SPEERMINT Working Group                                        J-F. Mule
Internet-Draft                                                 CableLabs
Expires: April 26, 2007                                 October 23, 2006

       SPEERMINT Requirements for SIP-based VoIP Interconnection

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Copyright Notice

   Copyright (C) The Internet Society (2006).


   This document describes high-level guidelines and general
   requirements for Session PEERing for Multimedia INTerconnect.  It
   also defines a minimum set of requirements applicable to session
   peering for Voice over IP interconnects.  It is intended to become
   best current practices based on the use cases discussed in the
   speermint working group.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  General Requirements . . . . . . . . . . . . . . . . . . . . .  5
   4.  Requirements for SIP-based VoIP Interconnection  . . . . . . .  8
     4.1.  DNS, Call Addressing Data (CAD) and ENUM . . . . . . . . .  8
     4.2.  Minimum set of SIP-SDP-related requirements  . . . . . . .  8
     4.3.  Media-related Requirements . . . . . . . . . . . . . . . .  9
     4.4.  Security Requirements  . . . . . . . . . . . . . . . . . .  9
       4.4.1.  Security in today's VoIP networks  . . . . . . . . . .  9
       4.4.2.  TLS Considerations for session peering . . . . . . . . 10
   5.  Annex A - List of Policy Parameters for VoIP
       Interconnections . . . . . . . . . . . . . . . . . . . . . . . 12
     5.1.  Categories of parameters and Justifications  . . . . . . . 12
     5.2.  Summary of Parameters for Consideration in Session
           Peering Policies . . . . . . . . . . . . . . . . . . . . . 14
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 16
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 18
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21
   Intellectual Property and Copyright Statements . . . . . . . . . . 22

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1.  Introduction

   The Session PEERing for Multimedia INTerconnect (SPEERMINT) Working
   Group is chartered to focus on architectures to identify, signal, and
   route delay-sensitive communication sessions.  These sessions use the
   Session Initiation Protocol (SIP) protocol to enable peering between
   two or more administrative domains over IP networks.

   This document describes high-level guidelines and general
   requirements for session peering; these requirements are applicable
   to any type of multimedia session peering such as Voice over IP
   (VoIP), video telephony, and instant messaging.  The document also
   defines a minimum set of requirements for a sub-set of the session
   peering use cases: VoIP interconnects.

   The intent of this version of this document is to describe what
   mechanisms are used for establishing SIP session peering with a
   special look at VoIP interconnects, and in doing so, it defines some
   of requirements associated with the secure establishment of VoIP
   interconnects between a large number of peers.
   The primary focus is on the requirements applicable to the boundaries
   of layer-5 SIP networks: SIP UA or end-device requirements are
   considered out of scope.
   It is also not the goal of this document to mandate any particular
   use of any IETF protocols to establish session peering by users or
   service providers.  However, when protocol mechanisms are used, the
   document aims at providing guidelines or best current practices on
   how they should be implemented, or configured and enabled in order to
   facilitate session peering.

   Finally, a list of parameters for the definition of a session peering
   policy is provided in an informative annex.  It should be considered
   as an example of the information a Voice Service Provider, or
   Application Service Provider may require in order to connect to
   another using SIP.

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2.  Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   and "OPTIONAL" are to be interpreted as described in RFC 2119

   This specification makes use of terms defined in
   [I-D.ietf-speermint-terminology], the Session Description Protocol
   (SDP) [RFC4566] and the Session Initiation Protocol (SIP) [RFC3261].
   We also use the terms Voice Service Provider (VSP) and Application
   Service Provider (ASP) as defined in [I-D.ietf-ecrit-requirements].

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3.  General Requirements

   The following section defines general guidelines and requirements
   applicable to session peering for multimedia sessions.

   o  Session peering should be independent of lower layers.  The
      mechanisms used to establish session peering SHOULD accommodate
      diverse supporting lower layers.

      Session peering is about layer 5 mechanisms.  It should not matter
      whether lower layers rely on the public Internet or are
      implemented by private L3 connectivity, using firewalls or L2/L3
      Virtual Private Networks (VPNs), IPSec tunnels or Transport Layer
      Security (TLS) connections [RFC3546]...

   o  Session Peering Policies and Extensibility:
      Policies developed for session peering SHOULD be flexible and
      extensible to cover existing and future session peering models.
      It is also RECOMMENDED that policies be published via local
      configuration choices in a distributed system like DNS rather than
      in a centralized system like a 'peering registry'.
      In the context of session peering, a policy is defined as the set
      of parameters and other information needed by one VSP/ASP to
      connect to another.  Some of the session policy parameters may be
      statically exchanged and set throughout the lifetime of the
      peering relationship.  Others parameters may be discovered and
      updated dynamically using by some explicit protocol mechanisms.
      These dynamic parameters may also relate to a VSP/ASP's session-
      dependent or session independent policies as defined in

      It is critical that the solutions be flexible and extensible given
      the various emerging models: layer 5 peering may involve open
      federations of SIP proxies, or closed environments with systems
      that only accept incoming calls from selected peers based on a set
      of bilateral trust relationships.  Federations may also be based
      on memberships in peering fabrics or voice service provider clubs,
      etc.  Session peering may be direct or indirect.
      The maintenance of the "system" should scale beyond simple lists
      of peering partners.  In particular, it must incorporate
      aggregation mechanisms which avoid O(n^2) scaling (where n is the
      number of participating peers).  The distributed management of the
      DNS is a good example for the scalability of this approach.

   o  Administrative and Technical Policies:
      Various types of policy information may need to be discovered or

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      exchanged in order to establish session peering.  At a minimum, a
      policy SHOULD specify information related to call addressing data
      in order to avoid session establishment failures.  A policy MAY
      also include information related to QoS, billing and accounting,
      layer-3 related interconnect requirements which are out of the
      scope of this document.

      The reasons for declining or accepting incoming calls from a
      prospective peering partner can be both administrative
      (contractual, legal, commercial, or business decisions) and
      technical (certain QoS parameters, TLS keys, domain keys, ...).
      The objectives are to provide a baseline framework to define,
      publish and optionally retrieve policy information so that a
      session establishment does not need to be attempted to know that
      imcompatible policy parameters will cause the session to fail
      (this was originally referred to as "no blocked calls").

   o  URIs and Domain-Based Peering Context:
      Call Addressing Data SHOULD rely on URIs (Uniform Resource
      Identifiers, RFC 3986 [RFC3986]) for call routing and SIP URIs
      SHOULD be preferred over tel URIs (RFC 3966 [RFC3966]).  Although
      the initial call addressing data may be based on E.164 numbers for
      voice interconnects, a generic peering methodology SHOULD NOT rely
      on such E.164 numbers.

      Telephone numbers commonly appear in the username portion of a SIP
      URI.  When telephone numbers are in tel URIs, SIP requests cannot
      be routed in accordance with the traditional DNS resolution
      procedures standardized for SIP as indicated in RFC 3824
      [RFC3824].  Furthermore, we assume that all SIP URIs with the same
      domain-part share the same set of peering policies, thus the
      domain of the SIP URI may be used as the primary key to any
      information regarding the reachability of that SIP URI.

   o  URI Reachability and Minimal additional cost on call initiation:
      Based on a well-known URI (for e.g. sip, pres, or im URIs), it
      MUST be possible to determine whether the domain servicing the URI
      (VSP/ASP) allows for session peering, and if it does, it SHOULD be
      possible to locate and retrieve the domain's policy and signaling
      functions.  For example, an originating service provider must be
      able to determine whether a SIP URI is open for direct
      interconnection without requiring to initiate a SIP request.
      Furthermore, since each call setup implies the execution of any
      proposed algorithm, the establishment of a SIP session via peering
      SHOULD incur minimal overhead and delay, and employ caching
      wherever possible to avoid extra protocol round trips.

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      This requirement is important as unsuccessful call attempts are
      highly undesirable since they can introduce high delays due to
      timeouts and can act as an unintended denial of service attack
      (e.g., by repeated TLS handshakes).  There should be a high
      probability of successful call completion for policy-conforming

   o  Variability of the Call Address Data:
      A terminating VSP/ASP or user SHOULD be able to indicate its
      domain ingress points (Signaling Path Border Element(s)) based on
      the identity of the originating VSP/ASP or user.
      The mechanisms recommended for the use and resolution of the call
      addressing data SHOULD allow for variability or customization of
      the response(s) depending on various elements, such as the
      identity of the originating or terminating user or user domain.

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4.  Requirements for SIP-based VoIP Interconnection

   This section defines some requirements for SIP-based VoIP
   Interconnection.  It should be considered as the minimal set of
   requirements to be implemented to perform SIP VoIP interconnects.

4.1.  DNS, Call Addressing Data (CAD) and ENUM

   Call Addressing Data can be derived from various mechanisms available
   to the user, such as ENUM when the input is a telephone number, or
   other DNS queries using SRV and NAPTR resource records when the entry
   is a SIP URI for example.  The SPEERMINT Working Group is focused on
   the use of CAD.

   The following requirements are best current practices for VoIP
   session peering:

   o  SIP URIs SHOULD be preferred over tel URIs when establishing a SIP
      session for voice interconnects.

   o  The recommendations defined in [RFC3824] SHOULD be followed by
      implementers when using E.164 numbers with SIP, and by authors of
      NAPTR records for ENUM for records with an 'E2U+sip' service
      field.  Other ENUM implementation issues and experiences are
      described in [I-D.ietf-enum-experiences] that may be relevant for
      VoIP interconnects using ENUM.

   o  The use of DNS domain names and hostnames is RECOMMENDED in SIP
      URIs and they MUST be resolvable on the public Internet.

   o  The DNS procedures specified in [RFC3263] SHOULD be followed to
      resolve a SIP URI into a reachable host (IP address and port), and
      transport protocol.  Note that RFC 3263 relies on DNS SRV
      [RFC2782] and NAPTR Resource Records [RFC2915].

4.2.  Minimum set of SIP-SDP-related requirements

   The main objective of VoIP interconnects being the establishment of
   successful SIP calls between peer VSPs/ASPs, this section provides a
   minimum set of SIP-related requirements.

   o  The Core SIP Specifications as defined in [RFC3261] and
      [I-D.ietf-sip-hitchhikers-guide] MUST be supported by Signaling
      Path Border Elements and any other SIP implementations involved in
      session peering.
      The specifications contained in the Core SIP group provide the
      fundamental and basic mechanisms required to enable VoIP

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      interconnects.  This includes: the SIP protocol for session
      establishment and its updates such as RFC 3853 and RFC 4320, SDP
      [RFC4566] and its Offer/Answer model [RFC3264] for VoIP media
      session descriptions and codec negotiations, SIP Asserted Identity
      for caller ID services, and various other extensions to support
      NAT traversal, etc.

   o  The following RFCs SHOULD be supported: Reliability of Provisional
      Responses in SIP - PRACK [RFC3262], the SIP UPDATE method (for
      e.g. for codec changes during a session) [RFC3311], the Reason
      header field [RFC3326].

   In the context of session peering where peers desire to maximize the
   chances of successful call establishment, the recommendations
   contained in RFC 3261 regarding the use of the Supported and Require
   headers MUST be followed.  Signaling Path Border Elements SHOULD
   include the supported SIP extensions in the Supported header and the
   use of the Require header must be configurable on a per target domain
   basis in order to match a network peer policy and to maximize

4.3.  Media-related Requirements

   VSPs engaged in session peering SHOULD support of compatible codecs
   and include media-related parameters in their domain's policy.
   Transcoding SHOULD be avoided by proposing commonly agreed codecs.

   Motivations: The media capabilities of a VSP's network are either a
   property of the SIP end-devices, or, a combination of the property of
   end-devices and Data Path Border Elements that may provide media
   transcoding.  The choice of one or more common codecs for VoIP
   sessions between VSPs is therefore outside the scope of speermint.
   Indeed, as stated in introduction, requirements applicable to end-
   devices of a VSP are considered out of scope.  A list of media-
   related policy parameters are provided in the informative Section 5.

4.4.  Security Requirements

4.4.1.  Security in today's VoIP networks

   In today's VoIP deployments, various approaches exist to secure
   exchanges between VSPs/ASPs.  Signaling and media security are the
   two primary topics for consideration in most deployments.  A number
   of transport-layer and network-layer mechanisms are widely used by
   some categories of VSPs: TLS in the enterprise networks for
   applications such as VoIP and secure Instant Messaging, IPSec and
   L2/L3 VPNs in some VSP networks where there is a desire to secure all
   signaling and media traffic at or below the IP layer.  Media level

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   security is not widely deployed for RTP, even though it is in use in
   few deployments where the privacy of voice communications is
   A detailed security threat analysis of session peering exchanges
   should provide more guidance on what scalable and efficient methods
   should be used to help mitigate the the main security risks in large-
   scale session peering.

   A recent IETF BoF at IETF 66 (rtpsec) was organized to analyze SIP
   requirements for SRTP keying; a number of security requirements for
   VoIP were discussed.  A few Internet-Drafts have since been released
   and focus on media security requirements for SIP sessions
   ([I-D.ietf-wing-media-security-requirements]).  Some of these
   scenarios may be applicable to interdomain VSP/ASP session peering or
   they may be augmented in the future by interdomain scenarios.

4.4.2.  TLS Considerations for session peering

   The remaining of Section 4 covers some details on how TLS could be
   deployed and used between 2 VSPs/ASPs to secure SIP exchanges.  The
   intent is to capture what two VSPs/ASPs should discuss and agree on
   in order to establish TLS connections for SIP session peering.

      1.  Peers SHOULD agree on one or more Certificate Authorities
      (CAs) to trust for securing session peering exchanges.
      A VSP/ASP should have control over which root CA it trusts for SIP
      communications.  This may imply creating a certificate trust list
      and including the peer's CA for each authorized domain.  This
      requirement allows for the initiating side to verify that the
      server certificate chains up to a trusted root CA.  This also
      means that SIP servers SHOULD allow the configuration of a
      certificate trust list in order to allow a VSP/ASP to control
      which peer's CAs are trusted for TLS connections.  Note that these
      considerations seem to be around two themes: one is trusting a
      root, the other is trusting intermediate CAs.

      2.  Peers SHOULD indicate whether their domain policies require
      proxy servers to inspect and verify the identity provided in SIP
      requests as defined in [RFC4474].

      3.  SIP servers involved in the secure session establishment over
      TLS MUST have valid X.509 certificates and MUST be able to receive
      a TLS connection on a well-known port.

      4.  The following TLS/SIP Protocol parameters SHOULD be agreed
      upon as part of session peering policies: the version of TLS
      supported by Signaling Border Elements (TLSv1, TLSv1.1), the SIP

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      TLS port (default 5061), the server-side session timeout (default
      300 seconds), the list of supported or recommended ciphersuites,
      and the list of trusted root CAs.

      5.  SIP servers involved in the session establishment over TLS
      MUST verify and validate the client certificates: the client
      certificate MUST contain a DNS or URI choice type in the
      subjectAltName which corresponds to the domain asserted in the
      host portion of the URI contained in the From header.  It is also
      recommended that VSPs/ASPs convey the domain identity in the
      certificates using both a canonical name of the SIP server(s) and
      the SIP URI for the domain as described in section 4 of
      [I-D.gurbani-sip-domain-certs].  On the client side, it is also
      critical for the TLS client to authenticate the server as defined
      in [RFC3261] and in section 9 of draft-ietf-sip-certs-01.txt.

      6.  A session peering policy SHOULD include details on SIP session
      establishment over TLS if TLS is supported.

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5.  Annex A - List of Policy Parameters for VoIP Interconnections

   This informative annex lists the various types of parameters that
   should be considered when discussing the technical aspects of a VoIP
   Peering policy .

5.1.  Categories of parameters and Justifications

   It is intended as an initial list of topics that should be addressed
   by peers when establishing a VoIP peering relationship.

   o  IP Network Connectivity:
      It is assumed that IP network connectivity exists between peers.
      While this is out of scope of session peering, VSPs must agree
      upon a common mechanism for IP transport of Layer 5 session
      signaling and media.  This may be accomplish via private (e.g.
      IPVPN, IPSEC, etc.) or public IP networks.

   o  Media-related Parameters:

      *  Media Codecs: list of supported media codecs for audio, real-
         time fax (version of T.38, if applicable), real-time text (RFC
         4103), DTMF transport, voice band data communications (as
         applicable) along with the supported or recommended codec
         packetization rates, level of RTP paylod redundancy, audio
         volume levels, etc.

      *  Media Transport: level of support for RTP-RTCP [RFC3550], RTP
         Redundancy (RTP Payload for Redundant Audio Data - [RFC2198]) ,
         T.38 transport over RTP, etc.

      *  Other: support of the VoIP metric block as defined in RTP
         Control Protocol Extended Reports [RFC3611] , etc.

   o  SIP:

      *  A session peering policy SHOULD include the list of supported
         and required SIP RFCs, supported and required SIP methods
         (including p headers if applicable), error response codes,
         supported or recommended format of some header field values ,

      *  It should also be possible to describe the list of supported
         SIP RFCs by various functional groupings.  A group of SIP RFCs
         may represent how a call feature is implemented (call hold,
         transfer, conferencing, etc.), or it may indicate a functional
         grouping as in [I-D.ietf-sip-hitchhikers-guide].

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   o  Accounting:
      Call accounting may be required for tracking session usage on a
      peer's network.  It is critical for peers to determine whether the
      support of any SIP extensions for accounting is a pre-requisite
      for SIP interoperability.  In some cases, call accounting may feed
      data for billing purposes but not always: some operators may
      decide to use accounting as a 'bill and keep' model to track
      session usage and monitor usage against service level agreements.
      [RFC3702] defines the terminology and basic requirements for
      accounting of SIP sessions.  A few private SIP extensions have
      also been defined and used over the years to enable call
      accounting between VSP domains such as the P-Charging* headers in
      [RFC3455], the P-DCS-Billing-Info header in [RFC3603], etc.

   o  Performance Metrics:
      Layer-5 performance metrics should be defined and shared between
      peers.  The performance metrics apply directly to signaling or
      media; they may be used pro-actively to help avoid congestion,
      call quality issues or call signaling failures, and as part of
      monitoring techniques, they can be used to evaluate the
      performance of peering exchanges.
      Examples of SIP performance metrics include the maximum number of
      SIP transactions per second on per domain basis, Session
      Completion Rate (SCR), Session Establishment Rate (SER), etc.
      Some SIP end-to-end performance metrics are defined in
      [I-D.Malas-sip-performance]; a subset of these may be applicable
      to session peering and interconnects.
      Some media-related metrics for monitoring VoIP calls have been
      defined in the VoIP Metrics Report Block, in Section 4.7 of

   o  Security:
      A VSP/ASP SHOULD describe the security requirements that other
      peers must meet in order to terminate calls to its network.  While
      such a list of security-related policy parameters often depends on
      the security models pre-agreed to by peers, it is expected that
      these parameters will be discoverable or signaled in the future to
      allow session peering outside VSP clubs.  The list of security
      parameters may be long and composed of high-level requirements
      (e.g. authentication, privacy, secure transport) and low level
      protocol configuration elements like TLS parameters.
      The following list is not intended to be complete, it provides a
      preliminary list in the form of examples:

      *  Call admission requirements: for some providers, sessions can
         only be admitted if certain criteria are met.  For example, for
         some providers' networks, only incoming SIP sessions signaled
         over established IPSec tunnels or presented to the well-known

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         TLS ports are admitted.  Other call admission requirements may
         be related to some performance metrics as descrived above.
         Finally, it is possible that some requiremetns be imposed on
         lower layers, but these are considered out of scope of session

      *  Call authorization requirements and validation: the presence of
         a caller or user identity MAY be required by a VSP/ASP.
         Indeed, some VSPs/ASPs may further authorize an incoming
         session request by validating the caller's identity against
         white/black lists maintained by the service provider or users
         (traditional caller ID screening applications or IM white

      *  Privacy requirements: a VSP/ASP MAY demand that its SIP
         messages be securely transported by its peers for privacy
         reasons so that the calling/called party information be
         protected.  Media sessions may also require privacy and some
         ASP/VSP policies may include requirements on the use of secure
         media transport protocols such as sRTP, along with some
         contraints on the minimum authentication/encryption options for
         use in sRTP.

      *  Network-layer security parameters: this covers how IPSec
         security associated may be established, the IPSec key exchange
         mechanisms to be used and any keying materials, the lifetime of
         timed Security Associated if applicable, etc.

      *  Transport-layer security parameters: this covers how TLS
         connections should be established as described in Section 4.4.2

5.2.  Summary of Parameters for Consideration in Session Peering

   The following is a summary of the parameters mentioned in the
   previous section.  They may be part of a session peering policy and
   appear with a level of requirement (mandatory, recommended,
   supported, ...).

   o  IP Network Connectivity (assumed, requirements out of scope of
      this document)

   o  Media session parameters:

      *  Codecs for audio, video, real time text, instant messaging
         media sessions

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      *  Modes of communications for audio (voice, fax, DTMF), IM (page
         mode, MSRP)

      *  Media transport and means to establish secure media sessions

   o  SIP

      *  SIP RFCs, methods and error responses

      *  headers and header values

      *  possibly, list of SIP RFCs supported by groups (e.g. by call

   o  Accounting

   o  Performance Metrics: SIP signaling performance metrics; media-
      level VoIP metrics.

   o  Security: Call admission control, call authorization, network and
      transport layer security parameters, media security parameters

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6.  Acknowledgments

   This document is a work-in-progress and it is based on the input and
   contributions made by a large number of people in the SPEERMINT
   working group, including: Scott Brim, Mike Hammer, Richard Shocky,
   Henry Sinnreich, Richard Stastny, Patrik Faltstrom, Otmar Lendl,
   Daryl Malas, Dave Meyer, Jason Livingood, Bob Natale, Brian Rosen,
   Eric Rosenfeld and Adam Uzelac.

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7.  Security Considerations

   Securing session peering communications involves numerous protocol
   exchanges, first and foremost, the securing of SIP signaling and
   media sessions.  The security considerations contained in RF 3261,
   RFC 4474 are applicable to the SIP protocol exchanges.  A number of
   security considerations are also described in Section 4.4 for VoIP

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8.  References

8.1.  Normative References

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

8.2.  Informative References

              Malas, D., "SIP End-to-End Performance Metrics",
              September 2006.

              Gurbani, V., Jeffrey, A., and S. Lawrence, "Domain
              Certificates in the Session Initiation Protocol (SIP)",
              draft-gurbani-sip-domain-certs-03 (work in progress),
              August 2006.

              Schulzrinne, H. and R. Marshall, "Requirements for
              Emergency Context  Resolution with Internet Technologies",
              August 2006.

              Conroy, L. and K. Fujiwara, "ENUM Implementation Issues
              and Experiences", June 2006.

              Rosenberg, J., "A Hitchhikers Guide to the Session
              Initiation Protocol (SIP)", October 2006.

              Hilt, V., "A Framework for Session Initiation Protocol
              (SIP) Session Policies",
              draft-ietf-sipping-session-policy-framework-01 (work in
              progress), June 2006.

              Meyer, R., "SPEERMINT Terminology", September 2006.

              Wing, D., Fries, S., and H. Tschofenig, "A Framework for
              Session Initiation Protocol (SIP) Session Policies",
              draft-wing-media-security-requirements-00 (work in
              progress), October 2006.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,

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              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

   [RFC2782]  Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
              specifying the location of services (DNS SRV)", RFC 2782,
              February 2000.

   [RFC2915]  Mealling, M. and R. Daniel, "The Naming Authority Pointer
              (NAPTR) DNS Resource Record", RFC 2915, September 2000.

   [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.

   [RFC3262]  Rosenberg, J. and H. Schulzrinne, "Reliability of
              Provisional Responses in Session Initiation Protocol
              (SIP)", RFC 3262, June 2002.

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

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, October 2002.

   [RFC3326]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, December 2002.

   [RFC3455]  Garcia-Martin, M., Henrikson, E., and D. Mills, "Private
              Header (P-Header) Extensions to the Session Initiation
              Protocol (SIP) for the 3rd-Generation Partnership Project
              (3GPP)", RFC 3455, January 2003.

   [RFC3546]  Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J.,
              and T. Wright, "Transport Layer Security (TLS)
              Extensions", RFC 3546, June 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

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   [RFC3603]  Marshall, W. and F. Andreasen, "Private Session Initiation
              Protocol (SIP) Proxy-to-Proxy Extensions for Supporting
              the PacketCable Distributed Call Signaling Architecture",
              RFC 3603, October 2003.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611,
              November 2003.

   [RFC3702]  Loughney, J. and G. Camarillo, "Authentication,
              Authorization, and Accounting Requirements for the Session
              Initiation Protocol (SIP)", RFC 3702, February 2004.

   [RFC3824]  Peterson, J., Liu, H., Yu, J., and B. Campbell, "Using
              E.164 numbers with the Session Initiation Protocol (SIP)",
              RFC 3824, June 2004.

   [RFC3966]  Schulzrinne, H., "The tel URI for Telephone Numbers",
              RFC 3966, December 2004.

   [RFC3986]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
              Resource Identifier (URI): Generic Syntax", STD 66,
              RFC 3986, January 2005.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

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Author's Address

   Jean-Francois Mule
   858 Coal Creek Circle
   Louisville, CO  80027

   Email: jf.mule@cablelabs.com

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