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Versions: 00 01 02 draft-ietf-stir-problem-statement

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
   (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  . . . . . . . . . . . . . . . . . . . . . . . .   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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