxies, of voice or text service providers.
We generally assume that the PSTN component of any call path cannot
be altered.
4.1. VoIP-to-VoIP Call
For the IP-to-IP communication case, a group of service providers
that offer interconnected VoIP service exchange calls using SIP end-
to-end, but may also deliver some calls via circuit-switched
facilities, as described in separate use cases below. These service
providers use telephone numbers as source and destination
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identifiers, either as the user component of a SIP URI (e.g.,
sip:12125551234@example.com) or as a tel URI [RFC3966].
As illustrated in Figure 1, if Alice calls Bob, the call will use SIP
end-to-end. (The call may or may not traverse the Internet.)
+------------+
| IP-based |
| SIP Phone |<--+
| of Bob | |
|+19175551234| |
+------------+ |
|
+------------+ |
| IP-based | |
| SIP Phone | ------------
| of Alice | / | \
|+12121234567| // | \\
+------------+ // ,' \\\
| /// / -----
| //// ,' \\\\
| / ,' \
| | ,' |
+---->|......: IP-based |
| Network |
\ /
\\\\ ////
-------------------------
Figure 1: VoIP-to-VoIP Call.
4.2. IP-PSTN-IP Call
Frequently, two VoIP-based service providers are not directly
connected by VoIP and use TDM circuits to exchange calls, leading to
the IP-PSTN-IP use case. In this use case, Dan's VSP is not a member
of the interconnect federation Alice's and Bob's VSP belongs to. As
far as Alice is concerned Dan is not accessible via IP and the PSTN
is used as an interconnection network. Figure 2 shows the resulting
exchange.
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--------
//// \\\\
+--- >| PSTN |
| | |
| \\\\ ////
| --------
| |
| |
| |
+------------+ +--+----+ |
| IP-based | | PSTN | |
| SIP Phone | --+ VoIP +- v
| of Alice | / | GW | \ +---+---+
|+12121234567| // `''''''' \\| PSTN |
+------------+ // | \+ VoIP +
| /// | | GW |\
| //// | `'''''''\\ +------------+
| / | | \ | IP-based |
| | | | | | Phone |
+---->|---------------+ +------|---->| of Dan |
| | |+12039994321|
\ IP-based / +------------+
\\\\ Network ////
-------------------------
Figure 2: IP-PSTN-IP Call.
Note: A B2BUA/Session Border Controller (SBC) exhibits behavior that
looks similar to this scenario since the original call content would,
in the worst case, be re-created on the call origination side.
4.3. PSTN-to-VoIP Call
Consider Figure 3 where Carl is using a PSTN phone and initiates a
call to Alice. Alice is using a VoIP-based phone. The call from
Carl traverses the PSTN and enters the Internet via a PSTN/VoIP
gateway. This gateway attaches some identity information to the
call, for example, based on the caller identification information it
had received through the PSTN, if available.
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--------
//// \\\\
+->| PSTN |--+
| | | |
| \\\\ //// |
| -------- |
| |
| v
| +----+-------+
+---+------+ |PSTN / VoIP | +-----+
|PSTN Phone| |Gateway | |SIP |
|of Carl | +----+-------+ |UA |
+----------+ | |Alice|
Invite +-----+
| ^
V |
+---------------+ Invite
|VoIP | |
|Interconnection| Invite +-------+
|Provider(s) |----------->+ |
+---------------+ |Alice's|
|VSP |
| |
+-------+
Figure 3: PSTN-to-VoIP Call.
4.4. VoIP-to-PSTN Call
Consider Figure 4 where Alice calls Carl. Carl uses a PSTN phone and
Alice an IP-based phone. When Alice initiates the call, the E.164
number is get translated to a SIP URI and subsequently to an IP
address. The call of Alice traverses her VoIP provider where the
call origin identification information is added. It then hits the
PSTN/VoIP gateway. It is desirable that the gateway verify that
Alice can claim the E.164 number she is using before it populates the
corresponding calling party number field in telephone network
signaling. Carl's phone must be able to verify that it is receiving
a legitimate call from the calling party number it will render to
Carl.
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+-------+ +-----+ -C
|PSTN | |SIP | |a
|Phone |<----------------+ |UA | |l
|of Carl| | |Alice| |l
+-------+ | +-----+ |i
--------------------------- | |n
//// \\\\ | |g
| PSTN | Invite |
| | | |P
\\\\ //// | |a
--------------------------- | |r
^ | |t
| v |y
+------------+ +--------+|
|PSTN / VoIP |<--Invite----|VoIP ||D
|Gateway | |Service ||o
+------------+ |Provider||m
|of Alice||a
+--------+|i
-n
Figure 4: IP-to-PSTN Call.
4.5. PSTN-VoIP-PSTN Call
Consider Figure 5 where Carl calls Alice. Both users have PSTN
phones but interconnection between the two PSTN networks is
accomplished via an IP network. Consequently, Carl's operator uses a
PSTN-to-VoIP gateway to route the call via an IP network to a gateway
to break out into the PSTN again.
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+----------+
|PSTN Phone|
-------- |of Alice |
//// \\\\ +----------+
+->| PSTN |------+ ^
| | | | |
| \\\\ //// | |
| -------- | --------
| v //// \\\\
| ,-------+ | PSTN |
| |PSTN | | |
+---+------+ __|VoIP GW|_ \\\\ ////
|PSTN Phone| / '`''''''' \ --------
|of Carl | // | \\ ^
+----------+ // | \\\ |
/// -. Invite ----- |
//// `-. \\\\ |
/ `.. \ |
| IP-based `._ ,--+----+
| Network `.....>|VoIP |
| |PSTN GW|
\ '`'''''''
\\\\ ////
-------------------------
Figure 5: PSTN-VoIP-PSTN Call.
4.6. PSTN-to-PSTN Call
For the "legacy" case of a PSTN-to-PSTN call, otherwise beyond
improvement, we may be able to use out-of-band IP connectivity at
both the originating and terminating carrier to validate the call
information.
5. Limitations of Current Solutions
From the inception of SIP, the From header field value has held an
arbitrary user-supplied identity, much like the From header field
value of an SMTP email message. During work on [RFC3261], efforts
began to provide a secure origin for SIP requests as an extension to
SIP. The so-called "short term" solution, the P-Asserted-Identity
header described in [RFC3325], is deployed fairly widely, even though
it is limited to closed trusted networks where end-user devices
cannot alter or inspect SIP messages and offers no cryptographic
validation. As P-Asserted-Identity is used increasingly across
multiple networks, it cannot offer any protection against identity
spoofing by intermediaries or entities that allow untrusted entities
to set the P-Asserted-Identity information. An overview of
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addressing spam in SIP, and explaining how it differs from simiilar
problems with email, appeared in [RFC5039].
Subsequent efforts to prevent calling origin identity spoofing in SIP
include the SIP Identity effort (the "long term" identity solution)
[RFC4474] and Verification Involving PSTN Reachability (VIPR)
[I-D.jennings-vipr-overview]. SIP Identity attaches a new header
field to SIP requests containing a signature over the From header
field value combined with other message components to prevent replay
attacks. SIP Identity is meant to prevent both: (a) SIP UAs from
originating calls with spoofed From headers; and (b) intermediaries,
such as SIP proxies, from launching man-in-the-middle attacks by
altering calls as they pass through the intermediaries. The VIPR
architecture attacked a broader range of problems relating to spam,
routing and identity with a new infrastructure for managing
rendezvous and security, which operated alongside of SIP deployments.
As we will describe in more detail below, both SIP Identity and VIPR
suffer from serious limitations that have prevented their deployment
at significant scale, but they may still offer ideas and protocol
building blocks for a solution.
5.1. P-Asserted-Identity
The P-Asserted-Identity header field of SIP [RFC3325] provides a way
for trusted network entities to share with one another an
authoritative identifier for the originator of a call. The value of
P-Asserted-Identity cannot be populated by a user, though if a user
wants to suggest an identity to the trusted network, a separate
header (P-Preferred-Identity) enables them to do so. The features of
the P-Asserted-Identity header evolved as part of a broader effort to
reach parity with traditional telephone network signaling mechanisms
for selectively sharing and restricting presentation of the calling
party number at the user level, while still allowing core network
elements to know the identity of the user for abuse prevention and
accounting.
In order for P-Asserted-Identity to have these properties, it
requires the existence of a trust domain as described in [RFC3324].
Any entity in the trust domain may add a P-Asserted-Identity header
to a SIP message, and any entity in the trust domain may forward a
message with a P-Asserted-Identity header to any other entity in the
trust domain. If a trusted entity forwards a SIP request to an
untrusted entity, however, the P-Asserted-Identity header must first
be removed; most sorts of end user devices are outside trust domains.
Sending a P-Asserted-Identity request to an untrusted entity could
leak potentially private information, such as the network-asserted
calling party number in a case where a caller has requested
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presentation restriction. This concept of a trust domain is modeled
on the trusted network of devices that operate the traditional
telephone network.
P-Asserted-Identity has been very successful in telephone replacement
deployments of SIP. It is an extremely simple in-band mechanism,
requiring no cryptographic operations. Since it is so reminiscent of
legacy mechanisms in the traditional telephone network, and it
interworks so seamlessly with those protocols, it has naturally been
favored by providers comfortable with these operating principles.
In practice, a trust domain exhibits many of the same merits and
flaws as the traditional telephone network when it comes to securing
a calling party number. Any trusted entity may provide P-Asserted-
Identity, and a recipient of a SIP message has no direct assurance of
who generated the P-Asserted-Identity header field value: all trust
is transitive. Trust domains are dictated by business arrangements
more than by security standards, and thus the level of assurance of P
-Asserted-Identity is only as good as the least trustworthy member of
a trust domain. Since the contents of P-Asserted-Identity are not
intended for consumption by end users, end users must trust that
their service provider participates in an appropriate trust domain,
as there will be no direct evidence of the trust domain in SIP
signaling that end user devices receive. Since the mechanism is so
closely modeled on the traditional telephone network, it is unlikely
to provide a higher level of security than that.
Since [RFC3325] was written, the whole notion of P- headers intended
for use in private SIP domains has also been deprecated (see
[RFC5727], largely because of overwhelming evidence that these
headers were being used outside of private contexts and leaking into
the public Internet. It is unclear how many deployments that make
use of P-Asserted-Identity in fact conform with the Spec-T
requirements of RFC3324.
P-Asserted-Identity also complicates the question of which URI should
be presented to a user when a call is received. Per RFC3261, SIP
user agents would render the contents of the From header field to a
user when receiving an INVITE request, but what if the P-Asserted-
Identity contains a more trustworthy URI, and presentation is not
restricted? Subsequent proposals have suggested additional header
fields to carry different forms of identity related to the caller,
including billing identities. As the calling identities in a SIP
request proliferate, the question of how to select one to render to
the end user becomes more difficult to answer.
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5.2. SIP Identity
The SIP Identity mechanism [RFC4474] provided two header fields for
securing identity information in SIP requests: the Identity and
Identity-Info header fields. Architecturally, the SIP Identity
mechanism assumes a classic "SIP trapezoid" deployment in which an
authentication service, acting on behalf of the originator of a SIP
request, attaches identity information to the request which provides
partial integrity protection; a verification service acting on behalf
of the recipient validates the integrity of the request when it is
received.
The Identity header field value contains a signature over a hash of
selected elements of a SIP request, including several header field
values (most significantly, the From header field value) and the
entirety of the body of the request. The set of header field values
was chosen specifically to prevent cut-and-paste attacks; it requires
the verification service to retain some state to guard against
replays. The signature over the body of a request has different
properties for different SIP methods, but all prevent tampering by
man-in-the-middle attacks. For a SIP MESSAGE request, for example,
the signature over the body covers the actual message conveyed by the
request: it is pointless to guarantee the source of a request if a
man-in-the-middle can change the content of the message, as in that
case the message content is created by an attacker. Similar threats
exist against the SIP NOTIFY method. For a SIP INVITE request, a
signature over the SDP body is intended to prevent a man-in-the-
middle from changing properties of the media stream, including the IP
address and port to which media should be sent, as this provides a
means for the man-in-the-middle to direct session media to resource
that the originator did not specify, and thus to impersonate an
intended listener.
The Identity-Info header field value contains a URI designating the
location of the certificate corresponding to the private key that
signed the hash in the Identity header. That certificate could be
passed by-value along with the SIP request, in which case a "cid" URI
appears in Identity-Info, or by-reference, for example when the
Identity-Info header field value has the URL of a service that
delivers the certificate. [RFC4474] imposes further constraints
governing the subject of that certificate: namely, that it must cover
the domain name indicated in the domain component of the URI in the
From header field value of the request.
The SIP Identity mechanism, however, has two fundamental limitations
that have precluded its deployment: first, that it provides Identity
only for domain names rather than other identifiers; second, that it
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does not tolerate intermediaries that alter the bodies, or certain
header fields, of SIP requests.
As deployed, SIP predominantly mimics the structures of the telephone
network, and thus uses telephone numbers as identifiers. Telephone
numbers in the From header field value of a SIP request may appear as
the user part of a SIP URI, or alternatively in an independent tel
URI. The certificate designated by the Identity-Info header field as
specified, however, corresponds only to the domain portion of a SIP
URI in the From header field. As such, [RFC4474] does not have any
provision to identify the assignee of a telephone number. While it
could be the case that the domain name portion of a SIP URI signifies
a carrier (like "att.com") to whom numbers are assigned, the SIP
Identity mechanism provides no assurance that a number is assigned to
any carrier. For a tel URI, moreover, it is unclear in [RFC4474]
what entity should hold a corresponding certificate. A caller may
not want to reveal the identity of its service provider to the
callee, and may thus prefer tel URIs in the From header field.
This lack of authority gives rise to a whole class of SIP identity
problems when dealing with telephone numbers, as is explored in
[I-D.rosenberg-sip-rfc4474-concerns]. That document shows how the
Identity header of a SIP request targeting a telephone number
(embedded in a SIP URI) could be dropped by an intermediate domain,
which then modifies and re-signs the request, all without alerting
the verification service: the verification service has no way of
knowing which original domain signed the request. Provided that the
local authentication service is complicit, an originator can claim
virtually any telephone number, impersonating any chosen Caller ID
from the perspective of the verifier. Both of these attacks are
rooted in the inability of the verification service to ascertain a
specific certificate that is authoritative for a telephone number.
As deployed, SIP is moreover highly mediated, and mediated in ways
that [RFC3261] did not anticipate. As request routing commonly
depends on policies dissimilar to [RFC3263], requests transit
multiple intermediate domains to reach a destination; some forms of
intermediaries in those domains may effectively re-initiate the
session.
One of the main reasons that SIP deployments mimic the PSTN
architecture is because the requirement for interconnection with the
PSTN remains paramount: a call may originate in SIP and terminate on
the PSTN, or vice versa; and worse still, a PSTN-to-PSTN call may
transit a SIP network in the middle, or vice versa. This necessarily
reduces SIP's feature set to the least common dominator of the
telephone network, and mandates support for telephone numbers as a
primary calling identifier.
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Interworking with non-SIP networks makes end-to-end identity
problematic. When a PSTN gateway sends a call to a SIP network, it
creates the INVITE request anew, regardless of whether a previous leg
of the call originated in a SIP network that later dropped the call
to the PSTN. As these gateways are not necessarily operated by
entities that have any relationship to the number assignee, it is
unclear how they could provide an identity signature that a verifier
should trust. Moreover, how could the gateway know that the calling
party number it receives from the PSTN is actually authentic? And
when a gateway receives a call via SIP and terminates a call to the
PSTN, how can that gateway verify that a telephone number in the From
header field value is authentic, before it presents that number as
the calling party number in the PSTN?
Similarly, some SIP networks deploy intermediaries that act as back-
to-back user agents (B2BUAs), typically in order to provide policy or
interworking functions at network boundaries (hence the nickname
"Session Border Controller"). These functions range from topology
hiding, to alterations necessary to interoperate successfully with
particular SIP implementations, to simple network address translation
from private address space. To achieve these aims, these entities
modify SIP INVITE requests in transit, potentially changing the From,
Contact and Call-ID header field values, as well as aspects of the
SDP, including especially the IP addresses and ports associated with
media. Consequently, a SIP request exiting a B2BUA has no necessary
relationship to the original request received by the B2BUA, much like
a request exiting a PSTN gateway has no necessary relationship to any
SIP request in a pre-PSTN leg of the call. An Identity signature
provided for the original INVITE has no bearing on the post-B2BUA
INVITE, and, were the B2BUA to preserve the original Identity header,
any verification service would detect a violation of the integrity
protection.
The SIP community has long been aware of these problems with
[RFC4474] in practical deployments. Some have therefore proposed
weakening the security constraints of [RFC4474] so that at least some
deployments of B2BUAs will be compatible with integrity protection of
SIP requests. However, such solutions do not address one key problem
identified above: the lack of any clear authority for telephone
numbers, and the fact that some INVITE requests are generated by
intermediaries rather than endpoints. Removing the signature over
the SDP from the Identity header will not, for example, make it any
clearer how a PSTN gateway should assert identity in an INVITE
request.
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5.3. VIPR
Verification Involving PSTN Reachability (VIPR) directly attacks the
twin problems of identifying number assignees on the Internet and
coping with intermediaries that may modify signaling. To address the
first problem, VIPR relies on the PSTN itself: it discovers which
endpoints on the Internet are reachable via a particular PSTN number
by calling the number on the PSTN to determine whom a call to that
number will reach. As VIPR-enabled Internet endpoints associated
with PSTN numbers are discovered, VIPR provides a rendez-vous service
that allows the endpoints of a call to form an out-of-band connection
over the Internet; this connection allows the endpoints to exchange
information that secures future communications and permits direct,
unmediated SIP connections.
VIPR provides these services within a fairly narrow scope of
applicability. Its seminal use case is the enterprise IP PBX, a
device that has both PSTN connectivity and Internet connectivity,
which serves a set of local users with telephone numbers; after a
PSTN call has connected successfully and then ended, the PBX searches
a distributed hash-table to see if any VIPR-compatible devices have
advertised themselves as a route for the unfamiliar number on the
Internet. If advertisements exist, the originating PBX then
initiates a verification process to determine whether the entity
claiming to be the assignee of the unfamiliar number in fact received
the successful call: this involves verifying details such as the
start and stop times of the call. If the destination verifies
successfully, the originating PBX provisions a local database with a
route for that telephone number to the URI provided by the proven
destination. The destination moreover gives a token to the
originator that can be inserted in future call setup messages to
authenticate the source of future communications.
Through this mechanism, the VIPR system provides a suite of
properties, ones that go well beyond merely securing the origins of
communications. It also provides a routing system which dynamically
discovers mappings between telephone numbers and URIs, effectively
building an ad hoc ENUM database in every VIPR implementation. The
tokens exchanged over the out-of-band connection established by VIPR
moreover provide an authorization mechanism for accepting calls over
the Internet that significantly reduces the potential for spam.
Because the token can act as a cookie due to the presence of this
out-of-band connectivity, the VIPR token is less susceptible to cut-
and-paste attacks and thus needs to cover with its signature far less
of a SIP request.
Due to its narrow scope of applicability, and the details of its
implementation, VIPR has some significant limitations. The most
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salient for the purposes of this document is that it only has bearing
on repeated communications between entities: it has no solution to
the classic "robocall" problem, where the target typically receives a
call from a number that has never called before. All of VIPR's
strengths in establishing identity and spam prevention kick in only
after an initial PSTN call has been completed, and subsequent
attempts at communication begin. Every VIPR-compliant entity
moreover maintains its own stateful database of previous contacts and
authorizations, which lends itself to more aggregators like IP PBXs
that may front for thousands of users than to individual phones.
That database must be refreshed by periodic PSTN calls to determine
that control over the number has not shifted to some other entity;
figuring out when data has grown stale is one the challenges of the
architecture. As VIPR requires compliant implementations to operate
both a PSTN interface and an IP interface, it has little apparent
applicability to ordinary desktop PCs or similar devices with no
ability to place direct PSTN calls.
The distributed hash table also creates a new attack surface for
impersonation. Attackers who want to pose as the owners of telephone
numbers can advertise themselves as routes to a number in the hash
table. VIPR has no inherent restriction on the number of entities
that may advertise themselves as routes for a number, and thus an
originator may find multiple advertisements for a number on the DHT
even when an attack is not in progress. As for attackers, even if
they cannot successfully verify themselves to the originators of
calls (because they lack the call detail information), they may learn
from those verification attempts which VIPR entities recently placed
calls to the target number: it may be that this information is all
the attacker hopes to glean. The fact that advertisements and
verifications are public results from the public nature of the DHT
that VIPR creates. The public DHT prevents any centralized control,
or attempts to impede communications, but those come at the cost of
apparently unavoidable privacy losses.
Because of these limitations, VIPR, much like SIP Identity, has had
little impact in the marketplace. Ultimately, VIPR's utility as an
identity mechanism is limited by its reliance on the PSTN, especially
its need for an initial PSTN call to complete before any of VIPR's
benefits can be realized, and by the drawbacks of the highly-public
exchanges requires to create the out-of-band connection between VIPR
entities. As such, there is no obvious solution to providing secure
origin services for SIP on the Internet today.
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6. Environmental Changes
6.1. Shift to Mobile Communication
In the years since [RFC4474] was conceived, there have been a number
of fundamental shifts in the communications marketplace. The most
transformative has been the precipitous rise of mobile smart phones,
which are now arguably the dominant communications device in the
developed world. Smart phones have both a PSTN and an IP interface,
as well as an SMS and MMS capabilities. This suite of tools suggests
that some of the techniques proposed by VIPR could be adapted to the
smart phone environment. The installed base of smart phones is
moreover highly upgradable, and permits rapid adoption of out-of-band
rendezvous services for smart phones that circumvent the PSTN.
Mobile messaging services that use telephone numbers as identities
allow smart phone users to send text messages to one another over the
Internet rather than over the PSTN. Like VIPR, such services create
an out-of-band connection over the Internet between smart phones;
unlike VIPR, the rendezvous service is provided by a trusted
centralized database rather than by a DHT, and it is the centralized
database that effectively verifies and asserts the telephone number
of the sender of a message. While such messaging services are
specific to the users of the specific service, it seems clear that
similar databases could be provided by neutral third parties in a
position to coordinate between endpoints.
6.2. Failure of Public ENUM
At the time [RFC4474] was written, the hopes for establishing a
certificate authority for telephone numbers on the Internet largely
rested on public ENUM deployment. The e164.arpa DNS tree established
for ENUM could have grown to include certificates for telephone
numbers or at least for number ranges. It is now clear however that
public ENUM as originally envisioned has little prospect for
adoption. That said, some national authorities for telephone numbers
are migrating their provisioning services to the Internet, and
issuing credentials that express authority for telephone numbers to
secure those services. These new authorities for numbers could
provide to the public Internet the necessary signatory authority for
securing calling partys' numbers. While these systems are far from
universal, the authors of this draft believe that a solution devised
for the North American Numbering Plan could have applicability to
other country codes.
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6.3. Public Key Infrastructure Developments
Also, there have been a number of recent high-profile compromises of
web certificate authorities. The presence of numerous (in some
cases, of hundreds) of trusted certificate authorities in modern web
browsers has become a significant security liability. As [RFC4474]
relied on web certificate authorities, this too provides new lessons
for any work on revising [RFC4474]: namely, that innovations like
DANE [RFC6698] that designate a specific certificate preferred by the
owner of a DNS name could greatly improve the security of a SIP
identity mechanism; and moreover, that when considering new
certificate authorities for telephone numbers, we should be wary of
excessive pluralism. While a chain of delegation with a
progressively narrowing scope of authority (e.g., from a regulatory
entity to a carrier to a reseller to an end user) is needed to
reflect operational practices, there is no need to have multiple
roots, or peer entities that both claim authority for the same
telephone number or number range.
6.4. Prevalence of B2BUA Deployments
Given the prevalence of established B2BUA deployments, we may have a
further opportunity to review the elements signed by [RFC4474] and to
decide on the value of alternative signature mechanisms. Separating
the elements necessary for (a) securing the From header field value
and preventing replays, from (b) the elements necessary to prevent
men-in-the-middle from tampering with messages, may also yield a
strategy for identity that will be practicable in some highly
mediated networks. Solutions in this space must however remain
mindful of the requirements for securing cryptographic material
necessary to support DTLS-SRTP or future security mechanisms.
6.5. Stickiness of Deployed Infrastructure
One thing that has not changed, and is not likely to change in the
future, is the transitive nature of trust in the PSTN. When a call
from the PSTN arrives at a SIP gateway with a calling party number,
the gateway will have little chance of determining whether the
originator of the call was authorized to claim that calling party
number. Due to roaming and countless other factors, calls on the
PSTN may emerge from administrative domains that were not assigned
the originating number. This use case will remain the most difficult
to tackle for an identity system, and may prove beyond repair. It
does however seem that with the changes in the solution space, and a
better understanding of the limits of [RFC4474] and VIPR, we are
today in a position to reexamine the problem space and find solutions
that can have a significant impact on the secure origins problem.
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6.6. Concerns about Pervasive Monitoring
While spoofing the origins of communication is a source of numerous
security concerns, solutions for identifying communications must also
be mindful of the security risks of pervasive monitoring (see
[I-D.farrell-perpass-attack]). Identifying information, once it is
attached to communications, can potentially be inspected by parties
other than the intended recipient and collected for any number of
reasons. As stated above, the purpose of this work is not to
eliminate anonymity, but furthermore, to be viable and in the public
interest, solutions should not facilitate the unauthorized collection
of calling data.
6.7. Relationship with Number Assignment and Management
Currently, telephone numbers are typically managed in a loose
delegation hierarchy. For example, a national regulatory agency may
task a private, neutral entity with administering numbering
resources, such as area codes, and a similar entity with assigning
number blocks to carriers and other authorized entities, who in turn
then assign numbers to customers. Resellers with looser regulatory
obligations can complicate the picture, and in many cases it is
difficult to distinguish the roles of enterprises from carriers. In
many countries, individual numbers are portable between carriers, at
least within the same technology (e.g., wireline-to-wireline).
Separate databases manage the mapping of numbers to switch
identifiers, companies and textual caller ID information.
As the PSTN transitions to using VoIP technologies, new assignment
policies and management mechanisms are likely to emerge. For
example, it has been proposed that geography could play a smaller
role in number assignments, and that individual numbers are assigned
to end users directly rather than only to service providers, or that
the assignment of numbers does not depend on providing actual call
delivery services.
Databases today already map telephone numbers to entities that have
been assigned the number, e.g., through the LERG (originally, Local
Exchange Routing Guide) in the United States. Thus, the transition
to IP-based networks may offer an opportunity to integrate
cryptographic bindings between numbers or number ranges and service
providers into databases.
7. Basic Requirements
This section describes only the high level requirements of the
effort, which we expected will be further articulated as work
continues:
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Generation: Intermediaries as well as end system must be able to
generate the source identity information.
Validation: Intermediaries as well as end system must be able to
validate the source identity information.
Usability: Any validation mechanism must work without human
intervention, that is, without for exammple CAPTCHA-like
mechanisms.
Deployability: Must survive transition of the call to the PSTN and
the presence of B2BUAs.
Reflecting existing authority: Must stage credentials on existing
national-level number delegations, without assuming the need for
an international golden root on the Internet.
Accommodating current practices: Must allow number portability among
carriers and must support legitimate usage of number spoofing
(doctor's office and call centers)
Minimal payload overhead: Must lead to minimal expansion of SIP
headers fields to avoid fragmentation in deployments that use UDP.
Efficiency: Must minimize RTTs for any network lookups and minimize
any necessary cryptographic operations.
Privacy: A solution must prevent unauthorized third parties from
learning what numbers have been called by a specific caller.
Some requirements specifically outside the scope of the effort
include:
Display name: This effort does not consider how the display name of
the caller might be validated.
Response authentication: This effort only considers the problem of
providing secure telephone identity for requests, not for
responses to requests; no solution is here proposed for the
problem of determining to which number a call has connected.
8. Acknowledgments
We would like to thank Sanjay Mishra, Fernando Mousinho, David
Frankel, Penn Pfautz, Mike Hammer, Dan York, Andrew Allen, Philippe
Fouquart, Hadriel Kaplan, Richard Shockey, Russ Housley, Alissa
Cooper, Bernard Aboba, Sean Turner, Brian Rosen, Eric Burger, and
Eric Rescorla for their discussion input that lead to this document.
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9. IANA Considerations
This memo includes no request to IANA.
10. Security Considerations
This document is about improving the security of call origin
identification; security considerations for specific solutions will
be discussed in solutions documents.
11. Informative References
[I-D.cooper-iab-secure-origin]
Cooper, A., Tschofenig, H., Peterson, J., and B. Aboba,
"Secure Call Origin Identification", draft-cooper-iab-
secure-origin-00 (work in progress), November 2012.
[I-D.farrell-perpass-attack]
Farrell, S. and H. Tschofenig, "Pervasive Monitoring is an
Attack", draft-farrell-perpass-attack-06 (work in
progress), February 2014.
[I-D.jennings-vipr-overview]
Barnes, M., Jennings, C., Rosenberg, J., and M. Petit-
Huguenin, "Verification Involving PSTN Reachability:
Requirements and Architecture Overview", draft-jennings-
vipr-overview-06 (work in progress), December 2013.
[I-D.peterson-sipping-retarget]
Peterson, J., "Retargeting and Security in SIP: A
Framework and Requirements", draft-peterson-sipping-
retarget-00 (work in progress), February 2005.
[I-D.rosenberg-sip-rfc4474-concerns]
Rosenberg, J., "Concerns around the Applicability of RFC
4474", draft-rosenberg-sip-rfc4474-concerns-00 (work in
progress), February 2008.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263, June
2002.
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[RFC3324] Watson, M., "Short Term Requirements for Network Asserted
Identity", RFC 3324, November 2002.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
November 2002.
[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC
3966, December 2004.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[RFC5039] Rosenberg, J. and C. Jennings, "The Session Initiation
Protocol (SIP) and Spam", RFC 5039, January 2008.
[RFC5727] Peterson, J., Jennings, C., and R. Sparks, "Change Process
for the Session Initiation Protocol (SIP) and the Real-
time Applications and Infrastructure Area", BCP 67, RFC
5727, March 2010.
[RFC6698] Hoffman, P. and J. Schlyter, "The DNS-Based Authentication
of Named Entities (DANE) Transport Layer Security (TLS)
Protocol: TLSA", RFC 6698, August 2012.
[TDOS] Krebs, B., "DHS Warns of 'TDoS' Extortion Attacks on
Public Emergency Networks", URL:
http://krebsonsecurity.com/2013/04/dhs-warns-of-tdos-
extortion-attacks-on-public-emergency-networks/, Apr 2013.
[news-hack]
Wikipedia, , "News International phone hacking scandal",
URL: http://en.wikipedia.org/wiki/
News_International_phone_hacking_scandal, Apr 2013.
[robocall-competition]
FTC, , "FTC Robocall Challenge", URL:
http://robocall.challenge.gov/, Apr 2013.
[robocall-fcc]
FCC, , "Robocalls", URL:
http://www.fcc.gov/guides/robocalls, Apr 2013.
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[swatting]
Wikipedia, , "Don't Make the Call: The New Phenomenon of
'Swatting'", URL: http://www.fbi.gov/news/stories/2008/
february/swatting020408, Feb 2008.
Authors' Addresses
Jon Peterson
Neustar, Inc.
1800 Sutter St Suite 570
Concord, CA 94520
US
Email: jon.peterson@neustar.biz
Henning Schulzrinne
Columbia University
Department of Computer Science
450 Computer Science Building
New York, NY 10027
US
Phone: +1 212 939 7004
Email: hgs@cs.columbia.edu
URI: http://www.cs.columbia.edu
Hannes Tschofenig
Nokia Siemens Networks
Linnoitustie 6
Espoo 02600
Finland
Phone: +358 (50) 4871445
Email: Hannes.Tschofenig@gmx.net
URI: http://www.tschofenig.priv.at
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