Network Working Group A. Cooper
Internet-Draft CDT
Intended status: Informational H. Tschofenig
Expires: June 3, 2013 Nokia Siemens Networks
J. Peterson
NeuStar
B. Aboba
Microsoft
November 30, 2012
Secure Call Origin Identification
draft-cooper-iab-secure-origin-00.txt
Abstract
A number of parties have suggested creating mandates such that
networks receiving voice calls would be capable of securely
identifying the call origin. This document provides insights about
the capabilities and limitations of supporting call origin
identification in a secure and privacy- friendly way in the PSTN and
for IP-based real-time communications.
Status of this Memo
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Secure Origin Challenges in the PSTN . . . . . . . . . . . . . 3
3. Secure Origin Challenges for VoIP . . . . . . . . . . . . . . . 4
4. Secure Origin Challenges for Real-Time Communication on
the Web . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
5. Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . . 7
7. Informative References . . . . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 7
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1. Introduction
A number of parties have suggested creating mandates such that
networks receiving voice calls would be capable of securely
identifying the call origin [TD-62]. These proposals are primarily
motivated by concerns over fraudulent calls and the associated
economic impact that spoofed or fraudulent origin identification can
have on telecommunications settlement agreements. Concerns have also
been raised about ensuring secure origin identification for law
enforcement and abuse tracking purposes.
Support for caller identification in the public switched telephone
network (PSTN) has been developed to meet existing regulatory needs
and for other purposes, but it has limitations. As real-time
communication has become IP-based, it has become significantly more
difficult to identify the origin of real-time communication for a
number of reasons. Furthermore, to the extent that new mandates are
being suggested to require origin identification for all IP traffic
-- real-time or not -- it is unclear how such identification would be
accomplished given the complexity and diversity of today's IP network
traffic. This document provides insights about the capabilities and
limitations of supporting call origin identification in a secure and
privacy-friendly way in the PSTN and for IP-based real-time
communications.
2. Secure Origin Challenges in the PSTN
The problems facing origin identification are not limited to the
Internet. In the traditional public switched telephone network,
information about the calling party is often missing from signaling
messages. This is because, per the ITU-T and related national
specifications (beginning with Q.761), the calling party number field
of a call establishment message (IAM) is an optional parameter.
There remain a number of legitimate reasons today why the origin of a
telephone call might be absent from call establishment signaling:
because of interworking with a legacy (pre-SS7) network or a private
branch exchange which lacks the capability to identify subscribers,
for example, or because the call has been handled by an interexchange
carrier using calling cards that require the customer to dial a relay
number. So long as there are legitimate reasons why the calling
party number might be missing from a call, and the parameter remains
optional in the SS7 standards, carriers will sometimes fail to
provide origin identification in the telephone network. Parties who
want to obscure their identity can rely on equipment or carriers that
do not provide the calling party number.
Whether or not the calling party number should be trusted, if it is
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present, is a separate but related question. Private branch
exchanges that signal calls via ISDN (Q.931) can provide an arbitrary
calling party number in their own call establishment message (SETUP),
and some operators translate those numbers directly to the calling
party number field of SS7. Due to the transitive trust inherent in
the SS7 network, there is no way for the recipient of a call to
determine how trustworthy the calling party number field is:
effectively, all carriers in the telephone network necessarily trust
the origin identification provided by any carrier. These
difficulties have been exacerbated by the widespread deployment of
Internet gateways. For these gateways, it is almost always better to
supply no calling party number to the SS7 network than it is to
accept a number provided by Internet signaling. Because some
gateways do accept the numbers provided by Internet callers, however,
this further weakens the trustworthiness of calling party number
information on the telephone network. These concerns are not limited
to calls either: text messages have similar problems resulting from
email-to-text gateways.
Our ability to solve origin identification for Internet calling
depends on solving it for the telephone network, as Internet
telephony solutions inevitably exchange traffic with the telephone
network. Given the inherent limitations of SS7 standards for origin
identification, the transitive trust properties of the telephone
network, and the widespread acceptance of calls without origin
identification in the telephone system today, the prospects are very
doubtful for remedying this problem by simply mandating that carriers
provide origin identification.
3. Secure Origin Challenges for VoIP
Standards for Voice Over IP (VoIP) originally focused on Session
Initiation Protocol (SIP) and Extensible Messaging and Presence
Protocol (XMPP)-based systems, with SIP becoming a popular foundation
for many proprietary VoIP systems. SIP provides a number of
different mechanisms for asserting the identity of a caller,
including "P-Asserted-Identity" (PAI) [RFC3325], "SIP Cert" [RFC6072]
and the "SIP Identity" mechanism [RFC4474]. PAI and SIP Identity
allow SIP application servers in the network to insert caller
information into the 'From' header of outgoing calls.
However, not all calls will necessarily identify the calling party in
any way, and there are no standardized requirements to use any
particular caller identification solution. Anonymous calls and calls
made from outbound-only calling services generally do not contain
identity information. In addition, in some situations, calls made
through relay services may identify the relay as the calling party
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rather than the original caller.
In addition, the identifier syntax used in SIP varies from email-
address-style identifiers to ones that use E.164 telephone numbers.
Because of the security shortcomings of the PSTN described above,
SIP-based services that seek to make authentication and
identification guarantees cannot do so purely with E.164 numbers.
Such guarantees would require a universal move toward email-style
identifiers.
Even in a SIP-only environment, the choice of syntax, made separately
by different implementers and users, impacts the security mechanisms
that can be used for attesting to the authenticity of the identifier.
Without any form of cryptographic identity assertion, the 'From'
header can be easily forged, and headers are often stripped or
modified by intermediaries in transit. Attempts at enhancing the
integrity protection of SIP identity have not seen wide deployment.
Finally, SIP supports a number of privacy mechanisms that allow SIP
users to shield their identities from the network and the called
party [RFC3323] [RFC5767] [RFC5379]. SIP privacy has valid uses; for
example, it enables users to avoid exposing their identity to
destinations that might make them a target for unsolicited
advertising or other undesirable consequences. As in the PSTN,
parties that do not wish to disclose their identities can use
services that support this functionality.
4. Secure Origin Challenges for Real-Time Communication on the Web
While SIP and XMPP were originally designed to facilitate
interoperable real-time communications between systems developed by
different vendors, the last 10 years have seen the rise of a new
platform for deployment of applications: the world wide web. Web
applications that run in a secure execution environment inside a web
browser are now commonplace. As a result, real-time communications
-- including voice and video telephony -- are migrating to the web as
well.
This new trend in application development enables real-time
application to be downloaded on-demand and executed within the
browser utilizing new interfaces in the browser platform to interact
with the microphone, camera, and other sensors. The integration of
real-time communication into the huge web ecosystem opens a number of
new possibilities that were previously only possible with proprietary
browser plug-ins.
This migration to the web does not eliminate the challenges
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associated with providing secure call origin identification; in some
ways, it even serves to complicate the situation. While web servers
have a common and widely used means of authenticating themselves --
public key-based authentication infrastructure for use with SSL -- no
single client-side authentication mechanism has emerged to
authenticate users sitting in front of their browsers. Instead, a
variety of technologies have been deployed, often with applicability
only in narrow sectors.
Enterprise networks, for example, often use hardware tokens to
authenticate employees. Banking web sites often require one-time
secrets in combination with knowledge-based security. Many consumer-
focused web sites tend to rely on insecure password-based
authentication. With so many web sites requiring authentication, Web
Single-Sign-On (WebSSO) deployments have emerged to attempt to create
a uniform authentication mechanism. But while individual identity
providers offering such WebSSO solutions offer security benefits and
great convenience for end users, they too are typically limited as
far as the scope of web sites that can rely on them for
authentication. Relying web sites, likewise, usually only support a
single or limited set of identity providers. Consequently, users are
confronted with islands on the web that use different identity
technologies.
Because web-based real-time applications extends the existing web
ecosystem, they also inherit its identity management ecosystem, a
system that is still in flux. With new technology being developed
every day it is unlikely that a single identity management technology
will dominate in the near future. Furthermore, because of the
technological differences between web identity management and caller
identification in SIP and other previously developed real-time
communication technologies, the seamless flow of identity information
across these technologies will likely remain elusive for the
foreseeable future.
5. Conclusion
Every calling technology presents significant challenges to the
secure identification of the caller. The interoperation of all of
these technologies -- from legacy pre-SS7 telephone networks to
cutting edge web-based calling services -- further complicates the
task of identifying the origin of a call in a trusted and
interoperable way. While industry efforts are underway to address
some of these challenges, a uniform origin identification system is
unlikely to emerge, regardless of potential regulatory mandates.
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6. Security Considerations
This document describes, at a high level, some of the security
challenges of providing trustworthy call origin information. There
are further detailed privacy and security aspects related to call
origin identification that will be addressed in a future version of
this document.
7. Informative References
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Initiation Protocol (SIP)", RFC 3323, 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.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC5379] Munakata, M., Schubert, S., and T. Ohba, "Guidelines for
Using the Privacy Mechanism for SIP", RFC 5379,
February 2010.
[RFC5767] Munakata, M., Schubert, S., and T. Ohba, "User-Agent-
Driven Privacy Mechanism for SIP", RFC 5767, April 2010.
[RFC6072] Jennings, C. and J. Fischl, "Certificate Management
Service for the Session Initiation Protocol (SIP)",
RFC 6072, February 2011.
[TD-62] Council Working Group to Prepare for the 2012 World
Conference on International Telecommunications, "CWG-
WCIT12 Temporary Document 62 Rev.2 - Draft Compilation of
Proposals with Options for Revisions to the ITRs", 2012, <
http://files.wcitleaks.org/public/
T09-CWG.WCIT12-120620-TD-PLEN-0062R2.pdf>.
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Authors' Addresses
Alissa Cooper
CDT
1634 Eye St. NW, Suite 1100
Washington, DC 20006
USA
Email: acooper@cdt.org
Hannes Tschofenig
Nokia Siemens Networks
Email: hannes.tschofenig@gmx.net
Jon Peterson
NeuStar
Email: jon.peterson@neustar.biz
Bernard Aboba
Microsoft
Email: bernard.aboba@gmail.com
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