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Versions: (draft-peterson-secure-origin-ps) 00 01 02 03 04 05 RFC 7340

Network Working Group                                        J. Peterson
Internet-Draft                                             NeuStar, Inc.
Intended status: Informational                            H. Schulzrinne
Expires: November 10, 2014                           Columbia University
                                                           H. Tschofenig

                                                             May 9, 2014

      Secure Telephone Identity Problem Statement and Requirements


   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

   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
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   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
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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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   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

   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

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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

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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

                        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

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

   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

   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

   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

   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

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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

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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

   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

   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

              Cooper, A., Tschofenig, H., Peterson, J., and B. Aboba,
              "Secure Call Origin Identification", draft-cooper-iab-
              secure-origin-00 (work in progress), November 2012.

              Farrell, S. and H. Tschofenig, "Pervasive Monitoring is an
              Attack", draft-farrell-perpass-attack-06 (work in
              progress), February 2014.

              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.

              Peterson, J., "Retargeting and Security in SIP: A
              Framework and Requirements", draft-peterson-sipping-
              retarget-00 (work in progress), February 2005.

              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

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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:
              extortion-attacks-on-public-emergency-networks/, Apr 2013.

              Wikipedia, , "News International phone hacking scandal",
              URL: http://en.wikipedia.org/wiki/
              News_International_phone_hacking_scandal, Apr 2013.

              FTC, , "FTC Robocall Challenge", URL:
              http://robocall.challenge.gov/, Apr 2013.

              FCC, , "Robocalls", URL:
              http://www.fcc.gov/guides/robocalls, Apr 2013.

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              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

   Email: jon.peterson@neustar.biz

   Henning Schulzrinne
   Columbia University
   Department of Computer Science
   450 Computer Science Building
   New York, NY  10027

   Phone: +1 212 939 7004
   Email: hgs@cs.columbia.edu
   URI:   http://www.cs.columbia.edu

   Hannes Tschofenig
   Hall, Tirol  6060

   Email: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at

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