Secure Origin Identification: Problem Statement, Requirements, and Roadmap
draft-peterson-secure-origin-ps-00
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft whose latest revision state is "Replaced".
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|
|---|---|---|---|
| Authors | Jon Peterson , Henning Schulzrinne , Hannes Tschofenig | ||
| Last updated | 2013-05-27 | ||
| Replaced by | draft-ietf-stir-problem-statement, draft-ietf-stir-problem-statement, RFC 7340 | ||
| RFC stream | (None) | ||
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| Consensus boilerplate | Unknown | ||
| RFC Editor Note | (None) | ||
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draft-peterson-secure-origin-ps-00
Network Working Group J. Peterson
Internet-Draft NeuStar, Inc.
Intended status: Informational H. Schulzrinne
Expires: November 28, 2013 Columbia University
H. Tschofenig
Nokia Siemens Networks
May 27, 2013
Secure Origin Identification: Problem Statement, Requirements, and
Roadmap
draft-peterson-secure-origin-ps-00.txt
Abstract
Over the past decade, SIP has become a major signaling protocol for
voice communications, one which has replaced many traditional
telephony deployments. However, interworking SIP with the
traditional telephone network has ultimately reduced the security of
Caller ID systems. Given the widespread interworking of SIP with the
telephone network, the lack of effective standards for identifying
the calling party in a SIP session has granted attackers new powers
as they 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. This document therefore 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.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 28, 2013.
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Copyright Notice
Copyright (c) 2013 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 . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. VoIP-to-VoIP Call . . . . . . . . . . . . . . . . . . . . 5
3.2. IP-PSTN-IP Call . . . . . . . . . . . . . . . . . . . . . 6
3.3. PSTN-to-VoIP Call . . . . . . . . . . . . . . . . . . . . 7
3.4. VoIP-to-PSTN Call Call . . . . . . . . . . . . . . . . . 8
3.5. PSTN-VoIP-PSTN Call . . . . . . . . . . . . . . . . . . . 9
3.6. PSTN-to-PSTN Call . . . . . . . . . . . . . . . . . . . . 10
4. Limitations of Current Solutions . . . . . . . . . . . . . . 10
4.1. SIP Identity . . . . . . . . . . . . . . . . . . . . . . 11
4.2. VIPR . . . . . . . . . . . . . . . . . . . . . . . . . . 14
5. Environmental Changes . . . . . . . . . . . . . . . . . . . . 15
5.1. Shift to Mobile Communication . . . . . . . . . . . . . . 16
5.2. Failure of Public ENUM . . . . . . . . . . . . . . . . . 16
5.3. Public Key Infrastructure Developments . . . . . . . . . 16
5.4. Pervasive Nature of B2BUA Deployments . . . . . . . . . . 17
5.5. Stickiness of Deployed Infrastructure . . . . . . . . . . 17
5.6. Relationship with Number Assignment and Management . . . 18
6. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 18
7. Roadmap . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 19
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19
10. Security Considerations . . . . . . . . . . . . . . . . . . . 20
11. Informative References . . . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 21
1. Introduction
In many communication architectures that allow users to communicate
with other users the need for identifying the originating party that
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initiates a call or a messaging interaction arises. The desire for
identifying the communication parties in the end-to-end communication
attempt arises 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 4. The goal
of these mechanisms is to validate that originator of a call is
authorized to use the From identifier. Protocols, like XMPP, use
mechanisms that are conceptional 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 improve in the future. In
[I-D.cooper-iab-secure-origin] we illustrate what challenges arise.
In particular, the interworking with different communication
architectures (e.g., SIP, PSTN, XMPP, RTCWeb) breaks the end-to-end
semantic of the communication interaction and destroys the
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.
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 an attempt to develop a common understanding of the problem
statement as well as 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.
2. Problem Statement
In the classical public-switched telephone network, a limited number
of carriers trusted each other, without any cryptographic validation,
to provide accurate caller origination information. 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., through the concept of certificated
carriers) 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
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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. 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 FCC webpage
[robocall] on this topic). Robocalls is a problem that has been
recognized already by various regulators, for example the Federal
Communications Commission (FCC) 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
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 blacklist numbers. It also makes tracing such calls
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much more labor-intensive, as each such call has to be identified in
each transit carrier hop-by-hop, based on destination number and time
of call.
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.
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. 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.
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. 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.
3.1. 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 below. These service providers use
telephone numbers as source and destination 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.)
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+------------+
| IP-based |
| SIP Phone |<--+
| of Bob | |
|+19175551234| |
+------------+ |
|
+------------+ |
| IP-based | |
| SIP Phone | ------------
| of Alice | / | \
|+12121234567| // | \\
+------------+ // ,' \\\
| /// / -----
| //// ,' \\\\
| / ,' \
| | ,' |
+---->|......: IP-based |
| Network |
\ /
\\\\ ////
-------------------------
Figure 1: VoIP-to-VoIP Call.
3.2. 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.
--------
//// \\\\
+--- >| PSTN |
| | |
| \\\\ ////
| --------
| |
| |
| |
+------------+ +--+----+ |
| IP-based | | PSTN | |
| SIP Phone | --+ VoIP +- v
| of Alice | / | GW | \ +---+---+
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|+12121234567| // `''''''' \\| PSTN |
+------------+ // | \+ VoIP +
| /// | | GW |\
| //// | `'''''''\\ +------------+
| / | | \ | IP-based |
| | | | | | Phone |
+---->|---------------+ +------|---->| of Dan |
| | |+12039994321|
\ IP-based / +------------+
\\\\ Network ////
-------------------------
Figure 2: IP-PSTN-IP Call.
3.3. 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 of 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 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.
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.
3.4. VoIP-to-PSTN Call 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 needs to 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. Ideally, Alice would like to know whether she,
for example, talks to someone at her bank rather than to someone
intercepting the call. If Alice wants to be assured that she's being
connected to the right party, it is a slightly different aspect to
what [RFC3325][RFC4474]. Problem statements and solutions are
offered with [I-D.peterson-sipping-retarget] and [RFC4916].
+-------+ +-----+ -C
|PSTN | |SIP | |a
|Phone |<----------------+ |UA | |l
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|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.
3.5. 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. Consequenly, 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.
+----------+
|PSTN Phone|
-------- |of Alice |
//// \\\\ +----------+
+->| PSTN |------+ ^
| | | | |
| \\\\ //// | |
| -------- | --------
| v //// \\\\
| ,-------+ | PSTN |
| |PSTN | | |
+---+------+ __|VoIP GW|_ \\\\ ////
|PSTN Phone| / '`''''''' \ --------
|of Carl | // | \\ ^
+----------+ // | \\\ |
/// -. Invite ----- |
//// `-. \\\\ |
/ `.. \ |
| IP-based `._ ,--+----+
| Network `.....>|VoIP |
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| |PSTN GW|
\ '`'''''''
\\\\ ////
-------------------------
Figure 5: PSTN-VoIP-PSTN Call.
3.6. 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.
4. 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 end users to set
the P-Asserted-Identity information.
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 both to prevent originating calls
with spoofed From headers and intermediaries, such as SIP proxies,
from launching man-in-the-middle attacks to alter calls passing
through. 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.
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4.1. 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 web 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.
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The SIP Identity mechanism, however, has two fundamental limitations
that have precluded its deployment: first, that it provides Identity
only for domain name rather than other identifiers; second, that it
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. In general, the
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 resigns 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
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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.
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 enforce policy at
network boundaries (hence the nickname "Session Border Controller").
As a common practice, these entities modify SIP INVITE requests in
transit in such a way that they no longer satisfy the transaction-
mapping semantics of [RFC3261], commonly 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. The policies that motivate these changes may be associated
with topology hiding, or may alter messages to interoperate
successfully with particular SIP implementations, or may simply
involve network address translation from private address space. But
effectively, 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 not violate (or remove) the 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
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example, make it any clearer how a PSTN gateway should assert
identity in an INVITE request.
4.2. 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 nonce 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.
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Due to its narrow scope of applicability, and the details of its
implementation, VIPR has some significant limitations. The most
salient for the purposes of this document is that it only has bearing
on repeated communications between entities: it has no bearing on the
classic "robocall" problem, where the target receives a call from a
number that has never called before. All of VIPRs 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 is rooted in 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.
5. Environmental Changes
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5.1. 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 out-of-band
rendezvous services for smart phones that circumvent the PSTN: for
example, the Apple iMessage service, which allows iPhone users to
send SMS messages to one another over the Internet rather than over
the PSTN. Like VIPR, iMessage creates an out-of-band connection over
the Internet between iPhones; unlike VIPR, the rendezvous service is
provided by a trusted centralized database of iPhones rather than by
a DHT. While Apple's service is specific to customers of its smart
phones, it seems clear that similar databases could be provided by
neutral third parties in a position to coordinate between endpoints.
5.2. 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, national authorities for telephone numbers are
increasingly migrating their provisioning services to the Internet,
and issuing credentials that express authority for telephone numbers
to secure those services. This new class of certificate authority
for numbers could be opened to the public Internet to provide the
necessary signatory authority for securing calling partys' numbers.
While these systems are far from universal, the authors of this draft
believe a certificate authority can be erected for the North American
Numbering Plan in a way that numbering authorities for other country
codes could follow.
5.3. 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
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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 architecting 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.
5.4. Pervasive Nature 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. The ongoing
efforts in the STRAW working group
[I-D.ietf-straw-b2bua-loop-detection] provide one possibility worth
investigating for changes to the signed elements. 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. It could be possible, for example, to provide two
signatures: one over the elements required for (b), and then a
separate signature over the elements necessary for (a) and the
signature over (b); this would allow verification services in
mediated networks to ignore the failure of a (b) signature while
still verifying (a). Any solution along these lines must however
always secure any cryptographic material necessary to support DTLS-
SRTP or future security mechanisms.
5.5. 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 have no relationship
with the number assignee. 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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5.6. 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. 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.
6. Requirements
This section describes the high level requirements:
Usability Any validation mechanism must work without human
intervention, e.g., CAPTCHA-like mechanisms.
Deployability Must survive transition of the call to the PSTN and
the presence of B2BUAs.
Validation by intermediaries Intermediaries as well as end system
must be able to validate the source identity information.
Display name The display name of the caller must also be validated
or the callee must be able to determine that only the calling
number has been validated.
Consider existing structures must allow number portability among
carriers and must support legitimate usage of number spoofing
(doctor's office and call centers)
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Minimal payload overhead Must lead to minimal expansion of SIP
headers fields to avoid fragmentation in deployments that use UDP.
Privacy Any out-of-band validation protocol must not allows third
parties to learn what numbers have been called by a specific
caller.
7. Roadmap
The authors of this document believe that the entire solution scope
consists of a couple of separable aspects:
In-band caller ID Conveyance: This functionality allows call origin
identification information to be conveyed within SIP, and takes
the nature of E.164 numbers and the prevalence of B2BUAs into
account. This may consist of a revised version of the SIP
Identity specification that takes E.164 numbers into account and
allows for separate validation of the SIP request headers and the
SIP request body. This approach addresses the case where
intermediaries do not remove header fields.
Out-of-Band Caller-ID Verification: This functionality determines
whether the E.164 number used by the calling party actually
exists, 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 when the in-band technique does not work
due to intermediaries or due to interworking with PSTN networks.
Certificate Delegation Infrastructure: This functionality defines
how certificates with E.164 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.
Extended Validation: This functionality describes how to describes
attributes of the calling party beyond the caller-id and these
attributes (e.g., the calling party is a bank) need to be verified
upfront.
8. Acknowledgments
We would like to thank Alissa Cooper, Bernard Aboba, Sean Turner,
Eric Burger, and Eric Rescorla for their discussion input that lead
to this document.
9. IANA Considerations
This memo includes no request to IANA.
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10. Security Considerations
This document is about improving the security of call origin
identification.
11. Informative References
[10] Rosenberg, J., "Concerns around the Applicability of RFC
4474", draft-rosenberg-sip-rfc4474-concerns-00 (work in
progress), February 2008.
[11] Kaplan, H. and V. Pascual, "Loop Detection Mechanisms for
Session Initiation Protocol (SIP) Back-to- Back User
Agents (B2BUAs)", draft-ietf-straw-b2bua-loop-detection-00
(work in progress), April 2013.
[12] Barnes, M., Jennings, C., Rosenberg, J., and M. Petit-
Huguenin, "Verification Involving PSTN Reachability:
Requirements and Architecture Overview", draft-jennings-
vipr-overview-04 (work in progress), February 2013.
[13] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263, June
2002.
[14] 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.
[15] FCC, , "Robocalls", URL:
http://www.fcc.gov/guides/robocalls, Apr 2013.
[16] FCC, , "FCC Robocall Challenge", URL:
http://robocall.challenge.gov/, Apr 2013.
[17] Wikipedia, , "News International phone hacking scandal",
URL: http://en.wikipedia.org/wiki/
News_International_phone_hacking_scandal, Apr 2013.
[18] Wikipedia, , "Don't Make the Call: The New Phenomenon of
'Swatting'", URL: http://www.fbi.gov/news/stories/2008/
february/swatting020408, Feb 2008.
[1] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
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[2] 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.
[3] 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.
[4] 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.
[5] Hoffman, P. and J. Schlyter, "The DNS-Based Authentication
of Named Entities (DANE) Transport Layer Security (TLS)
Protocol: TLSA", RFC 6698, August 2012.
[6] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, June 2007.
[7] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC
3966, December 2004.
[8] Cooper, A., Tschofenig, H., Peterson, J., and B. Aboba,
"Secure Call Origin Identification", draft-cooper-iab-
secure-origin-00 (work in progress), November 2012.
[9] Peterson, J., "Retargeting and Security in SIP: A
Framework and Requirements", draft-peterson-sipping-
retarget-00 (work in progress), February 2005.
Authors' Addresses
Jon Peterson
NeuStar, Inc.
1800 Sutter St Suite 570
Concord, CA 94520
US
Email: jon.peterson@neustar.biz
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Henning Schulzrinne
Columbia University
Department of Computer Science
450 Computer Science Building
New York, NY 10027
US
Phone: +1 212 939 7004
Email: hgs+ecrit@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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