VIPR WG                                                   M. Barnes, Ed.
Internet-Draft                                                   Polycom
Intended status:  Standards Track                            C. Jennings
Expires:  August 29, 2013                                          Cisco
                                                            J. Rosenberg
                                                       M. Petit-Huguenin
                                                       February 25, 2013

Verification Involving PSTN Reachability: Requirements and Architecture


   The Session Initiation Protocol (SIP) has seen widespread deployment
   within individual domains, typically supporting voice and video
   communications.  Though it was designed from the outset to support
   inter-domain federation over the public Internet, such federation has
   not materialized.  The primary reasons for this are the complexities
   of inter-domain phone number routing and concerns over security.
   This document reviews this problem space, outlines requirements, and
   then describes a new model and technique for inter-domain federation
   with SIP involving the Public Switched Telephone Network (PSTN),
   called Verification Involving PSTN Reachability (VIPR).  VIPR
   addresses the problems that have prevented inter-domain federation
   over the Internet.  It provides fully distributed inter-domain
   routing for phone numbers, authorized mappings from phone numbers to
   domains, a new technique for automated SIP anti-spam, and privacy of
   number ownership, all while preserving the trapezoidal model of SIP.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

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   material or to cite them other than as "work in progress."

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   This Internet-Draft will expire on August 29, 2013.

Copyright Notice

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   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   publication of this document.  Please review these documents
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Conventions and Terminology  . . . . . . . . . . . . . . . . .  4
   3.  Problem Statement  . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  The Phone Number Routing Problem . . . . . . . . . . . . .  5
     3.2.  The Open Pinhole Problem . . . . . . . . . . . . . . . . .  6
     3.3.  Quality of Service Problem . . . . . . . . . . . . . . . .  7
     3.4.  Troubleshooting Problem  . . . . . . . . . . . . . . . . .  7
   4.  Summary of Existing Solutions  . . . . . . . . . . . . . . . .  8
     4.1.  Domain Routing . . . . . . . . . . . . . . . . . . . . . .  8
     4.2.  Public ENUM  . . . . . . . . . . . . . . . . . . . . . . .  8
     4.3.  Private Federations  . . . . . . . . . . . . . . . . . . .  9
   5.  Key Requirements . . . . . . . . . . . . . . . . . . . . . . . 10
   6.  Executive Overview . . . . . . . . . . . . . . . . . . . . . . 10
     6.1.  Key Properties . . . . . . . . . . . . . . . . . . . . . . 11
     6.2.  Challenging Past Assumptions . . . . . . . . . . . . . . . 12
     6.3.  Technical Overview . . . . . . . . . . . . . . . . . . . . 13
       6.3.1.  Storage of Phone Numbers . . . . . . . . . . . . . . . 14
       6.3.2.  PSTN First Call  . . . . . . . . . . . . . . . . . . . 15
       6.3.3.  Validation and Caching . . . . . . . . . . . . . . . . 16
       6.3.4.  SIP Call . . . . . . . . . . . . . . . . . . . . . . . 20
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
     7.1.  Attacks on the DHT . . . . . . . . . . . . . . . . . . . . 21
     7.2.  Theft of Phone Numbers . . . . . . . . . . . . . . . . . . 21
     7.3.  Spam . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     7.4.  Eavesdropping  . . . . . . . . . . . . . . . . . . . . . . 23
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 23
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 24
     10.2. Informative References . . . . . . . . . . . . . . . . . . 25
   Appendix A.  Changes since last version  . . . . . . . . . . . . . 26
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27

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1.  Introduction

   The Session Initiation Protocol (SIP) was originally published as
   [RFC2543] in May of 1999.  This was followed by subsequent
   publication of [RFC3261], which brought the protocol to sufficient
   maturity to enable large scale market adoption.

   SIP has achieved large scale market adoption with hundreds of
   implementations, spanning consumer products, enterprise servers, and
   large scale carrier equipment.  It carries billions and billions of
   minutes of calls, and has become the standard for interconnection
   between products from different vendors.  If one measures success in
   deployment, then clearly SIP is a success.

   SIP was designed from the ground up to enable communications between
   users in different domains, all over the public Internet.  The
   intention was that real-time communications should be no different
   than email or the web, with the same any-to-any connectivity that has
   fueled the successes of those technologies.  However, when SIP is
   used between domains, it is typically through private federation
   agreements.  While such agreements are positive, they have typically
   been limited to voice, which has limited the use of video and the
   growth of advanced SIP features, thus preventing the innovation that
   SIP was expected to drive.  Thus, the any-to-any Internet federation
   model envisioned by SIP has not materialized at scale.

   This document introduces a new technology, called Verification
   Involving PSTN Reachability (VIPR), that breaks down the barriers
   that have prevented inter-domain voice, video and other multimedia
   services.  By stepping back and changing some of the most fundamental
   assumptions about federation, VIPR is able to address the key
   problems preventing its deployment.  VIPR focuses on incremental
   deployability.  At the same time, VIPR ensures that SIP's trapezoidal
   model of direct federation between domains without any intermediate
   processing beyond IP transport is realized.  That model is required
   in order to allow innovative new services to be deployed.

2.  Conventions and Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in

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   Call Agent:    An entity in a SIP enabled domain that supports VIPR.
      The Call Agent performs call processing on behalf of one or more
      user agents represented by E.164 numbers within the domain.
   Ticket:    A shared secret that is generated after a PSTN call to
      enable secure call setup on a subsequent inter-domain IP call
      enabled by VIPR.
   User Agent:    As defined in [RFC3261], with the restriction that the
      user agent must have an associated E.164 number.

3.  Problem Statement

   The first question that must be asked is this - why haven't we seen
   widespread adoption of inter-domain SIP federation?  The reason for
   this is due to problems with the following - summarized in order of

   1.  Phone number routing
   2.  Open pinhole
   3.  Quality of service
   4.  Troubleshooting

   The first two are the most significant.

3.1.  The Phone Number Routing Problem

   Inter-domain federation requires that the sending domain determine
   the address of the receiving domain, in the form of a DNS name
   ( or one or more IP addresses that can be used to reach
   the domain.  In email and in the web, this is easy.  The identifiers
   used by those services - the email address and web URL respectively -
   embed the address of the receiving domain.  A simple DNS lookup is
   all that is required to route the connection.  SIP was designed to
   use the same email-style identifiers.

   However, most SIP deployments utilize phone numbers in the form of
   E.164 numbers [E.164], and not email-style SIP URIs.  This is due to
   the huge installed base of users that continue to exist solely on the
   PSTN.  In order to be reached by users on the PSTN, and in order to
   reach them, users in SIP deployments need to be assigned a PSTN phone
   number.  Users in SIP deployments need to place that phone number on
   business cards, use it in their email signatures, and in general,
   give it out to their friends and colleagues, in order to be reached.
   While those users could additionally have an email style SIP URI, the
   phone number serves as a single, global identifier that works for
   receiving calls from users on the PSTN as well as users within the
   same SIP domain.

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   There are several reasons why two identifiers are used when one will
   suffice.  The universality of PSTN phone numbers is the reason why
   most SIP deployments continue to use them - often exclusively.

   Another reason is that many SIP deployments utilize hardphones or
   telephony adaptors, and the user interfaces on these devices -
   patterned after existing phones - only allow phone number based
   dialing.  Consequently, these users are only allocated PSTN phone
   numbers, and not email-style SIP URI.

   Finally, a large number of SIP deployments are in domains where the
   endpoints are not IP.  Rather, they are circuit based devices,
   connected to a SIP network through a gateway.  SIP is used within the
   core of the network, providing lower cost transit, or providing
   add-on services.  Clearly, in these deployments, only phone numbers
   are used.

   Consequently, to make inter-domain federation incrementally
   deployable and widely applicable, it needs to work with PSTN phone
   numbers rather than email-style SIP URIs.  Telephone numbers, unlike
   email addresses, do not provide any indication of the address of the
   domain which "owns" the phone number.  Indeed, the notion of phone
   number ownership is somewhat cloudy.  Phone numbers can be ported
   between carriers.  They can be assigned to a user or enterprise, and
   then later re-assigned to someone else.  Phone numbers are granted to
   users and enterprises through a complex delegation process involving
   the ITU, governments, and telecommunications carriers, often
   involving local regulations that vary from country to country.

   Therefore, in order to deploy inter-domain federation, domains are
   required to utilize some kind of mechanism to map phone numbers to
   the address of the domain to which calls should be routed.  Though
   several techniques have been developed to address this issue, none
   have achieved large-scale Internet deployments.

3.2.  The Open Pinhole Problem

   The inter-domain federation mechanism built into SIP borrows heavily
   from email.  Each domain runs a SIP server on an open port.  When one
   domain wishes to contact another, it looks up the domain name in the
   DNS, and connects to that server on the open port.  Here, "open"
   means that the server is reachable from anywhere on the public
   Internet, and is not blocked by firewalls.

   This simple design worked well in the early days of email.  However,
   the email system has now become plagued with spam.  This has resulted
   in administrators spending a significant amount of time maintaining
   spam filters.  This does not always benefit the end users as in some

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   cases valid emails are dropped without the user being notified.
   Thus, administrators of SIP domains are rightfully concerned that if
   they make a SIP server available for anyone on the Internet to
   contact, it will open the floodgates for SIP spam, which is far more
   disruptive than email-based spam [RFC5039].  Administrators are also
   concerned that an open server will create a back-door for denial-of-
   service and other attacks that can potentially disrupt their voice
   and video services.  Administrators are often not willing to take
   that risk since voice deployments demand higher uptimes and better
   levels of reliability than email, especially for enterprises.

   Fears around spam and denial-of-service attacks, when put together,
   form the "open pinhole problem" - that domains are not willing to
   enable SIP on an open port facing the Internet.

   To fix this, a new model for federation is needed - a model where
   these problems are addressed as part of the fundamental design rather
   than after the functionality has been deployed.

3.3.  Quality of Service Problem

   The Internet does not provide any Quality of Service (QoS)
   guarantees.  All traffic is best effort.  This is not an issue for
   data transaction services, like web and email.  It is, however, a
   concern when using real-time services, such as voice and video.

   That said, there are a large number of existing SIP deployments that
   run over the Internet.  Though the lack of QoS is a concern, it has
   not proven a barrier to deployment.  It is believed that if if the
   more fundamental issues - the phone number routing and open pinhole
   problems - can be addressed, the QoS problem will be a non-issue.  As
   such, QoS is not discussed further in this or other VIPR

3.4.  Troubleshooting Problem

   The final problem that is prohibing large scale inter-domain
   federation is troubleshooting.  When connecting calls between
   domains, problems can occur.  Calls can be blocked.  Calls can be
   misdelivered.  Features sometimes don't work.  There can be one-way
   media or no media at all.  The video may not start.  Call quality can
   be poor.

   These problems are common in SIP deployments, and they are tough to
   troubleshoot even within a single administrative domain.  When real-
   time services extend inter-domain, the problem becomes worse.

   Fortunately, some work has been completed to improve the ability for

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   network administrators to diagnose SIP problems.  A Common log format
   [RFC6873] has been developed.  Other work underway, such as
   consistent session IDs [I-D.ietf-insipid-session-id-reqts] and
   [I-D.jones-insipid-session-id] can help troubleshoot interdomain

   In addition to the above, any new technology that facilitates inter-
   domain federation needs to have troubleshooting built-in, so that it
   is not a barrier to deployment.  Further consideration of necessary
   built-in techniques for troubleshooting is required for successful
   deployment of VIPR.

4.  Summary of Existing Solutions

   Given the value of inter-domain SIP federation, there are existing
   deployed solutions summarized below.  However, each solution approach
   has fundamental limitations that have inhibited widespread

4.1.  Domain Routing

   The first solution for SIP inter-domain federation is built into SIP
   itself - domain routing.  In this technique, users utilize email-
   style SIP URIs as identifiers.  By utilizing the DNS lookup mechanism
   defined in [RFC3263], SIP enables calls to be routed between domains
   in much the same way email is routed between domains.

   This technique works well in theory, but it has two limitations which
   have limited its deployment:

   1.  The majority of SIP deployments utilize phone numbers, often
       exclusively.  In such a case, domain routing cannot be used.
   2.  Domain federation brings with it the possibility (and strong
       likelihood) of the same levels of spam and DoS attacks that have
       plagued the email system.

   These issues have already been discussed in sections Section 3.1 and
   Section 3.2 respectively.

4.2.  Public ENUM

   Public ENUM, defined in [RFC6116] addresses the phone number routing
   problem by placing phone numbers into the public DNS.  Clients can
   then perform a simple DNS lookup on a phone number, and retrieve a
   SIP URI which can be used to route to that phone number.

   Unfortunately, public ENUM requires that the entries placed into the

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   DNS be populated following a chain of responsibility that mirrors the
   ownership of the numbers themselves.  This means that, in order for a
   number to be placed into the DNS, authorization to do so must start
   with the ITU, and from there, move to the country, telecom regulator,
   and ultimately the end user.  The number of layers of bureaucracy
   required to accomplish this is non-trivial.  In addition, the telecom
   operators - that would be partly responsible for populating the
   numbers into the DNS - have little incentive to do so.  As a
   consequence, public ENUM is largely empty, and is likely to remain so
   for the foreseeable future.

   Instead, ENUM has evolved into a technique for federation amongst
   closed peering partners, called private ENUM or infrastructure ENUM
   [RFC5067].  While there is value in this technology, it does not
   enable the open federation that public ENUM was designed to solve.

4.3.  Private Federations

   Private federations are a cooperative formed amongst a small number
   of participating domains.  The cooperative agrees to use a common
   technique for federation, and through it, is able to connect to each
   other.  There are many such federations in use today.

   Some of these federations rely on a central database, typically run
   by the federation provider, that can be queried by participating
   domains.  The database contains mappings from phone numbers to
   domains, and is populated by each of the participating domains, often
   manually.  Each domain implements an agreed-upon query interface that
   can be used to access the database when a number is called.
   Sometimes ENUM is used for this interface (called private ENUM),
   other times, a SIP redirection is used.  Some federations also
   utilize private IP networks in order to address QoS problems.

   Private federations work, but they have one major limitation:  scale.
   As the number of participating domains grows, several problems arise.
   Firstly, the size of the databases become difficult to manage.
   Secondly, the correctness of the database becomes an issue, since the
   odds of misconfigured numbers (either intentionally or accidentally)
   increases.  As the membership grows further, the odds increase that
   malicious domains will be let in, introducing a source of spam and
   further problems.  The owner of the federation can - and often does -
   assume responsibility for this, and can attempt to identify and shut
   down misbehaving participants.  Indeed, as the size of the
   federations grow, the owner of the federation needs to spend
   increasing levels of capital on maintaining it.  This often results
   in the owners charging for membership, which can be a barrier to

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5.  Key Requirements

   From the discussion on the problems of inter-domain federation and
   the solutions that have been attempted so far, several key
   requirements emerge:

   REQ-1:  The solution must allow for federation between any number of
   REQ-2:  The solution must enable users in one domain to identify
      users in another domain through the use of their existing E.164
      based phone numbers.
   REQ-3:  The solution must work with deployments that utilize any kind
      of endpoint, including non-IP phones connected through gateways,
      IP softphones and hardphones.
   REQ-4:  The solution must not require any change in user behavior.
      The devices and techniques that users have been using previously
      to make inter-domain calls must continue to work, but now result
      in inter-domain calls using IP.
   REQ-5:  The solution must work worldwide, for any domain anywhere.
   REQ-6:  The solution must not require any new services from any kind
      of centralized provider.  A domain should be able to deploy
      equipment and connect to the federation without any interaction
      with or authorization from a centralized provider.
   REQ-7:  The solution must not require any prior arrangement between
      domains in order to facilitate federation between those domains.
      Federation must occur opportunistically - connections established
      when they can be.
   REQ-8:  The solution must work for domains of any size - starting
      with a single phone up to the largest telecom operator with tens
      of millions of numbers.
   REQ-9:  The solution must have built-in mechanisms for preventing
      spam and DoS attacks.  These mechanisms must be fully automated.
   REQ-10:  The solution must not require any processing whatsoever by
      SIP or RTP intermediaries.  It must be possible for a direct SIP
      connection to be established between participating domains.
   REQ-11:  The solution should adapt to VIPR call failures.  The
      solution should allow the user to make calls using the inter-
      domain calling mechanism used prior to the initial VIPR-enabled

6.  Executive Overview

   Verification Involving PSTN Reachability (VIPR) is aimed at solving
   the problems that have prevented large-scale Internet-based SIP
   federation of voice and video.  VIPR solves these problems by
   creating a hybrid of three technologies - the PSTN itself, a Peer to
   Peer (P2P) network, and SIP.  By using these three technologies

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   together, VIPR enables an incrementally deployable solution to

6.1.  Key Properties

   VIPR has several important properties that enable it to solve the
   federation problem:

   Works With Numbers:  VIPR enables federation for existing PSTN phone
      numbers.  It does not require users or administrators to know or
      configure email-style identifiers.  It does not require the
      allocation of new numbers.  It does not require a change in user
   Works with Existing Endpoints:  VIPR does not require any changes to
      endpoints.  Consequently, it works with existing SIP endpoints and
      with non-IP endpoints connected through gateways.
   Verified Mappings:  VIPR ensures that phone calls cannot be misrouted
      or numbers stolen.  The biggest issue in mapping from a phone
      number to a domain or IP address, is determining whether the
      mapping is correct - i.e., does the domain really own the given
      phone number?  While solutions like ENUM have solved this problem
      by relying on centralized delegations of authorization, VIPR
      provides a secure mapping in a fully distributed way.
   Worldwide:  VIPR works worldwide.  Any domain that is connected to
      both the PSTN and the Internet can participate.  Since VIPR does
      not depend on availability of any regional services beyond IP and
      PSTN access - both of which are already available globally - VIPR
      itself is globally available.
   Scalibility:  VIPR is scaleable.  Any number of domains can
   Self-Scale:  VIPR self-scales.  This means that the amount of
      computation, memory, and bandwidth that a domain must deploy
      scales in direct proportion to the size of their own user base.
   Self-Learning:  VIPR is completely automated.  A domain does not
      require configuration of any information about another domain.  It
      does not require provisioning of IP addresses, domain names,
      certificates, phone number prefixes or routing rules.
   Automated Anti-Spam  VIPR has a built-in mechanism for preventing SIP
      spam, which is specific to SIP.  It is fundamentally different
      from existing SIP anti-spam techniques which borrow from email
      [RFC5039].  This new technique is fully automated, and requires no
      configuration by administrators and no participation from end
   Feature Velocity:  VIPR enables direct SIP connections between two
      domains seeking to federate.  There are no SIP intermediaries of
      any sort between the two.  This means that domains have no
      dependencies on intermediaries for deployment of new features.

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   Secure:  Security is a fundamental part of VIPR and cannot be
   Reliable:  VIPR is reliable.  Through its hybridization of the PSTN
      and the Internet, it ensures that calls always go through, even in
      cases of network failure or limited IP connectivity.

   In order to achieve a solution with these properties, past
   assumptions about how federations should work must be challenged.

6.2.  Challenging Past Assumptions

   Two unstated assumptions of SIP federation are challenged by VIPR.

   The first assumption that federation solutions have made is this:
      The purpose of SIP federation is to eliminate the PSTN, and
      consequently, we cannot assume the PSTN itself as part of the
   Though unstated, this assumption has clearly been part of the design
   of existing solutions.  SIP federation based on email-style URIs, as
   defined in RFC 3261, doesn't utilize nor make mention of the PSTN.
   Solutions like ENUM, or private registries, also do not utilize nor
   make mention of the PSTN.  However, such approaches ignore an
   incremental solution - a solution which utilizes the PSTN itself to
   solve the hard problems in SIP federation.

   There are many advantages to leveraging the PSTN.  It reaches
   worldwide.  It provides a global numbering translation service that
   maps phone numbers to circuits.  It is highly reliable, and provides
   QoS.  It has been built up over decades to achieve these goals.
   Thus, building upon rather than replacing the PSTN, can provide the
   necessary functionality once another assumption is challenged.

   This second assumption is:
      A federation solution must be the same as the final target
      federation architecture, and not just a step towards it.
   SIP's email-style federation was a pure 'target architecture'.  ENUM
   was the same - a worldwide global DNS database with everyone's phone
   numbers providing open connectivity.

   Historically, technologies are more successful when they are
   incrementally deployable.  As such, VIPR is very much focused on
   incremental deployability.  It discards the notion of perfect IP
   federation for a solution that federates most, but not all calls, by
   relying on the PSTN to fill in the gaps.

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6.3.  Technical Overview

   A high level view of the VIPR architecture with an example is shown
   in Figure 1.  The figure shows four different domains,,, and, federated using VIPR
   technology.  Each domain is connected to both the public Internet and
   to the traditional PSTN.  For simplicity, the connection for the call
   agents in and to the PSTN is not indicated in
   the diagram as that interface is not relevant to the subsequent

                         +-------+    +-------+
                         |  Call |    |  Call |
    | Agent |    | Agent |
                         |       |    |       |
                         +-------+    +-------+
                             \           /
                              \         /
                               \       /
                                \     /
                           |//          \\|
                           |   Internet   |
             +-------+     |\\          //|    +-------+
             |  Call |------ \\ _______//------|  Call |
     //\\    | Agent |                         | Agent |    //\\
     \  /    |       |                         |       |    \  /
      \/  ---|       |      +-----------+      |       |---- \/
     User    |       |======|           |======|       |    User
    Agent    +-------+      |    PSTN   |      +-------+   Agent
       |           |

                     Figure 1: High Level Architecture

   For purposes of explanation, it is easiest to think of each domain as
   having a single call agent which participates in the federation
   solution.  The functionality is decomposed into several sub-
   components, and this is discussed in more detail below.  The call
   agent is connected to one or more user agents in the domain, and is
   responsible for routing calls, handling features, and processing call
   state.  The call agent is stateful, and is aware of when calls start
   and stop.  Additional detail for the functional components of this
   architecture are provided in [I-D.petithuguenin-vipr-framework].

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   Assume that all four domains have a 'fresh' installation of VIPR, and
   that domain 'owns' +1 408 555 5xxx, a block of 1000
   numbers allocated by its PSTN provider.

   The VIPR mechanism can be broken into four basic steps:  storage of
   phone numbers, PSTN first call, validation and caching, and
   subsequent SIP call(s).

6.3.1.  Storage of Phone Numbers

   The first step is that the call agents form a single, worldwide P2P
   network, using a VIPR specific usage
   [I-D.petithuguenin-vipr-reload-usage] of RELOAD
   [I-D.ietf-p2psip-base] with a variant of the Chord algorithm.  This
   P2P network forms a distributed hash table (DHT) running amongst all
   participating domains.  A distributed hash table is like a simple
   database, allowing storage of key-value pairs, and lookup of objects
   by key.  Unlike a normal hash table, which resides in the memory of a
   single computer, a distributed hash table is spread across all of the
   servers which make up the P2P network.  In this case, it is spread
   across all of the domains participating in the VIPR federation.

   The problem solved by the variant of the Chord algorithm (and by
   other DHT algorithms), is an answer to the following:  given that the
   desired operation is to read or write an object with key K, which
   node in the DHT is the box that currently stores the object with that
   key?  The P2P SIP variant of the Chord algorithm provides an
   algorithm which routes read and write operations through nodes in the
   DHT until they eventually arrive at the right place.  With Chord,
   this will take no more than log2N hops, where N is the number of
   nodes in the DHT.  Consequently, for a DHT with 1024 nodes, 10 hops
   are required in the worst case.  For 2048, 11 hops.  And so on.  The
   logarithmic factor allows DHTs to achieve efficient scale and to
   provide a large amount of storage summed across all of the nodes that
   make up the DHT.

   This logarithmic hopping behavior also means that each node in the
   DHT does not need to establish a TCP/TLS connection to every other
   node.  Rather, connections are established to a smaller subset - just
   log(N) of the nodes.

   In DHTs, each participating entity is identified by a Node-ID.  The
   Node-ID is a 128 bit number, assigned randomly to each entity.  They
   have no inherent semantic meaning; they are not like domain names or
   IP addresses.

   In the case of VIPR, each call agent is identified by one or more
   Node-IDs.  For purposes of discussion, consider the case where the

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   call agent has just one Node-ID.  Each participating domain,
   including in our example, uses the DHT to store a mapping
   from each phone number that it owns, to the domain's Node-ID.  In the
   case of, it would store 1000 entries into the DHT, each
   one being a mapping from one of its phone numbers, to the domain's
   Node-ID.  Furthermore, when the mappings are stored, the mapping is
   actually from the SHA-1 hash of the phone number, to the Node-ID of
   the call agent which claims ownership of that number.

   For example, if the Node-ID of the call agent in domain
   is 0x1234 (a shorter 16 bit value to simplify discussion), the
   entries stored into the DHT by would be:

      Key             |    Value
   SHA1(+14085555000)  |   0x1234
   SHA1(+14085555001)  |   0x1234
   SHA1(+14085555002)  |   0x1234
   SHA1(+14085555999)  |   0x1234

                          Figure 2: DHT Contents

   It is important to note that the DHT does not contain phone numbers
   (it contains hashes of them), nor does it contain IP addresses or
   domain names.  Instead, it is a mapping from the hash of a phone
   number (in E.164 format) to a Node-ID. will store this mapping when it starts up, or when a new
   number is provisioned.  The information is refreshed periodically by  The actual server on which these mappings are stored
   depends on the variant of the Chord algorithm.  Typically, the
   entries will be uniformly distributed amongst all of the call agents
   participating in the network.

6.3.2.  PSTN First Call

   At some point, a user agent (Alice) in makes a call to +1
   408 555 5432, which is her colleague Bob. Even though both sides have
   VIPR, the call takes place over the plain old PSTN, per Figure 3.
   Alice talks to Bob for a bit, and they hang up.

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             +-------+                         +-------+
             |  Call |                         |  Call |
     //\\    | Agent |                         | Agent |    //\\
     \  /    |       |                         |       |    \  /
      \/  ---|       |      +-----------+      |       |---- \/
      Alice  |       |<=======<========>======>|       |    Bob
             +-------+      |    PSTN   |      +-------+
       |           |

                         Figure 3: PSTN First Call

   At a random point in time after the call has completed, the call
   agent in "wakes up" and says to itself, "that's
   interesting, someone in my domain called +1 408 555 5432, and it went
   over the PSTN.  I wonder if that number is reachable over IP
   instead?".  To make this determination, it hashes the called phone
   number, and looks it up in the DHT.  It is important to note that
   this lookup is not at the time of an actual phone call - this lookup
   process happens outside of any phone call, and is a background

   The query for +1 408 555 5432 will traverse the DHT, and eventually
   arrive at the node that is responsible for storing the mapping for
   that number.  Typically, that node will not be, but
   rather one of the other nodes in the network (e.g.,  In
   many cases, the called number will not find a matching mapping in the
   DHT.  This happens when the number that was dialed is not owned by a
   domain participating in VIPR.  When that happens, takes
   no further action.  Next time there is another call to the same
   number, it will repeat the process and check once more whether the
   dialed number is in the DHT.

   In this case, there is a match in the DHT, and learns the
   Node-ID of  It then proceeds to the validation step per
   Section 6.3.3.  It is also possible that there are multiple matches
   in the DHT.  This can happen if another domain - for
   example - also claims ownership of that number.  When there are
   multiple matching results, learns all of them, and
   performs the validation step with each.

6.3.3.  Validation and Caching

   Why not just store the domain in the DHT, instead of the Node-ID?  If
   the domain was stored in the DHT, once performed the

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   lookup, it would immediately learn that the number maps to, and could then make a direct SIP call next time.

   The main reason this doesn't work is security.  The information in
   the DHT is completely untrusted.  There is nothing so far that
   enables to know that does, in fact, own the
   phone number in question.  Indeed, if multiple domains make a claim
   on the number, it has no way to know which one (if any) actually owns

   To address this critical problem, VIPR requires a mechanism called
   phone number validation.  Phone number validation is a key concept in
   VIPR.  There are several models for this validation as detailed in
   [I-D.petithuguenin-vipr-pvp].  The essential idea is that
   will connect to the server, by asking the DHT to form a
   connection to's Node-ID.  Once connected,
   demands proof of ownership of the phone number.  This proof comes in
   the form of demonstrated knowledge of the previous PSTN call.  When a
   call was placed from to +1 408 555 5432, the details of
   that call - including its caller ID, start time, and stop time,
   create a shared secret referred to as a "ticket", - information that
   is only known to entities that participated in the call.  Thus, to
   obtain proof that really owns the number in question, will demand a knowledge proof - that is aware
   of the details of the call.  A consequence of this is that the
   following property is maintained:

      A domain can only call a specific number over SIP, if it had
      previously called that exact same number over the PSTN.

   This property is key in fighting spam and denial-of-service attacks.
   Because calling numbers on the PSTN costs money - especially
   international calls - VIPR creates a financial disincentive for
   spammers.  For a spammer to ring every phone in a domain with a SIP
   call, it must have previously called every number in the domain with
   a PSTN call, and had a successfully completed call to each and every
   one of them.  [I-D.petithuguenin-vipr-sip-antispam] provides an
   overview and further details on the security mechanisms for VIPR for
   mitigation of SPAM.

   There are a great many details required for this validation protocol
   to be secured.  For example, the mechanism needs to handle the fact
   that call start and stop times won't exactly match on both sides.  It
   needs to deal with the fact that many calls start on the top of the
   hour.  It needs to deal with the fact that caller ID is not often
   delivered, and when it is delivered, is not reliable.  It needs to
   deal with the fact that may in fact be the attacker,
   trying to use the validation protocol to extract the shared secret

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   from  All of this is, in fact, handled by the protocol.
   The protocol is based on the Secure Remote Password for TLS
   Authentication (SRP-TLS) [RFC5054], and is described more fully in

   Towards the end of the validation process, domains and had determined that each was, in fact in possession of
   the shared secret information about the prior PSTN call.  However,
   neither side has any information about the domain names of the other

   At the end of the validation process, both and have been able to ascertain that the other side did in
   fact participate in the previous PSTN call.  At that point, sends its domain name to as shown in
   Figure 4.

                         +-------+    +-------+
                         |  Call |    |  Call |
    | Agent |    | Agent |
                         |       |    |       |
                         +-------+    +-------+
                            \             /
   +----------------------+  \           /
   | Hi, I am|   \         /
   | How do I reach you?  |    \       /
   +--------------\-------+  //-------\\
                   \        //         \\
                ^          |    Internet  |        |
                |          |              |        v
             +-------+     |\\          //|    +-------+
             |  Call |------ \\ _______//------|  Call |
     //\\    | Agent |                         | Agent |    //\\
     \  /    |       |                         |       |    \  /
      \/  ---|       |                         |       |---- \/
      Alice  |       |                         |       |    Bob
             +-------+                         +-------+

                    Figure 4: Ticket Validation Step 1

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   Next, the domain generates the ticket.  The ticket has
   three fundamental parts to it:

   1.  The phone number that was just validated - in this case, +1 408
       555 5432.
   2.  The domain name that the originating side claims it has - in this case.
   3.  A signature generated by, using a key known to itself
       only, over the other two pieces of information.

   Then, sends to - all over a secured channel -
   a SIP URI to use for routing calls to this number, and a ticket, as
   shown in Figure 5.  The ticket is a cryptographic object, opaque to, but used by to allow incoming SIP calls.  It
   is similar in concept to kerberos tickets - it is a grant of access.
   In this case, it is a grant of access for to call +1 408
   555 5432, and only +1 408 555 5432.

                         +-------+    +-------+
                         |  Call |    |  Call |
    | Agent |    | Agent |
                         |       |    |       |
                         +-------+    +-------+
                            \             /
                             \           /    +------------------------+
                              \         /     | Here is your ticket    |
                               \       /      | & SIP URI to reach Bob |
                             //-------\\      +----/-------------------+
                            //         \\         /
                |          |    Internet  |        ^
                v          |              |        |
             +-------+     |\\          //|    +-------+
             |  Call |------ \\ _______//------|  Call |
     //\\    | Agent |                         | Agent |    //\\
     \  /    |       |                         |       |    \  /
      \/  ---|       |                         |       |---- \/
      Alice  |       |                         |       |    Bob
             +-------+                         +-------+

                    Figure 5: Ticket Validation Step 2

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   The call agent receives the SIP URI and ticket, and
   stores both of them in an internal cache.  This cache builds up
   slowly over time, containing the phone number, SIP URI, and ticket,
   for those numbers which are called by and validated using
   VIPR.  Because the cache entries are only built for numbers which
   have actually been called by users in the enterprise, the size of the
   cache self-scales.  A call agent supporting only ten users will build
   up a cache proportional to the volume of numbers called by ten
   people, whereas a call agent supporting ten thousand users will build
   up a cache which is typically a thousand times larger.

   This cache, containing the phone number, SIP URI and ticket will be
   accessed later when Alice (or another caller from the same call
   agent) makes another call to Bob, as detailed in Section 6.3.4.

6.3.4.  SIP Call

   At some point in the future, another call is made to +1 408 555 5432.
   The caller could be Alice, or it could be any other user attached to
   the same call agent.  This time, the call agent notes that it has a
   cached entry (including the SIP URI and ticket) for the number in
   question.  It is possible that there are multiple entries for a given
   number.  For example, both an Enterprise and Service Provider may
   register the same number in the RELOAD distributed database.  It may
   also be possible to fork a call using the multiple entries .
   [Editor's note:  this requires further discussion as to whether we
   want to allow multiple entries.]

   The call agent attempts to contact the SIP URI by
   establishing a TCP/TLS connection to the SIP URI it learned.  If a
   connection cannot be made and there are no other cached entries for
   the number in question, the call agent proceeds with the call over
   the PSTN.  This ensures that, in the event of an Internet failure or
   server failure, the call can still proceed.  Assuming the connection
   is established, the call agent sends a SIP INVITE to the
   terminating call agent, over this newly formed secure connection.
   The SIP INVITE request also contains the ticket, placed into a new
   SIP header field in the message.

   When the SIP INVITE arrives at the call agent, the call
   agent can extract the ticket from the new SIP header field.  This
   ticket is an object, opaque to, that was previously
   generated by the call agent as described in
   Section 6.3.3. first verifies the signature over the
   ticket.  Remember that the agent is the one that
   generated the ticket in the first place; as such, it is in possession
   of the key required to validate the signature.  Once validated, it
   performs two checks:

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   1.  It compares the phone number in the call setup request (the
       Request URI) against the phone number stored in the ticket.
   2.  It compares the domain name of the calling domain, learned from
       the certificates in the mutual TLS exchange, against the domain
       name stored in the ticket.

   If both match, the call agent knows that the calling
   party is in fact the domain they claimed previously, and that they
   had in fact gone through the validation process successfully for the
   number in question.  At this time, the call is now completed per
   normal SIP processing.

7.  Security Considerations

   This section provides an overview of some of the key threats and how
   they are handled at a high level.  Note that the detailed security
   solutions to handle the threats are detailed in the other relevant
   VIPR documents as referenced in the sections below.

7.1.  Attacks on the DHT

   Attackers could attempt to disrupt service through a variety of
   attacks on the DHT.

   Firstly, it must be noted that the DHT is never used at call setup
   time.  It is accessed as a background task, solely to learn NEW
   numbers and SIP URIs that are not already known.  If an attacker was
   able to completely destroy the P2P network, it would not result in a
   single call to fail.  Furthermore, it would not cause calls to revert
   to the PSTN - calls to SIP URIs learned previously would still go
   over the IP network.  The only impact to such a devastating attack is
   that a domain could not learn SIP URIs for new numbers, until the DHT
   is restored to service.  This service failure is hard for users and
   administrators to even notice.

   That said, VIPR prevents many of these attacks.  The DHT itself is
   secured using TLS - its usage is mandatory.  Quota mechanisms are put
   into place that prevent an attacker from storing large amounts of
   data in the DHT as described in
   [I-D.petithuguenin-vipr-proportional-quota].  Other attacks are
   prevented by mechanisms defined by RELOAD [I-D.ietf-p2psip-base]
   itself, and are not VIPR specific.

7.2.  Theft of Phone Numbers

   A key security threat that VIPR is trying to address is the theft of
   phone numbers.  In particular, a malicious domain could store, in the

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   DHT, phone numbers that it does not own, in an attempt to steal calls
   targeted to those numbers.  This attack is prevented by the core
   validation mechanism as described in [I-D.petithuguenin-vipr-pvp] ,
   which performs a proof of knowledge check to verify ownership of

   An attacker could try to claim numbers it doesn't own, which are
   claimed legitimately by other domains in the VIPR network.  This
   attack is prevented as well.  Each domain storing information into
   the DHT can never overwrite information stored by another domain.  As
   a consequence, if two domains claim the same number, two records are
   stored in the DHT.  An originating domain will validate against both,
   and only one will validate - the real owner.

   An attacker could actually own a phone number, use it for a while,
   validate with it, and build up a cache of routes at other domains.
   Then, it gives back the phone number to the PSTN provider, who
   allocates it to someone else.  However, the attacker still claims
   ownership of the number, even though they no longer have it.  This
   attack is prevented by expiring the learned routes after a while.
   Typically, operators do not re-assign a number for a few months, to
   allow out-of-service messages to be played to people that still have
   the old number.  Thus, the TTL for cached routes is set to match the
   duration that carriers typically hold numbers.

   An attacker could advertise a lot of numbers, most of which are
   correct, some of which are not.  VIPR prevents this by requiring each
   number to be validated individually.

   An attacker could make a call so they know the call details of the
   call they made and use this to forge a validation for that call.
   They could then try to convince other users, which would have to be
   in the same domain as the attacker, to trust this validation.  This
   is mitigated by not sharing validations inside of domains where the
   users that can originate call from that domain are not trusted by the

7.3.  Spam

   Another serious concern is that attackers may try to launch SIP spam
   (also known as SPIT) calls into a domain.  As described in
   Section 6.3.3 and as detailed in
   [I-D.petithuguenin-vipr-sip-antispam], VIPR prevents this by
   requiring that a domain make a PSTN call to a number before it will
   allow a SIP call to be accepted to that same number.  This provides a
   financial disincentive to spammers.  The current relatively high cost
   of international calling, and the presence of national do-not-call
   regulations, have prevented spam on the PSTN to a large degree.  VIPR

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   applies those same protections to SIP connections.

   VIPR still lowers the cost of communications, but it does so by
   amortizing that savings over a large number of calls.  The costs of
   communications remain high for infrequent calls to many numbers, and
   become low for frequent calls to a smaller set of numbers.  Since the
   former is more interesting to spammers, VIPR gears its cost
   incentives away from the spammers, and towards domains which
   collaborate frequently.

   It is important to note that VIPR does not completely address the
   spam problem.  A large spamming clearing house organization could
   actually incur the costs of launching the PSTN calls to numbers, and
   then, in turn, act as a conduit allowing other spammers to launch
   their calls to those numbers for a fee.  The clearinghouse would
   actually need to transit the signaling traffic (or, divulge the
   private keys to their domain name), which would incur some cost.  As
   such, while this is not an impossible situation, the barrier is set
   reasonably high to start with - high enough that it is likely to
   deter spammers until it becomes a highly attractive target, at which
   point other mechanisms can be brought to bear.

7.4.  Eavesdropping

   Another class of attacks involves outsiders attempting to listen in
   on the calls that run over the Internet, or obtain information about
   the call through observation of signaling.

   All of these attacks are prevented by requiring the usage of SIP over
   TLS and SRTP.  These are mandatory to use.

8.  IANA Considerations

   This specification does not require any actions from IANA.

9.  Acknowledgements

   Thanks for review comments from Ken Fischer, Rob Maidhof, Michael
   Procter, Eric Burger, Richard Barnes and others.  Thanks to Theo
   Zourzouvillys for pointing out the 5th theft of phone numbers attack
   as described in Section 7.2 .

10.  References

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10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.

              Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and
              H. Schulzrinne, "REsource LOcation And Discovery (RELOAD)
              Base Protocol", draft-ietf-p2psip-base-26 (work in
              progress), February 2013.

              Petit-Huguenin, M., Rosenberg, J., and C. Jennings, "A
              Usage of Resource Location and Discovery (RELOAD) for
              Public Switched Telephone Network (PSTN) Verification",
              draft-petithuguenin-vipr-reload-usage-04 (work in
              progress), March 2012.

              Petit-Huguenin, M., Jennings, C., and J. Rosenberg,
              "Verification Involving PSTN Reachability (VIPR):
              Framework", draft-petithuguenin-vipr-framework-00 (work in
              progress), October 2011.

              Petit-Huguenin, M., Rosenberg, J., and C. Jennings,
              "Session Initiation Protocol (SIP) Extensions for Blocking
              VoIP Spam Using PSTN Validation",
              draft-petithuguenin-vipr-sip-antispam-03 (work in
              progress), January 2012.

              Jennings, C., Rosenberg, J., and M. Petit-Huguenin,
              "Verification Involving PSTN Reachability: The ViPR Access
              Protocol (VAP)", draft-jennings-vipr-vap-02 (work in
              progress), March 2012.

              Petit-Huguenin, M., Rosenberg, J., and C. Jennings, "The
              Public Switched Telephone Network (PSTN) Validation
              Protocol (PVP)", draft-petithuguenin-vipr-pvp-04 (work in
              progress), March 2012.

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              Petit-Huguenin, M., Rosenberg, J., and C. Jennings,
              "Proportional Quota in REsource LOcation And Discovery
              (RELOAD)", draft-petithuguenin-vipr-proportional-quota-00
              (work in progress), October 2011.

10.2.  Informative References

   [RFC2543]  Handley, M., Schulzrinne, H., Schooler, E., and J.
              Rosenberg, "SIP: Session Initiation Protocol", RFC 2543,
              March 1999.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation
              Protocol (SIP): Locating SIP Servers", RFC 3263,
              June 2002.

   [E.164]    ITU-T, "The International Public Telecommunication Number
              Plan", Recommendation E.164, May 1997.

   [RFC5039]  Rosenberg, J. and C. Jennings, "The Session Initiation
              Protocol (SIP) and Spam", RFC 5039, January 2008.

   [RFC6116]  Bradner, S., Conroy, L., and K. Fujiwara, "The E.164 to
              Uniform Resource Identifiers (URI) Dynamic Delegation
              Discovery System (DDDS) Application (ENUM)", RFC 6116,
              March 2011.

   [RFC5067]  Lind, S. and P. Pfautz, "Infrastructure ENUM
              Requirements", RFC 5067, November 2007.

   [RFC5054]  Taylor, D., Wu, T., Mavrogiannopoulos, N., and T. Perrin,
              "Using the Secure Remote Password (SRP) Protocol for TLS
              Authentication", RFC 5054, November 2007.

   [RFC6873]  Salgueiro, G., Gurbani, V., and A. Roach, "Format for the
              Session Initiation Protocol (SIP) Common Log Format
              (CLF)", RFC 6873, February 2013.

              Jones, P., Pearce, C., Polk, J., and G. Salgueiro, "End-
              to-End Session Identification in IP-Based Multimedia
              Communication Networks", draft-jones-insipid-session-id-02
              (work in progress), February 2013.

              Jones, P., Salgueiro, G., Polk, J., Liess, L., and H.
              Kaplan, "Requirements for an End-to-End Session
              Identification in IP-Based Multimedia Communication

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              Networks", draft-ietf-insipid-session-id-reqts-04 (work in
              progress), January 2013.

Appendix A.  Changes since last version

   This section must be removed before publication as an RFC.

   Modifications between jennings-04 and jennings-03:

   1.  Updating references to SIPCLF and Session ID (INSIPID) documents.

   Modifications between jennings-03 and jennings-02:

   1.  Reworded REQ -11 to clarify that in the case of call failures
       (i.e., IP calls), the system should fallback to inter-domain
       calling prior to VIPR.
   2.  Deleted REQ-12 (Handover) since it's really not specific
       functionality provided by VIPR.
   3.  Moved some text from the -01 version in the Technical Overview
       section back into the doc (not sure why it was removed
   4.  Other editorial changes:

      - Added a Terminology section.
      - Clarified the use of the term "Call Agent".
      - Reworded discussion of email in section 2.2 (i.e., it's not
      - Either changed or removed altogether terms like "neat",
      "clever", "incredible", "enormous" and any text that read like
      marketing literature as much as possible.
      - Removed some of the more subjective and superfluous language -
      i.e., condensed the text to be more concise (Section 5.2 and many
      others per the previous change)
      - Deleted explicit reference to "SIP Trunking" as the statement
      didn't introduce additional information in that paragraph and the
      term is not defined in this document.
      - and other minor editorial fixes.

   Modifications between jennings-02 and jennings-01:

   1.  Sections 6,7,8 moved to new VIPR framework document.
   2.  Editorial changes.
   3.  Clarifications to re-enforce that the primary objective is not
       PSTN bypass but rather to enable enhanced services such as video
       between domains.  Changed "VoIP" to "SIP" since the focus is not
       specifically voice.

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   4.  Added reference for new framework document.
   5.  Section 5.3:  Added references to other documents as appropriate
       - e.g., -pvp, -spam, etc.
   6.  Moved validation diagrams and text (from 5.3.4) into Validation
       and caching section (5.3.3).
   7.  Condensed discussion of spam in section 5.3.3 and updated SPAM
       section in security section.

   Modifications between jennings-01 and rosenberg-04:

   o  Not specified.

   Modifications between rosenberg-04 and rosenberg-03

   o  Nits.
   o  Shorter I-Ds references.
   o  Changed phone numbers to follow E.123 presentation.
   o  Expanded P2P initialisms.
   o  Uses +1 408 555 prefix for phone numbers in examples.

Authors' Addresses

   Mary Barnes


   Cullen Jennings
   170 West Tasman Drive
   MS: SJC-21/2
   San Jose, CA  95134

   Phone:  +1 408 421-9990

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   Jonathan Rosenberg
   Monmouth, NJ


   Marc Petit-Huguenin


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