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dispatch                                                  J.R. Rosenberg
Internet-Draft                                               jdrosen.net
Intended status:  Standards Track                            C. Jennings
Expires:  April 28, 2011                                           Cisco
                                                       M. Petit-Huguenin
                                                               Stonyfish
                                                        October 25, 2010


Verification Involving PSTN Reachability: Requirements and Architecture
                                Overview
               draft-rosenberg-dispatch-vipr-overview-04

Abstract

   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, 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 VoIP anti-spam, and
   privacy of number ownership, all while preserving the trapezoidal
   model of SIP.

Legal

   This documents and the information contained therein are provided on
   an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE
   IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL
   WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY
   WARRANTY THAT THE USE OF THE INFORMATION THEREIN WILL NOT INFRINGE
   ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS
   FOR A PARTICULAR PURPOSE.

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



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   Task Force (IETF).  Note that other groups may also distribute
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   Internet-Drafts are draft documents valid for a maximum of six months
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Copyright Notice

   Copyright (c) 2010 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
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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  5
   2.  Problem Statement  . . . . . . . . . . . . . . . . . . . . . .  5
     2.1.  The Phone Number Routing Problem . . . . . . . . . . . . .  6
     2.2.  The Open Pinhole Problem . . . . . . . . . . . . . . . . .  7
     2.3.  Quality of Service Problem . . . . . . . . . . . . . . . .  7
     2.4.  Troubleshooting Problem  . . . . . . . . . . . . . . . . .  8
   3.  Summary of Existing Solutions  . . . . . . . . . . . . . . . .  8
     3.1.  Domain Routing . . . . . . . . . . . . . . . . . . . . . .  8
     3.2.  Public ENUM  . . . . . . . . . . . . . . . . . . . . . . .  9
     3.3.  Private Federations  . . . . . . . . . . . . . . . . . . .  9
   4.  Key Requirements . . . . . . . . . . . . . . . . . . . . . . . 10
   5.  Executive Overview . . . . . . . . . . . . . . . . . . . . . . 11
     5.1.  Key Properties . . . . . . . . . . . . . . . . . . . . . . 11
     5.2.  Challenging Past Assumptions . . . . . . . . . . . . . . . 13
     5.3.  Technical Overview . . . . . . . . . . . . . . . . . . . . 14
       5.3.1.  Storage of Phone Numbers . . . . . . . . . . . . . . . 16
       5.3.2.  PSTN First Call  . . . . . . . . . . . . . . . . . . . 17
       5.3.3.  Validation and Caching . . . . . . . . . . . . . . . . 18
       5.3.4.  SIP Call . . . . . . . . . . . . . . . . . . . . . . . 19
   6.  Architecture Components and Functions  . . . . . . . . . . . . 24
     6.1.  ViPR Server  . . . . . . . . . . . . . . . . . . . . . . . 25
     6.2.  Call Agent . . . . . . . . . . . . . . . . . . . . . . . . 26
     6.3.  Border Element . . . . . . . . . . . . . . . . . . . . . . 27
     6.4.  Enrollment Server  . . . . . . . . . . . . . . . . . . . . 28
     6.5.  P2P Network  . . . . . . . . . . . . . . . . . . . . . . . 28
   7.  Protocols  . . . . . . . . . . . . . . . . . . . . . . . . . . 28
     7.1.  P2P: RELOAD  . . . . . . . . . . . . . . . . . . . . . . . 28
       7.1.1.  ViPR Usage . . . . . . . . . . . . . . . . . . . . . . 29
       7.1.2.  Certificate Usage  . . . . . . . . . . . . . . . . . . 30
     7.2.  ViPR Access Protocol (VAP) . . . . . . . . . . . . . . . . 30
     7.3.  Validation Protocol  . . . . . . . . . . . . . . . . . . . 31
     7.4.  SIP Extensions . . . . . . . . . . . . . . . . . . . . . . 32
   8.  Example Call Flows . . . . . . . . . . . . . . . . . . . . . . 32
     8.1.  PSTN Call and VCR Upload . . . . . . . . . . . . . . . . . 32
     8.2.  DHT Query and Validation . . . . . . . . . . . . . . . . . 34
     8.3.  DHT Query and No Match . . . . . . . . . . . . . . . . . . 35
     8.4.  SIP Call . . . . . . . . . . . . . . . . . . . . . . . . . 35
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 36
     9.1.  Attacks on the DHT . . . . . . . . . . . . . . . . . . . . 37
     9.2.  Theft of Phone Numbers . . . . . . . . . . . . . . . . . . 37
     9.3.  Spam . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
     9.4.  Eavesdropping  . . . . . . . . . . . . . . . . . . . . . . 38
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 39
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 39
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 39
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 39



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     12.2. Informative References . . . . . . . . . . . . . . . . . . 40
   Appendix A.  Release notes . . . . . . . . . . . . . . . . . . . . 41
     A.1.  Modifications between rosenberg-04 and rosenberg-03  . . . 41
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41















































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1.  Introduction

   The Session Initiation Protocol (SIP) was originally published as RFC
   2543 [RFC2543] in May of 1999.  This was followed by subsequent
   publication of RFC 3261 [RFC3261], which brought the protocol to
   sufficient maturity to enable large scale market adoption.

   And indeed, it has seen large scale market adoption.  SIP has seen
   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 lingua franca of
   interconnection between products from different vendors.  If one
   measures success in deployment, then clearly SIP is a success.

   However, in other ways, it has failed.  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.  Though SIP is used between domains, it is
   typically through private federation agreements.  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 enables us to break down the
   barriers that have prevented inter-domain VoIP.  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 over the unrealizable
   nirvana.  At the same time, ViPR ensures that SIP's trapezoidal model
   - 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.  Problem Statement

   The first question that must be asked is this - why haven't we seen
   widespread adoption of inter-domain SIP federation?

   There are many reasons for it.  They are - in order of importance -
   the phone number routing problem, the open pinhole problem, the
   quality of service problem, and the troubleshooting problem.  The two
   former ones are the most significant.






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2.1.  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
   (example.com) 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, and not email-
   style SIP URI.  This is due to the huge installed base of users that
   continue to exist solely on the public switched telephone network
   (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 regular
   PSTN number.  Users in SIP deployments need to place that PSTN 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
   PSTN 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.  Why have two identifiers when one will suffice?
   The universality of PSTN 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 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 numbers
   rather than email-style SIP URI.  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.  Numbers can be ported between
   carriers.  They can be assigned to a user or enterprise, and then
   later re-assigned to someone else.  Numbers are granted to users and
   enterprises through a complex delegation process involving the ITU,



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   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.

2.2.  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 the 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, to the point of
   becoming useless.  Administrators of SIP domains fear - rightfully so
   - that if they make a SIP server available for anyone on the Internet
   to contact, it will open the floodgates for VoIP spam, which is far
   more disruptive than email-based spam [RFC5039].  Administrators also
   worry - rightfully so - that an open server will create a back-door
   for denial-of-service and other attacks that can potentially disrupt
   their voice service.  Administrators are simply not willing to take
   that risk; rightly or wrongly, 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, and
   not as an after-thought.

2.3.  Quality of Service Problem

   The Internet does not provide any 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 VoIP deployments that
   run over the Internet.  Though the lack of QoS is a concern, it has



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   not proven a barrier to deployment.  We believe that, if the more
   fundamental issues - the phone number routing and open pinhole
   problems - can be addressed, the QoS problem will sort itself out.
   As such, we do not discuss this issue further here.

2.4.  Troubleshooting Problem

   The final problem that is stopping large scale inter-domain
   federation is the troubleshooting problem.  When connecting calls
   between domains, problems will happen.  Calls will get blocked.
   Calls will get misdelivered.  Features won't work.  There will be
   one-way media or no media at all.  The video won't start.  Call
   quality will be poor.

   These problems are common in VoIP deployments, and they are tough to
   troubleshoot even within a single administrative domain.  When real-
   time services extend inter-domain, the problem becomes worse.  A new
   angle is introduced:  the first step is identifying who is at fault.

   Fortunately, work is underway to improve the ability for network
   administrators to diagnose VoIP problems.  Common log formats
   [CLF-SYNTAX] and consistent session IDs [SESSION-ID], for example,
   can help troubleshoot interdomain calls.

   In addition to these, any new technology that facilitates inter-
   domain federation needs to have troubleshooting built-in, so that it
   is not a barrier to deployment.


3.  Summary of Existing Solutions

   Given the value that inter-domain SIP federation brings, it is no
   surprise that many attempts have been made at solving it.  Indeed,
   these have all been deployed to varying degrees.  However, all of
   them have fundamental limitations that have inhibited widespread
   deployment.

3.1.  Domain Routing

   The first solution that has been proposed for SIP inter-domain
   federation is built into SIP itself - domain routing.  In this
   technique, users utilize email-style SIP URI 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:



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   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 above.

3.2.  Public ENUM

   Public ENUM, defined in [RFC3761], tries to address the phone number
   routing problem by cleverly 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
   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 - which 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 morphed 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.

   It is clear from the legacy of ENUM deployments, that any kind of
   phone number routing solution should not rely on government or
   telecom processes for population of the databases.

3.3.  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



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   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.  "SIP
   trunking" - a service being offered by many telecom operators as a
   SIP-based PRI replacement - is a form of private federation.

   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 unruly.  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
   "bad" 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, in turn,
   requires them to charge money for membership, and this can be a
   barrier to entry.


4.  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 should allow for federation between any number
      of domains.
   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 should not require any change in user behavior.
      The devices and techniques that users have been using previously
      to make inter-domain calls should continue to work, but now result
      in inter-domain IP federation.
   REQ-5:  The solution should work worldwide, for any domain anywhere.
   REQ-6:  The solution should not require any new services from any
      kind of centralized provider.  A domain should be able, of its own
      free-will and accord, to deploy equipment and connect to the
      federation.




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   REQ-7:  The solution should 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 at
      a single phone 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.  This mechanism 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.

   These requirements, when put together, appear to be mutually
   unsolvable.  And indeed, they have been - until now.


5.  Executive Overview

   Verification Involving PSTN Reachability (ViPR) is a new technology
   that 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 combining all
   three, ViPR enables an incrementally deployable solution to
   federation.

5.1.  Key Properties

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

   Works With Numbers:  ViPR enables federation for existing PSTN
      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
      behaviors.  Whatever way users were dialing numbers yesterday,
      works with ViPR tomorrow.
   Works with Existing Endpoints:  ViPR does not require any changes to
      endpoints.  Consequently, it works with existing SIP endpoints, or
      with non-IP endpoints connected through gateways.
   Fully Distributed:  ViPR does not require any kind of central
      authority or provider.  A domain wishing to utilize ViPR just
      deploys it on their own.  ViPR utilizes the existing PSTN and
      existing Internet connectivity the domain already has, and by
      combining them, achieves inter-domain federation.  Domains do not
      need to wait for their service providers to roll out any kind of
      new features, databases, or functionality.



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   Verified Mappings:  The biggest issue in mapping from a phone number
      to a domain or IP address, is determining whether the mapping is
      correct.  Does that 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.  ViPR guarantees that phone
      calls cannot be misrouted or numbers stolen.
   Worldwide:  ViPR works worldwide.  Any domain that is connected to
      both the PSTN and the Internet can participate.  It doesn't matter
      whether the domain is in Africa, the Americas, or Australia.
      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.
   Unlimited Scale:  ViPR has nearly infinite scale.  Any number of
      domains can participate.
   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 never, ever
      has to configure any information about another domain.  It never
      has to provision IP addresses, domain names, certificates, phone
      number prefixes or routing rules.  Without any prior coordination,
      ViPR enables one domain to connect to a different domain.
   Automated Anti-Spam  ViPR comes with a built-in mechanism for
      preventing VoIP spam.  This mechanism is new, and specific to
      VoIP.  In this way, it is fundamentally different from existing
      VoIP 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 users.  Though it is
      not a 100% solution to the problem, it brings substantial economic
      and legal ammunition to the table to act as a good deterrent for a
      long while.
   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.
   Designed for the Modern Internet:  ViPR is built to run on the modern
      Internet.  It assumes the worst from everyone.  It assumes limited
      connectivity.  It assumes network failures.  It assumes there are
      attackers seeking to eavesdrop calls.  Security is built-in and
      cannot be disabled.
   Reliable:  ViPR is reliable.  Through its hybridization of the PSTN
      and the Internet, it makes sure that calls always go through.
      Indeed, to route a call between domains A and B, ViPR never
      depends on a server or service anywhere outside of domains A and B
      (besides vanilla PSTN and IP access) being operational.

   At first glance, these properties seem impossible to realize.  And



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   indeed, given the assumptions that have traditionally been made about
   how federation has to work, these properties are impossible to
   realize.  It is only by stepping back, and rethinking these
   fundamental assumptions, that a solution can be found.

5.2.  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
      solution.
   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 or make mention of the PSTN.
   Solutions like ENUM, or private registries, do not utilize or make
   mention of the PSTN.  In one sense, it's obvious that they shouldn't
   - after all, the purpose is to replace the PSTN.  However, such an
   approach ignores an incremental solution - a solution which utilizes
   the PSTN itself to solve the hard problems in SIP federation.

   After all, the PSTN has accomplished a great deal.  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.  This
   begs the question - can we build upon the capabilities already
   provided by the PSTN, and use them to solve the problems that plague
   SIP federation?

   Indeed, the answer is yes 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.
   Though unstated, this assumption has also been true.  SIP's email-
   style federation was a pure 'target architecture' - the place we want
   to get to.  ENUM was the same - a worldwide global DNS database with
   everyone's phone numbers - an unrealizable nirvana of open
   connectivity.

   Historically, technologies are more successful when they are
   incrementally deployable.  Indeed, in many cases, the target
   architecture is unrealizable because there is no obvious way to get
   there.  As such, the focus needs to be on the next incremental step
   that we can take, and that step in turn creates the technological and
   market pressures that will drive the next step.  In the end, the
   target may not be the perfect nirvana we all imagined, but we've at
   least arrived.



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   As such, ViPR is very much focused on incremental deployability.  It
   is not the end of the federation story, it is the beginning.  It
   discards the nirvana of perfect IP federation for a solution that
   federates most, but not all calls, by relying on the PSTN to fill in
   the gaps.  ViPR's philosophy is not to let the perfect be the enemy
   of the good.

5.3.  Technical Overview

   A high level view of the architecture is shown in Figure 1.  The
   figure shows four different domains, a.com, b.com, c.com and d.com,
   federating using ViPR technology.  Each domain is connected to both
   the public Internet and to the traditional PSTN.






































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                                  //\\
                                   \/
                                    |
                                    |
                                    |
                               +-------+
                               |  Call |
                               | Agent |
                               |       |
                               |       | d.com
                               +-------+
                                   |

                              //-------\\
                           |//           \\|
                          |    Internet     |
             +-------+     |\\           //|    +-------+
             |  Call |        \\-------//       |  Call |
     //\\    | Agent |--                      --| Agent |    //\\
      \/  ---|       |        //-------\\       |       |---- \/
             |       |     |//           \\|    |       |
             +-------+    |      PSTN       |   +-------+
                           |\\           //|
               a.com          \\-------//         b.com

                                   |
                               +-------+
                               |  Call |
                               | Agent |
                               |       |
                               |       |
                               +-------+
                                   |     c.com
                                   |
                                  //\\
                                   \/

                     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.  In actuality, 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 phones 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.




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   Assume that all four domains have a 'fresh' installation of ViPR, and
   that domain b.com 'owns' +1 408 555 5..., 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 SIP call.

5.3.1.  Storage of Phone Numbers

   The first step is that the call agents form a single, worldwide P2P
   network, using RELOAD [P2PSIP-BASE] with 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 neat trick solved by Chord (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?  Chord provides a clever
   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 incredible scale and to
   provide enormous 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
   call agent has just one.  Each participating domain, including b.com
   in our example, uses the DHT to store a mapping from each phone
   number that it owns, to its own Node-ID.  In the case of b.com, it
   would store 1000 entries into the DHT, each one being a mapping from



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   one of its phone numbers, to its own 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.

   Pretending that the Node-ID of the call agent in domain b.com is
   0x1234 (a shorter 16 bit value to simplify discussion), the entries
   stored into the DHT by b.com 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.

   b.com will store this mapping when it starts up, or when a new number
   is provisioned.  The information is refreshed periodically by b.com.
   The actual server on which these mappings are stored depends on the
   Chord algorithm.  Typically, the entries will be uniformly
   distributed amongst all of the call agents participating in the
   network.

5.3.2.  PSTN First Call

   At some point, a user (Alice) in a.com 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.  Alice talks to Bob for
   a bit, and they hang up.

   At a random point of time after the call has completed, the call
   agent in a.com "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 process.




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   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 b.com, but rather one
   of the other nodes in the network (for example. c.com).  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, a.com 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 a.com learns the
   Node-ID of b.com.  It then proceeds to the validation step.  It is
   also possible that there are multiple matches in the DHT.  This can
   happen if another domain - d.com for example - also claims ownership
   of that number.  When there are multiple matching results, a.com
   learns all of them, and performs the validation step with each.

5.3.3.  Validation and Caching

   Why not just store the domain in the DHT, instead of the Node-ID?  In
   that case, once a.com performed the lookup, it would immediately
   learn that the number maps to b.com, 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 a.com to know that b.com 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 it.

   To address this critical problem, ViPR utilizes a technique called
   phone number validation.  Phone number validation is the key concept
   in ViPR.  The essential idea is that a.com will connect to the b.com
   server, by asking the DHT to form a connection to b.com's Node-ID.
   Once connected, a.com 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 a.com to +1 408 555
   5432, the details of that call - including its caller ID, start time,
   and stop time, create a form of shared secret - information that is
   only known to entities that participated in the call.  Thus, to
   obtain proof that b.com really owns the number in question, a.com
   will demand a knowledge proof - that b.com is aware of the details of
   the call.  The only way that b.com could know these details is if it
   had received the call, and the only way it could have received the
   call is if it owned the phone number.

   There are a great many details required for this validation protocol



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   to be secured.  It 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
   a.com may in fact be the attacker, trying to use the validation
   protocol to extract the shared secret from b.com.  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 [VIPR-PVP].

   At the end of the validation process, both a.com and b.com have been
   able to ascertain that the other side did in fact participate in the
   previous PSTN call.  At that point, a.com sends its domain name to
   b.com (this is described in more detail below), and b.com sends to
   a.com - all over a secured channel - a SIP URL to use for routing
   calls to this number, and a ticket.  The ticket is a cryptographic
   object, opaque to a.com, but used by b.com 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 a.com to call
   +1 408 555 5432, and only +1 408 555 5432.

   The a.com 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 a.com 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.

5.3.4.  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 route for the number in question, along with a SIP URI that
   can be used to reach that route.  It also has a ticket.

   The a.com call agent attempts to contact the SIP URI by establishing
   a TCP/TLS connection to the SIP URI it learned.  If this connection
   cannot be made, it 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 a.com call agent sends a traditional SIP INVITE to the
   terminating call agent, over this newly formed secure connection.



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   The SIP call setup request also contains the ticket, placed into a
   new SIP header in the message.

   When this call setup request arrives at the b.com call agent, it
   extracts the ticket from the new SIP header.  This ticket is an
   object, opaque to a.com, that was previously generated by the b.com
   call agent.  Figure 3 illustrates how this ticket is generated and
   used.


                                 //\\
                                  \/
   +----------------------+         |
   |                      |         |
   | Hi, I am a.com.      |         |
   | How do I reach you?  |    +-------+
   |                      |    |  Call |
   +--------------\-------+    | Agent |
                   \           |       |
                    \          |       | d.com
                    \          +-------+
                     \             |
                      \
                       \      //-------\\
                +===================================+
                ^         |    Internet     |       V
             +-------+     |\\           //|    +-------+
             |  Call |        \\-------//       |  Call |
     //\\    | Agent |--                      --| Agent |    //\\
      \/  ---|       |        //-------\\       |       |---- \/
             |       |     |//           \\|    |       |
             +-------+    |      PSTN       |   +-------+
                           |\\           //|
               a.com          \\-------//         b.com

                                   |
                               +-------+
                               |  Call |
                               | Agent |
                               |       |
                               |       |
                               +-------+
                                   |     c.com
                                   |
                                  //\\
                                   \/

                    Figure 3: Ticket Validation Step 1



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   Towards the end of the validation process, domains a.com and b.com
   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 side.  The
   originating domain - a.com - tells b.com that its domain name is
   a.com.  It offers no proof of this assertion at this time.

   Next, the b.com 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 - a.com
       in this case.
   3.  A signature generated by b.com, using a key known to itself only,
       over the other two pieces of information.

   This ticket is then sent back to a.com at the end of the validation
   process, as shown in Figure 4.
































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                                 //\\
                                  \/
                                   |
                                   |
                                   |             +--------------------+
                              +-------+          | Here is your ticket|
                              |  Call |          |and a SIP URI to    |
                              | Agent |          | reach this number. |
                              |       |          |                    |
                              |       | d.com    +--------------------+
                              +-------+            /
                                  |               /
                                                 /
                             //-------\\        /
               +===================================+
               V         |    Internet     |       ^
            +-------+     |\\           //|    +-------+
            |  Call |        \\-------//       |  Call |
    //\\    | Agent |--                      --| Agent |    //\\
     \/  ---|       |        //-------\\       |       |---- \/
            |       |     |//           \\|    |       |
            +-------+    |      PSTN       |   +-------+
                          |\\           //|
              a.com          \\-------//         b.com

                                  |
                              +-------+
                              |  Call |
                              | Agent |
                              |       |
                              |       |
                              +-------+
                                  |     c.com
                                  |
                                 //\\
                                  \/


                    Figure 4: Ticket Validation Step 2

   When a.com generates a SIP INVITE, it will contain this ticket.  The
   INVITE arrives at the b.com call agent over the mutually
   authenticated TLS connection established between the domains.

   The b.com call agent looks for the SIP header field in the INVITE
   that contains the ticket.  First, it verifies the signature over the
   ticket.  Remember that the b.com agent is the one that generated the
   ticket in the first place; as such, it is in possession of the key



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   required to validate the signature.  Once validated, it performs two
   checks:

   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 b.com 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.  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.  Of course, once that PSTN call had been placed, the
   spammer would have already achieved their goals, and at cost.  The
   additional VoIP call is not so exciting.

   This property also means that, in order for an attacker to spam call
   numbers on VoIP, it must have already spam-called those same numbers
   on the PSTN.  This means that the attacker would clearly be subject
   to regulations and laws governing usage of the PSTN for calling.  As
   an example, a spammer in the United States would have already
   violated U.S. do-not-call rules by initiating the spam calls to the
   PSTN numbers.

   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.  This is, again, an
   example of the incremental deployability philosophy that ViPR takes -



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   let not the perfect be the enemy of the good.


6.  Architecture Components and Functions

   The architecture in Figure 1 is overly simplistic.  ViPR allows the
   functionality embedded within the call agent can be split up into
   three components, as shown in Figure 5:

                        +-+            +-+
                        | |            | |   +------+
                        | |      +-----| |---|Enroll|
                        | |      |     | |   +------+
                        |I|      |     |I|
                        |n|   +-----+  |n|
                 VAP    |t|   | ViPR|  |t|
             +----------|r|---|Srvr |--|e|-----------------
             |          |a|   |     |  |r|   P2P-Validation
             |          |n|   +-----+  |n|
             |          |e|            |e|
             |          |t|            |t|
          +-----+  SIP  | |   +-----+  | |
          | CA  |-------|F|---|     |--|F| ---------------
          +-----+       |i|   |     |  |i|  SIP/TLS
             .          |r|   |  .  |  |r|
      SIP/   .          |e|   |     |  |e|
      MGCP/  .          |w|   | BE  |  |w|
      TDM    .          |a|   |     |  |a|
             .          |l|   |     |  |l|
          +-----+       |l|   |     |  |l|
          | UA  |-------| |---|     |--| |-----------------
          +-----+       | |   +-----+  | |   SRTP
                        | |            | |
                        +-+            +-+
     |                                      |
     +--------------------+-----------------+
                          |
             Single administrative domain

                          Figure 5: Architecture

   Within each domain, there are three components that are ViPR-aware.
   These are the ViPR server, the call agent (CA), and the border
   element (BE).  Outside of the domain, there is a P2P network and an
   enrollment server.  A domain will typically have firewalls - an
   Internet firewall and an intranet firewall.

   The sections which follow describe the roles and responsibilities of



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   each component in more detail.

6.1.  ViPR Server

   The ViPR server is the heart of the system.  It performs several key
   functions:

   1.  It implements the P2P protocol, acting as one or more nodes in
       the DHT.  By placing this function separate from the call agent,
       it allows the call agent to be isolated from the traffic and
       security concerns that are often associated with a P2P network.
   2.  It implements the validation mechanism.  It is informed of call
       events by the call agent, and sometime after the call, looks up
       the number in the DHT, and if found, attempts to connect to the
       node claiming ownership of the number, and then validates it.
   3.  It pushes newly learned routes to the call agent once validation
       has occurred.  The ViPR server does not hold the call routes;
       this eliminates the need for an off-box query to perform call
       routing logic.
   4.  It stores numbers into the DHT.  The call agent informs the ViPR
       servers of numbers to be published, and the ViPR server places
       them into the P2P network.  Refreshing the stored numbers (by
       asking the ViPR server to restore them) is the responsibility of
       the call agent.
   5.  It implements a distributed quota enforcement algorithm, ensuring
       that malicious ViPR servers cannot store excessive data into the
       network.
   6.  It implements a policing function, pacing its store and fetch
       requests into the DHT to ensure that the network is not
       overwhelmed.

   In order to join the P2P network and be able to receive incoming
   validation requests, the ViPR server must have open access to the
   public Internet.  For this reason, it is typically placed into the
   DMZ.  The Internet firewall will require two pinholes to be opened
   towards the ViPR server:  one for the P2P protocol, and one for the
   validation protocol.

   It is important to understand that the ViPR server does not perform
   any call processing.  It does not process SIP or RTP traffic.  It is
   a non-real-time server that performs validation processing in the
   background, outside of actual call attempts.

   The ViPR server needs to connect with the call agent.  This is done
   through the ViPR Access Protocol (VAP).  VAP is described in more
   detail below.





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6.2.  Call Agent

   The call agent is a box within the domain which performs call
   processing on behalf of one or more phones within the domain.  ViPR
   can work with a wide variety of call agents, as long as they meet
   some specific criteria:

   o  The call agent must be know of the start time, stop time, caller
      ID, and called numbers of calls placed from phones towards the
      PSTN.
   o  The call agent must be capable of making routing decisions for
      outbound calls from phones that would otherwise go to the PSTN,
      directing them towards the PSTN or towards other domains (based on
      ViPR routing rules).

   Based on this definition, many different types of products typically
   found within a domain could act as the call agent.  An IP PBX or TDM
   PBX with a SIP interface can be the call agent.  A Session Border
   Controller (SBC) that connects calls from a PBX to the PSTN, can act
   as the call agent.  An IMS application server can act as the call
   agent.  A PSTN gateway, used for all calls egressing a domain from a
   set of phones, can act as a call agent.

   A SIP proxy can act as a call agent; as long as it is capable of
   stashing the relevant call information into Record-Route headers for
   usage at the end of the call, it can even operate without retaining
   call state.

   A single phone can also act as the call agent, representing itself
   and its own phone number.

   In ViPR, the call agent performs several key functions specific to
   ViPR:

   o  It informs the ViPR server of the phone numbers to be stored in
      the DHT for its domain.
   o  It refreshes those numbers in the DHT, redoing the storage
      operation periodically.
   o  At the end of a call, the call agent sends a ViPR Call Record
      (VCR) to the ViPR server, containing the start time, stop time,
      caller ID and called party number.
   o  It learns validated routes from the ViPR server.  These routes
      consist of a phone number, a SIP URI to utilize when contacting
      that phone number, and a corresponding ticket.  The call agent is
      responsible for storing those routes.
   o  When a call is to be made towards a PSTN number, the call agent is
      responsible for checking whether there is a route for that number,
      learned via a prior notification from the ViPR server.  If so, it



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      is responsible for sending the INVITE towards the learned SIP URI,
      and for including the ticket the X-Cisco-ViPR-Ticket header field.

   Those functions which require communications with the ViPR server are
   done by implementing VAP.  VAP is a client-server protocol, with the
   call agent acting as the client, and the ViPR server acting as the
   server.  For this reason, the call agent is sometimes called the VAP
   client or ViPR client.

6.3.  Border Element

   The border element is responsible for the SIP layer perimeter
   security functions.  In particular:

   o  The border element ensures that all egress SIP traffic is carried
      over TLS.  Border elements must reject any incoming SIP requests
      which are not over TLS.  SIP over TLS is mandatory-to-use in ViPR,
      and it must be performed using mutual TLS.
   o  The border element ensures that all egress RTP traffic is actually
      carried using SRTP.  If the traffic originated by the UA in the
      domain is inherently SRTP, the criteria is met.  However, many
      domains do not utilize SRTP internally, and if it is not used
      internally, the border element must convert to SRTP.  Similarly,
      the border element is responsible for rejecting any incoming SIP
      calls that are not set up with SRTP.  SRTP is mandatory in ViPR.
   o  The border element ensures that ingress and egress SIP traffic is
      'fixed up' so that it can pass through the Internet firewall
      successfully.  Typically, this is done using a traditional SBC/ALG
      function.
   o  The border element inspects all incoming SIP INVITEs, and performs
      ticket verification.  In this process, it looks for the X-Cisco-
      ViPR-Ticket header field in the INVITE.  If not present, it
      discards the request.  If present, it verifies the signature, and
      then compares the called number and remote TLS domain against the
      contents of the ticket.  If they do not match, the border element
      discards the INVITE.

   The border element can perform other, non-ViPR tasks, as is common
   for border elements.  These include header inspection and validation,
   anti-virus checks on embedded content, SIP state machine conformance,
   policy checks on various services, and so on.

   The role of the border element can be fulfilled by any number of
   products typically found within domains.  These include Session
   Border Controllers and firewalls.  Indeed, the border element
   function can be embedded directly in the Internet firewall.

   The border element is connected to the call agent via SIP, and to the



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   user agent (UA) via RTP.  The border element has no direct connection
   to the ViPR server.  However, in order for ticket processing to work
   in this model, the ViPR server and border element must share a secret
   that is used to create the tickets.  This is discussed in more detail
   below.

6.4.  Enrollment Server

   P2P protocols - including RELOAD - require the usage of an enrollment
   server in order to obtain the certificates that are used to secure
   the network.  ViPR uses, and indeed requires, that all RELOAD traffic
   be over TCP/TLS with mutual authentication.  The certificates used
   are obtained through an enrollment process.  The details on how P2P
   enrollment are done are beyond the scope of this document.

6.5.  P2P Network

   The collection of ViPR servers form a single, worldwide, P2P network
   utilizing RELOAD and the Chord algorithm.

   It is very important to understand that the DHT is never accessed in
   real-time.  It is not queried at call setup time.  This is because
   the DHT is slow, involving many hops.  Queries could take seconds.
   Furthermore, we don't want to rely on proper operation of the DHT to
   actually make calls.


7.  Protocols

   The overall ViPR solution utilizes several protocols, each performing
   a different function.

7.1.  P2P: RELOAD

   ViPR utilizes the RELOAD protocol [P2PSIP-BASE] to run amongst each
   of the ViPR servers.  Each ViPR server acts as one or more nodes in
   the DHT.  The number of nodes that the ViPR server implements
   directly determines the quota allocated to that ViPR server, and in
   turn, the amount of work it must perform storing data.

   ViPR, however, does not implement the SIP usage that has been defined
   for RELOAD [P2PSIP-SIP].  That is because the DHT is not used as a
   traditional distributed registrar.  Instead, it implements a new
   usage - the ViPR usage - which stores phone numbers.  It also
   utilizes the DHT for storage of certificates, using a certificate
   usage.





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7.1.1.  ViPR Usage

   The ViPR usage is described in detail in [VIPR-RELOAD-USAGE].  This
   section provides a brief overview.

   The ViPR usage makes use of the dictionary type.  Each resource-ID is
   a key, computed by taking the SHA1 hash of an E.164 formatted phone
   number.  The value stored at this resource-ID is a dictionary.  The
   dictionary entries are the set of virtual ViPR servers which claim
   ownership of those numbers.

   Since a ViPR server might support a multiplicity of call agents from
   different domains, it is necessary to logically segment a ViPR server
   so that - from a security perspective - it operates logically like
   different virtual ViPR servers, one for each call agent.  Each
   virtual instance of a ViPR server is called a VService.  Thus, the
   entries in the dictionary are key value pairs whose key is the
   concatenation of the Node-ID and an identifier for the VService
   within that node.  The value at each key is the Node-ID to contact
   for validation.

   When a node in the DHT receives a Store request, and it is the
   responsible node for the resource-ID, it will verify that the Node-ID
   in both the key and value of the dictionary entry match the Node-ID
   in the certificate it presents.  This ensures that one ViPR server
   can never overwrite data from another ViPR server.

   The ViPR usage also specifies a quota mechanism.  Unlike the SIP
   usage, where there are very specific rules about what resource-IDs a
   node may store into the DHT, with ViPR, there is no way to restrict
   what resource-IDs may be stored by a ViPR server.  This is because,
   in ViPR, the resource-IDs are derived from phone numbers, and at the
   time of storage, there is no way to know whether the node performing
   the store actually owns this phone number.  Consequently, a
   responsible node will accept stores from any node for any
   resource-ID.  However, to limit malicious users from consuming all of
   the resources of the DHT, the ViPR usage imposes a quota on storage.
   Each node performing a store is allocated a fixed quota on the number
   of records it can place into the DHT.  A probabilistic enforcement
   model is utilized at each responsible node based on the fraction of
   the hashspace owned by that responsible node.  Roughly speaking, if
   the system quota is 10,000 phone numbers per Node-ID, if a
   responsible node owns 10% of the DHT, it will accept an average of
   1000 phone numbers from any one single Node-ID.







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7.1.2.  Certificate Usage

   Further details pending.

7.2.  ViPR Access Protocol (VAP)

   The ViPR Access Protocol (VAP) is documented in [VIPR-VAP].

   VAP is a client-server protocol that runs between the call agent and
   the ViPR server.  VAP is a simple, binary based, request/response
   protocol.  It utilizes the same syntactic structure and transaction
   state machinery as STUN [RFC5389], but otherwise is totally distinct
   from it.  VAP clients initiate TCP/TLS connections towards the ViPR
   server.  The ViPR server never opens connections towards the call
   agent.  This allows the ViPR servers to run on the public side of
   NATs and firewalls.

   Once the connections are established, the call agent sends a Register
   message to the ViPR server.  This register message primarily provides
   authentication and connects the client to the ViPR server.  VAP
   provides several messages for different purposes:
   o  Publish:  The Publish message informs the ViPR server of service
      information.  There are two types of Publishes supported in ViPR.
      The first is the ViPR Service (VService).  This informs the ViPR
      server of the SIP URIs on the call agent and black and white lists
      used by the ViPR server to block validations.  The ViPR server
      stores that information locally and uses it during the validation
      process, as described above.  The second Publish is the ViPR
      number service.  The ViPR server, upon receiving this message,
      performs a Store operation into the DHT.
   o  UploadVCR:  This message comes in two flavors - an originating and
      terminating message.  An originating UploadVCR comes from a call
      agent upon completion of a non-ViPR call to the PSTN.  A
      terminating UploadVCR comes from an agent upon completion of a
      call received FROM the PSTN.  The ViPR server behavior for both
      messages is very different.  For Originating UploadVCR, the ViPR
      server will store these, and at a random time later, query the DHT
      for the called number and attempt validation against the ViPR
      servers that are found.  For a terminating UploadVCR, the ViPR
      server will store these, awaiting receipt of a validation against
      them.
   o  Subscribe:  Call agents can subscribe for information from the
      ViPR server.  There is one service that the call agent can
      subscribe for:  number Service.  When a new number is validated,
      the ViPR server will send a Notify to the call agent, containing
      the validated number, the ticket, and a set of SIP trunk URIs.





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   o  Notify:  The ViPR server sends this message to the call agent when
      it has an event to report for a particular subscription.

   The VAP protocol provides authentication by including an integrity
   object in each message.  This integrity message is the hash of the
   contents of the message and a shared secret between the ViPR server
   and the client.  VAP can also be run over TLS, which enhances
   security further.

   The P2P network introduces rate limits for the purposes of
   performance management and limiting denial of service attacks.  Each
   node in the DHT comes with it a limit on the amount of stores per
   second, reads per second, and total amount of data it can store in
   the DHT.  The ViPR server rigorously follows those limits.

   As a consequence, when numbers are stored into the DHT, they are
   written in slowly based on the rate limits.  The call agent will send
   a Publish operation for each individual number.  The ViPR server will
   perform the store in a rate-limited fashion.  When the store is
   complete, the ViPR server responds to the Publish, and the call agent
   can move to the next DID to publish.  Thus, it may take hours or even
   days to fully store the set of numbers into the DHT.  The process
   then repeats several days later in order to refresh the data in the
   DHT.

7.3.  Validation Protocol

   The core of ViPR is the validation protocol.  The validation protocol
   is used by one ViPR server to connect to another, demand proof-of-
   knowledge of a previous PSTN call, and once proven, securely learn a
   SIP URI and ticket for usage in future SIP calls between domains.

   The validation protocol is documented in [VIPR-PVP].

   The validation protocol is built using TLS-SRP [RFC5054].  TLS-SRP
   creates a secure TLS connection, but instead of using certificates,
   utilizes a password.  TLS-SRP was designed for cases where the
   passwords are relatively weak.  In the case of the validation
   protocol, the passwords are formed from parameters of a previous PSTN
   call.  Once a secure TLS connection is formed, a simple request/
   response protocol is run over it.  The request contains the domain
   name of the originating ViPR server, and the response contains the
   SIP URI and ticket for that number.

   The validation protocol properly handles time offsets between the two
   domains for the start and stop times of the calls, the relatively
   weak entropy of a single phone call, the grand chessmaster attack,
   and non-delivery or inaccurate delivery of caller-ID, amongst other



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   issues.  The validation protocol can be tuned by administrators to
   allow for arbitrary levels of security, measured in terms of
   equivalent entropy.  The equivalent entropy is the number of bits of
   entropy that must be demonstrated, as if the domains were
   authenticating each other using a password with that amount of
   entropy.  This gives domains a 'nerd knob' they can turn to trade off
   security for performance.

   Because the validation protocol utilizes TLS-SRP, it does not run
   directly through the DHT.  This is why a ViPR server requires a
   separate pinhole to be opened for the validation protocol.

7.4.  SIP Extensions

   The connection between the call agents in different domains is SIP.
   ViPR requires that the inter-domain connections run over TLS, and
   furthermore, utilize SRTP keyed with Sdescriptions.

   ViPR extends SIP with its anti-spam mechanism.  This takes the form
   of a ticket, present in a SIP header field.  [VIPR-SIP-ANTISPAM]
   defines this header field and the format of the ticket it contains.


8.  Example Call Flows

   This section provides call flows for the key use cases.

8.1.  PSTN Call and VCR Upload

   A call flow for the initial PSTN call and VCR upload is shown in
   Figure 6.

   Alice    CA+O     GW+O    VIPR+O    GW+T     CA+T    VIPR+T     Bob
     |(1) Call NumX    |        |        |        |        |        |
     |------->|        |        |        |        |        |        |
     |        |(2) INVITE NumX  |        |        |        |        |
     |        |------->|        |        |        |        |        |
     |        |        |(3) setup NumX   |        |        |        |
     |        |        |---------------->|        |        |        |
     |        |        |        |        |(4) INVITE NumX  |        |
     |        |        |        |        |------->|        |        |
     |        |        |        |        |        |(5) Call NumX    |
     |        |        |        |        |        |---------------->|
     |        |        |        |        |        |        |     Answers
     |        |        |        |        |        |(6) answer       |
     |        |        |        |        |        |<----------------|
     |        |        |        |        |(7) 200 OK       |        |
     |        |        |        |        |<-------|        |        |



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     |        |        |        |        |(8) ACK |        |        |
     |        |        |        |        |------->|        |        |
     |        |        |(9) answer       |        |        |        |
     |        |        |<----------------|        |        |        |
     |        |(10) 200 OK      |        |        |        |        |
     |        |<-------|        |        |        |        |        |
     |        |(11) ACK|        |        |        |        |        |
     |        |------->|        |        |        |        |        |
     |(12) accept      |        |        |        |        |        |
     |<-------|        |        |        |        |        |        |
     |hangs up|        |        |        |        |        |        |
     |(13) hangup      |        |        |        |        |        |
     |------->|        |        |        |        |        |        |
     |        |(14) BYE|        |        |        |        |        |
     |        |------->|        |        |        |        |        |
     |        |(15) 200 OK      |        |        |        |        |
     |        |<-------|        |        |        |        |        |
     |        |        |(16) hangup      |        |        |        |
     |        |        |---------------->|        |        |        |
     |        |        |        |        |(17) BYE|        |        |
     |        |        |        |        |------->|        |        |
     |        |        |        |        |(18) 200 OK      |        |
     |        |        |        |        |<-------|        |        |
     |        |        |        |        |        |(19) hangup      |
     |        |        |        |        |        |---------------->|
     |        |(20) Orig UploadVCR       |        |        |        |
     |        |---------------->|        |        |        |        |
     |        |(21) Success     |        |        |        |        |
     |        |<----------------|        |        |        |        |
     |        |        |        |Set timer        |        |        |
     |        |        |        |        |        |(22) Term UploadVCR
     |        |        |        |        |        |------->|        |
     |        |        |        |        |        |(23) Success     |
     |        |        |        |        |        |<-------|        |


                      Figure 6: PSTN Call and Upload

   In message 1, Alice calls the number of her colleague, Bob. This is
   NumX.  This call is routed over the PSTN, through the terminating
   call agent, and rings Bob's phone (messages 1-5).  Bob answers the
   phone, and this is propagated back to Alice (messages 6-12).  Bob and
   Alice talk for a while, and then Alice hangs up.  This hangup is
   propagated to Bob, and the call is terminated (messages 13-19).

   The originating call agent notes that this call went to the PSTN, and
   might be a candidate for a future SIP call.  It sends an UploadVCR
   message to its ViPR server (message 20), containing the start time,



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   stop time, callerID and called party number.  The ViPR server
   acknowledges this (message 21), and then sets a timer for a random
   time into the future, at which point it will attempt validation.  The
   terminating side is similar; it sends an UploadVCR to its ViPR server
   (message 22), which is acknowledged (message 23).  The terminating
   side does not set a timer; it waits for a possible validation attempt
   which may or may not arrive in the future.

8.2.  DHT Query and Validation

   This section provides the call flow for what happens on the
   originating ViPR server when the timer fires, in Figure 7.

  CA+O                VIPR+O                  DHT                 VIPR+T
    |                    |timer fires          |                     |
    |                    |(1) Query NumX       |                     |
    |                    |-------------------->|                     |
    |                    |(2) Node-ID T        |                     |
    |                    |<--------------------|                     |
    |                    |(3) Connect Node-ID T|                     |
    |                    |-------------------->|                     |
    |                    |                     |(4) Connect Node-ID T|
    |                    |                     |-------------------->|
    |                    |                     |(5) Connect resp.    |
    |                    |                     |<--------------------|
    |                    |(6) Connect resp.    |                     |
    |                    |<--------------------|                     |
    |                    |(7) TCP Connect      |                     |
    |                    |------------------------------------------>|
    |                    |(8) TLS-SRP          |                     |
    |                    |------------------------------------------>|
    |                    |(9) ValExchange(a.com)                     |
    |                    |------------------------------------------>|
    |                    |(10) ValResponse(URI, ticket)              |
    |                    |<------------------------------------------|
    |(11) Notify(NumX,URI,ticket)              |                     |
    |<-------------------|                     |                     |
    |Store route         |                     |                     |


                         Figure 7: Validation Flow

   First, the timer that was set by the originating ViPR server in
   Figure 6 fires.  When it fires, the ViPR server examines the called
   party number from the VCR.  It performs a query into the DHT, to see
   if this number has been stored by any domain (message 1).  In this
   case, it has, and the DHT returns with a successful query response
   (message 2).  This response indicates that the terminating ViPR



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   server, with node-ID T, claims ownership of the number.

   The originating ViPR server asks the DHT to form a connection between
   itself and the terminating ViPR server.  This message exchanges IP
   addresses and ports through which a TCP connection can be attempted;
   details are omitted (messages 3-6).  Now, the originating ViPR server
   can establish a TCP connection to the terminating ViPR server
   (message 7).  Next, the originating ViPR server begins negotiation of
   a TLS-SRP connection.  The TLS-SRP uses the caller ID and called
   number as a "username" for this exchange, and the start time and stop
   time of the call as a password.  As both sides share the same values
   for this secret, the secure connection is established.  This is now a
   TLS connection between the two ViPR servers.

   Over this secure connection, the originating ViPR server sends a
   ValExchange request.  This request contains the domain name that is
   claimed by the originating ViPR server (this claim is not verified at
   this time) (message 9).  This is received by the terminating ViPR
   server, which then creates a ticket for that domain and NumX, and
   passes the ticket and the SIP URI back to the originating ViPR server
   (message 10).  The originating ViPR server sends this information to
   its call agent (message 11), which then stores it for usage in a
   future call.

8.3.  DHT Query and No Match

   In this case, after the PSTN call of Figure 6, the timer fires, but
   the originating ViPR server finds no match in the DHT.  This is an
   alternative case to the flow in Figure 7.

   CA+O                VIPR+O                 DHT                VIPR+T
     |                    |timer fires         |                    |
     |                    |(1) Query NumX      |                    |
     |                    |------------------->|                    |
     |                    |(2) noMatch         |                    |
     |                    |<-------------------|                    |

                          Figure 8: DHT No-Match

8.4.  SIP Call

   In this case, shown in Figure 9, a user makes a call to a number
   which has been learned via ViPR.








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   Alfred      CA+O          BE+O          BE+T          CA+T        Bob
   |(1) Call X   |             |             |             |           |
   |------------>|             |             |             |           |
   |             |(2) INVITE X |             |             |           |
   |             |Ticket       |             |             |           |
   |             |------------>|             |             |           |
   |             |             |(3) TCP and TLS            |           |
   |             |             |w domain certs             |           |
   |             |             |------------>|             |           |
   |             |             |(4) INVITE X |             |           |
   |             |             |Ticket       |             |           |
   |             |             |------------>|             |           |
   |             |             |             |Validate Ticket          |
   |             |             |             |(5) INVITE X |           |
   |             |             |             |Ticket       |           |
   |             |             |             |------------>|           |
   |             |             |             |             |(6)INVITE X|
   |             |             |             |             |---------->|

                            Figure 9: SIP Call

   First, a user in the originating domain - Alfred - calls Bob's number
   (message 1).  The originating call agent notes that it has a cached
   route for that number.  It extracts the SIP URI, using it as the
   topmost Route header field, and then attaches the ticket to the
   X-Cisco-ViPR-Ticket header field.  This INVITE is sent to a default
   next hop border element (message 2).  The border element establishes
   a TCP/TLS connection with the domain in the Route header.  It uses a
   traditional domain certification for this TLS connection (message 3).
   Once established, it sends the INVITE over the connection (message
   4).

   This arrives at the terminating call agent, which extracts the ticket
   and verifies it.  To verify it, it checks the signature using the key
   that was used to create the ticket.  Then, it compares the domain
   name in the ticket with the domain name from the TLS connection
   handshake.  Finally, it compares the called party number in the
   Request-URI with the value from the ticket.  Assuming they all match,
   the call is forwarded to the terminating call agent (message 5),
   where it is finally delivered to Bob (message 6).


9.  Security Considerations

   Security is incredibly important for ViPR.  This section provides an
   overview of some of the key threats and how they are handled.





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9.1.  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 routes that are not already known.  If, by some tragedy,
   an attacker destroyed the P2P network completely, it would not cause
   a single call to fail.  Furthermore, it would not cause calls to
   revert to the PSTN - calls to routes learned previously would still
   go over the IP network.  The only impact to such a devastating
   attack, is that a domain could not learn *new* routes to 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.  Other attacks are prevented by mechanisms defined
   by RELOAD itself, and are not ViPR specific.

9.2.  Theft of Phone Numbers

   The key security threat that ViPR is trying to address is the theft
   of phone numbers.  In particular, a malicious domain could store,
   into the 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, which performs a proof of knowledge
   check to verify ownership of numbers.

   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



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   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
   domain.

9.3.  Spam

   Another serious concern is that attackers may try to launch VoIP spam
   (also known as SPIT) calls into a domain.  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
   applies those same protections to SIP connections.

   As noted above, 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.

   Of course, ViPR's built-in mechanism is not a guarantee.  A SPIT
   clearinghouse could shoulder the costs of the PSTN calls, and then
   re-sell its access for a fee.  However, this still causes the
   clearinghouse to utilize non-trivial resources in its attack.  Though
   these costs are less than the PSTN, they are more than zero, and
   should act as a deterrent for a long while.

9.4.  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.



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10.  IANA Considerations

   This specification does not require any actions from IANA.


11.  Acknowledgements

   Thanks to Theo Zourzouvillys for pointing out the fifth thief of
   phone numbers attack.


12.  References

12.1.  Normative References

   [P2PSIP-BASE]
              Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and
              H. Schulzrinne, "REsource LOcation And Discovery (RELOAD)
              Base Protocol", draft-ietf-p2psip-base-11 (work in
              progress), October 2010.

   [P2PSIP-SIP]
              Jennings, C., Lowekamp, B., Rescorla, E., Baset, S., and
              H. Schulzrinne, "A SIP Usage for RELOAD",
              draft-ietf-p2psip-sip-05 (work in progress), July 2010.

   [VIPR-RELOAD-USAGE]
              Rosenberg, J., Jennings, C., and M. Petit-Huguenin, "A
              Usage of Resource Location and Discovery (RELOAD) for
              Public Switched Telephone Network (PSTN) Verification",
              draft-rosenberg-dispatch-vipr-reload-usage-03 (work in
              progress), October 2010.

   [VIPR-SIP-ANTISPAM]
              Rosenberg, J., Jennings, C., and M. Petit-Huguenin,
              "Session Initiation Protocol (SIP) Extensions for Blocking
              VoIP Spam Using PSTN Validation",
              draft-rosenberg-dispatch-vipr-sip-antispam-03 (work in
              progress), October 2010.

   [VIPR-VAP]
              Rosenberg, J., Jennings, C., and M. Petit-Huguenin,
              "Verification Involving PSTN Reachability: The ViPR Access
              Protocol (VAP)", draft-rosenberg-dispatch-vipr-vap-03
              (work in progress), October 2010.

   [VIPR-PVP]
              Rosenberg, J., Jennings, C., and M. Petit-Huguenin, "The



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              Public Switched Telephone Network (PSTN) Validation
              Protocol (PVP)", draft-rosenberg-dispatch-vipr-pvp-03
              (work in progress), October 2010.

12.2.  Informative References

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

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

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

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

   [RFC3761]  Faltstrom, P. and M. Mealling, "The E.164 to Uniform
              Resource Identifiers (URI) Dynamic Delegation Discovery
              System (DDDS) Application (ENUM)", RFC 3761, April 2004.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [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.

   [CLF-SYNTAX]
              Roach, A., "Binary Syntax for SIP Common Log Format",
              draft-roach-sipping-clf-syntax-01 (work in progress),
              May 2009.

   [SESSION-ID]
              Kaplan, H., "A Session Identifier for the Session
              Initiation Protocol (SIP)", draft-kaplan-sip-session-id-02
              (work in progress), March 2009.





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Appendix A.  Release notes

   This section must be removed before publication as an RFC.

A.1.  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

   Jonathan Rosenberg
   jdrosen.net
   Monmouth, NJ
   US

   Email:  jdrosen@jdrosen.net
   URI:    http://www.jdrosen.net


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

   Phone:  +1 408 421-9990
   Email:  fluffy@cisco.com


   Marc Petit-Huguenin
   Stonyfish

   Email:  marc@stonyfish.com












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