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SPEERMINT Working Group                                        J-F. Mule
Internet-Draft                                                 CableLabs
Intended status: Informational                             June 27, 2008
Expires: December 29, 2008


          SPEERMINT Requirements for SIP-based Session Peering
                draft-ietf-speermint-requirements-05.txt

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
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   This Internet-Draft will expire on December 29, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2008).














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Abstract

   This memo captures protocol requirements identified for enabling
   session peering of voice, presence, instant messaging and other types
   of multimedia traffic.  It is an informational document linking the
   session peering use cases to protocol solutions.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  General Requirements . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Scope  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.2.  Border Elements  . . . . . . . . . . . . . . . . . . . . .  5
     3.3.  Session Establishment Data . . . . . . . . . . . . . . . .  8
       3.3.1.  User Identities and SIP URIs . . . . . . . . . . . . .  8
       3.3.2.  URI Reachability . . . . . . . . . . . . . . . . . . .  9
   4.  Considerations and Requirements for Session Peering of
       Presence and Instant Messaging . . . . . . . . . . . . . . . . 10
   5.  Security Requirements  . . . . . . . . . . . . . . . . . . . . 12
     5.1.  Security Properties for the Acquisition of Session
           Establishment Data . . . . . . . . . . . . . . . . . . . . 12
     5.2.  Security Properties for the SIP  exchanges . . . . . . . . 13
     5.3.  End-to-End Media Security  . . . . . . . . . . . . . . . . 13
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 14
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 16
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 17
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 17
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 17
   Appendix A.  Policy Parameters for Session Peering . . . . . . . . 20
     A.1.  Categories of Parameters and Justifications  . . . . . . . 20
     A.2.  Summary of Parameters for Consideration in Session
           Peering Policies . . . . . . . . . . . . . . . . . . . . . 23
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 25
   Intellectual Property and Copyright Statements . . . . . . . . . . 26














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1.  Introduction

   Peering at the session level represents an agreement between parties
   to allow the exchange of multimedia traffic.  It is assumed that
   these sessions use the Session Initiation Protocol (SIP) protocol to
   enable peering between two or more actors.  These actors are called
   SIP Service Providers (SSPs) and they are typically represented by
   users, user groups such as enterprises, real-time collaboration
   service communities, or other service providers offering voice or
   multimedia services.

   Common terminology for SIP session peering is defined in
   [I-D.ietf-speermint-terminology] and a reference architecture is
   described in [I-D.ietf-speermint-architecture].  A number of use
   cases describe how session peering has been or could be deployed
   based on the reference architecture
   ([I-D.ietf-speermint-voip-consolidated-usecases] and
   [I-D.ietf-speermint-consolidated-presence-im-usecases]).

   Peering at the session layer can be achieved on a bilateral basis
   (direct peering established directly between two SSPs), or on an
   indirect basis via a session intermediary (indirect peering via a
   third-party SSP that has a trust relationship with the SSPs) - see
   the terminology document for more details.

   This document first describes general requirements.  The use cases
   are then analyzed in the spirit of extracting relevant protocol
   requirements that must be met to accomplish the use cases.  These
   requirements are intended to be independent of the type of media
   exchanged such as Voice over IP (VoIP), video telephony, and instant
   messaging.  Media-specific requirements are defined in separate
   sections.

   It is not the goal of this document to mandate any particular use of
   IETF protocols by SIP Service Providers in order to establish session
   peering.  Instead, the document highlights what requirements should
   be met and what protocols may be used to define the solution space.

   Finally, we conclude with a list of parameters for the definition of
   a session peering policy, provided in an informative appendix.  It
   should be considered as an example of the information SIP Service
   Providers may have to discuss or agree on to exchange SIP traffic.









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2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   This document also reuses the terminology defined in [I-D.ietf-
   speermint-terminology].  It is assumed that the reader is familiar
   with the Session Description Protocol (SDP) [RFC4566] and the Session
   Initiation Protocol (SIP) [RFC3261].









































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3.  General Requirements

   The following sub-sections contain general requirements applicable to
   multiple use cases for multimedia session peering.

3.1.  Scope

   The primary focus of this document is on the requirements applicable
   to the boundaries of Layer 5 SIP networks: SIP entities and Signaling
   path Border Elements (SBEs); any requirements touching SIP UA or end-
   devices are considered out of scope.

   SSPs desiring to establish session peering relationships have to
   reach an agreement on numerous aspects.
   This document highlights only certain aspects of a session peering
   agreement, mostly the requirements relevant to protocols, including
   the declaration, advertisement and management of ingress and egress
   for session signaling and media, information related to the Session
   Establishment Data (SED), and the security mechanisms a peer may use
   to accept and secure session exchanges.
   Numerous other aspects of session peering arrangement are critical to
   reach a successful agreement but they are considered out of scope of
   the SPEERMINT working group.  They include aspects such as SIP
   protocol support (e.g.  SIP extensions and field conventions), media
   (e.g., type of media traffic to be exchanged, compatible media codecs
   and media transport protocols, mechanisms to ensure differentiated
   quality of service for media), SIP layer-3 IP connectivity between
   the Signaling Path and Data Path Border Elements, traffic capacity
   control (e.g. maximum number of SIP sessions at each ingress point,
   maximum number of concurrent IM or VoIP sessions), and accounting.

   The informative Appendix A lists parameters that may considered when
   discussing the technical aspects of SIP session peering.  The purpose
   of this list which has evolved through the working group use case
   discussions is to capture the parameters that are considered outside
   the scope of the protocol requirements.

3.2.  Border Elements

   For border elements to be operationally manageable, maximum
   flexibility should be given for how border elements are declared or
   dynamically advertised.

   Indeed, in any session peering environment, there is a need for a SIP
   Service Provider to declare or dynamically advertise the SIP entities
   that will face the peer's network.  The data path border elements are
   typically signaled dynamically in the session description.




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   The use cases defined
   ([I-D.ietf-speermint-voip-consolidated-usecases]) catalog the various
   border elements between SIP Service Providers; they include Signaling
   Path Border Elements (SBEs) and SIP proxies (or any SIP entity at the
   boundary of the Layer 5 network).

   o  Requirement #1:
      Protocol mechanisms must exist for a SIP Service Provider (SSP) to
      communicate the ingress Signaling Path Border Elements of its
      service domain.

      Notes on solution space:
      The SBEs may be advertised to session peers using static
      mechanisms or they may be dynamically advertised.  There seems to
      be general agreement that [RFC3263] provides a solution for
      dynamically advertising ingress SBEs in most cases of Direct or
      Indirect peering.  However, this DNS-based solution may be limited
      in cases where the DNS response varies based on who sends the
      query (peer-dependent SBEs, see below).

   o  Requirement #2:
      Protocol mechanisms should exist for a SIP Service Provider (SSP)
      to communicate the egress SBEs of its service domain.

      Notes on motivations for this requirement:
      For the purposes of capacity planning, traffic engineering and
      call admission control, a SIP Service Provider may be asked where
      it will generate SIP calls from.  The SSP accepting calls from a
      peer may wish to know where SIP calls will originate from (this
      information is typically used by the terminating SSP).
      Note that this may not be applicable to all types of session
      peering (voice may be a particular case where this is needed -- at
      least based on current practices).
      While provisioning requirements are out-of-scope of this document,
      some SSPs may find use for a mechanism to dynamically advertise or
      discover the egress SBEs of a peer.

   If the SSP also provides media streams to its users as shown in the
   use cases for "Originating" and "Terminating" SSPs, a mechanism
   should exist to allow SSPs to advertise their egress and ingress data
   path border elements (DBEs), if applicable.  While some SSPs may have
   open policies and accept media traffic from anywhere outside their
   network to anywhere inside their network, some SSPs may want to
   optimize media delivery and identify media paths between peers prior
   to traffic being sent (layer 5 to layer 3 QoS mapping).

   o  Requirement #3:
      Protocol mechanisms should be available to allow a SIP Service



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      Provider to communicate its DBEs to its peers.

      Notes: Some SSPs engaged in SIP interconnects do exchange this
      type of DBE information today in a static manner.  Some SSPs do
      not.

   Some SSPs may have some restrictions on the type of media traffic
   their SIP entities acting as SBEs are capable of establishing.  In
   order to avoid a failed attempt to establish a session, a mechanism
   may be provided to allow SSPs to indicate if some restrictions exist
   on the type of media traffic: ingress and egress SBE points may be
   peer-dependent, and/or media-dependent.

   o  Requirement #4:
      The mechanisms recommended for the declaration or advertisement of
      SBE and DBE entities must allow for peer variability.

      Notes on solution space:
      For advertising peer-dependent SBEs (peer variability), the
      solution space based on [RFC3263] is under specified and there are
      no know best current practices.  Is DNS the right place for
      putting data that varies based on who asks?

   In the use cases provided as part of direct and indirect scenarios,
   an SSP deals with multiple SIP entities and multiple SBEs in its own
   domain.  There is often a many-to-many relationship between SIP
   Proxies and Signaling path Border Elements.
   It should be possible for an SSP to define which egress SBE a SIP
   entity must use based on a given peer destination.  For example, in
   the case of an indirect peering scenario (section 5.1.5 of
   [I-D.ietf-speermint-voip-consolidated-usecases], Figure 5), it should
   be possible for the O-Proxy to choose the appropriate O-SBE based on
   the information the O-Proxy receives from the Lookup Function (LUF)
   and/or Location Routing Function (LRF) - message response labeled
   (3).  Note that this example also applies to the case of Direct
   Peering when a service provider has multiple service areas and each
   service area involves multiple SIP Proxies and a few SBEs.

   o  Requirement #5:
      The mechanisms recommended for the lookup and location routing
      service must be capable or returning both a target URI destination
      and a value for the SIP Route header.

      Notes: solutions exist if the protocol used between the Proxy and
      the LUF/LRF is SIP; if ENUM is used, the author of this document
      does not know of any solution today.

   It is desirable for an SSP to be able to communicate how



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   authentication of a peer's SBEs will occur (see the security
   requirements for more details).

   o  Requirement #6:
      The mechanisms recommended for locating a peer's SBE must be able
      to convey how a peer should initiate secure session establishment.

      Notes : certain mechanisms exist; for example, the required
      protocol use of SIP over TLS may be discovered via [RFC3263].

3.3.  Session Establishment Data

   The Session Establishment Data (SED) is defined in
   [I-D.ietf-speermint-terminology] as the data used to route a call to
   the next hop associated with the called domain's ingress point.  The
   following paragraphs capture some general requirements on the SED
   data.

3.3.1.  User Identities and SIP URIs

   User identities used between peers can be represented in many
   different formats.  Session Establishment Data should rely on URIs
   (Uniform Resource Identifiers, [RFC3986]) and SIP URIs should be
   preferred over tel URIs ([RFC3966]) for session peering of VoIP
   traffic.
   The use of DNS domain names and hostnames is recommended in SIP URIs
   and they should be resolvable on the public Internet.  It is
   recommended that the host part of SIP URIs contain a fully-qualified
   domain name instead of a numeric IPv4 or IPv6 address.  As for the
   user part of the SIP URIs, the mechanisms for session peering should
   not require an SSP to be aware of which individual user identities
   are valid within its peer's domain.

   o  Requirement #7:
      The protocols used for session peering must accommodate the use of
      different types of URIs.  URIs with the same domain-part should
      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  Requirement #8:
      The mechanisms for session peering should not require an SSP to be
      aware of which individual user identities are valid within its
      peer's domain.

   o  Notes on the solution space for #7 and #8:
      This is generally well supported by IETF protocols.  When
      telephone numbers are in tel URIs, SIP requests cannot be routed



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      in accordance with the traditional DNS resolution procedures
      standardized for SIP as indicated in [RFC3824].  This means that
      the solutions built for session peering must not solely use PSTN
      identifiers such as Service Provider IDs (SPIDs) or Trunk Group
      IDs (they should not be precluded but solutions should not be
      limited to these).
      Motivations:
      Although SED data may be based on E.164-based SIP URIs for voice
      interconnects, a generic peering methodology should not rely on
      such E.164 numbers.

3.3.2.  URI Reachability

   Based on a well-known URI type (for e.g. sip, pres, or im URIs), it
   must be possible to determine whether the SSP domain servicing the
   URI allows for session peering, and if it does, it should be possible
   to locate and retrieve the domain's policy and SBE entities.
   For example, an originating service provider must be able to
   determine whether a SIP URI is open for direct interconnection
   without requiring an SBE 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.

   o  Requirement #9:
      The mechanisms for session peering must allow an SBE to locate its
      peer SBE given a URI type and the target SSP domain name.























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4.  Considerations and Requirements for Session Peering of Presence and
    Instant Messaging

   This section describes requirements for presence and instant
   messaging session peering.  Several use cases for presence and
   instant messaging peering are described in
   [I-D.ietf-speermint-consolidated-presence-im-usecases], a document
   authored by A. Houri, E. Aoki and S. Parameswar.  Credits for this
   section must go to A. Houri, E. Aoki and S. Parameswar.

   The following requirements for presence and instant messaging session
   peering are derived from
   [I-D.ietf-speermint-consolidated-presence-im-usecases] and
   [I-D.houri-speermint-presence-im-requirements]:

   o  Requirement #10:
      The mechanisms recommended for the exchange of presence
      information between SSPs MUST allow a user of one SSP's presence
      community to subscribe presentities served by another SSP via its
      local community, including subscriptions to a single presentity, a
      personal, public or ad-hoc group list of presentities.

      Notes: see section 2.2 of
      [I-D.ietf-speermint-consolidated-presence-im-usecases].

   o  Requirement #11:
      The mechanisms recommended for Instant Messaging message exchanges
      between SSPs MUST allow a user of one SSP's community to
      communicate with users of the other SSP community via their local
      community using various methods.  Such methods include sending a
      one-time IM message, initiating a SIP session for transporting
      sessions of messages, participating in n-way chats using chat
      rooms with users from the peer SSPs, sending a file or sharing a
      document.

      Notes: see section 2.6 of
      [I-D.ietf-speermint-consolidated-presence-im-usecases].

   o  Requirement #12: Privacy Sharing
      In order to enable sending less notifications between communities,
      there should be a mechanism that will enable sharing privacy
      information of users between the communities.  This will enable
      sending a single notification per presentity that will be sent to
      the appropriate watchers on the other community according to the
      presentity's privacy information.
      The privacy sharing mechanism must be done in a way that will
      enable getting the consent of the user whose privacy will be sent
      to the other community prior to sending the privacy information.



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      if user consent is not give, it should not be possible to this
      optimization.  In addition to getting the consent of users
      regarding privacy sharing, the privacy data must be sent only via
      secure channels between communities.

      Notes: see section 2.3 of
      [I-D.ietf-speermint-consolidated-presence-im-usecases].

   o  Requirement #13: Multiple Recipients
      It should be possible to send a presence document with a list of
      watchers on the other community that should receive the presence
      document notification.  This will enable sending less presence
      document notifications between the communities while avoiding the
      need to share privacy information of presentities from one
      community to the other.

   o  Requirement #14: Mappings
      Early deployments of SIP based presence and IM gateways are done
      in front of legacy proprietary systems that use different names
      for different properties that exist in PIDF.  For example "Do Not
      Disturb" may be translated to "Busy" in another system.  In order
      to make sure that the meaning of the status is preserved, there is
      a need that either each system will translate its internal
      statuses to standard PIDF based statuses of a translation table of
      proprietary statuses to standard based PIDF statuses will be
      provided from one system to the other.

























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5.  Security Requirements

   This section describes the security properties that are desirable for
   the protocol exchanges in scope of session peering.  Three types of
   information flows are described in the architecture and use case
   documents: the acquisition of the Session Establishment Data (SED)
   based on a destination target via the Lookup and Location Routing
   Functions (LUF and LRF), the SIP signaling between SIP Service
   Providers, and the associated media exchanges.

   This section is focused on three security services, authentication,
   data confidentiality and data integrity as summarized in [RFC3365].
   However, this text does not specify the mandatory-to-implement
   security mechanisms as required by [RFC3365]; this is left for future
   protocol solutions that meet the requirements.

   A security threat analysis provides guidance for session peering
   ([I-D.draft-niccolini-speermint-voipthreats]).

5.1.  Security Properties for the Acquisition of Session Establishment
      Data

   The Look-Up Function (LUF) and Location Routing Function (LRF) are
   defined in [I-D.ietf-speermint-terminology].  They provide mechanisms
   for determining the SIP target address and domain the request should
   be sent to, and the associated SED to route the request to that
   domain.

   A mutual authentication service is desirable for the LUF and LRF
   protocol exchanges.  The response from the LUF and LRF may depend on
   the identity of the requestor: the authentication of the LUF/LRF
   requests is therefore a desirable property.  Mutual authentication is
   also desirable: the requestor may verify the identity of the systems
   that provided the LUF/LRF responses given the nature of the data
   returned in those responses.  Authentication also provides some
   protection for the availability of the LUF and LRF against attackers
   that would attempt to launch DoS attacks by sending bogus requests
   causing the LUF to perform a lookup and consume resources.

   Given the sensitive nature of the session establishment data
   exchanged with the LUF and LRF functions, the protocol mechanisms
   chosen for the lookup and location routing should offer data
   confidentiality and integrity protection (SED data may contain user
   addresses, SIP URI, location of SIP entities at the boundaries of SIP
   Service Provider domains, etc.).

   Requirement #15:
   The data exchanges for the lookup and location routing MUST support



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   mutual authentication, data confidentiality and integrity.

   Notes on the solution space: ENUM, SIP and proprietary protocols are
   typically used today for accessing these functions.  SSPs may use
   lower layer security mechanisms to guarantee some of those security
   properties.

5.2.  Security Properties for the SIP  exchanges

   The fundamental mechanisms for securing SIP are applicable (see
   Section 26.2 of [RFC3261], and [RFC4474]).

   Authentication of SIP communications are desirable, especially in the
   context of session peering involving SIP intermediaries.  Data
   confidentiality and integrity of the SIP message body may be
   desirable given some of the levels of session peering indirection
   (indirect/assisted peering), but they could be harmful as they may
   prevent intermediary SSPs from "inserting" SBEs/DBEs along the
   signaling and data paths.

5.3.  End-to-End Media Security

   Media security is critical to guarantee end-to-end confidentiality of
   the communication between the end-users' devices, independently of
   how many direct or indirect peers are along the signaling path.

   It is recommended that the establishment of media security be
   provided along the media path and not over the signaling path given
   the indirect peering use cases.

   Notes on the solution space:
   Media carried over the Real-Time Protocol (RTP) can be secured using
   secure RTP or sRTP ([RFC3711]).  A framework for establishing sRTP
   security using Datagram TLS [RFC4347] is described in
   [I-D.ietf-sip-dtls-srtp-framework]: it allows for end-to-end media
   security establishment using extensions to DTLS
   ([I-D.ietf-avt-dtls-srtp]).  This DTLS-SRTP framework meets the above
   requirement.

   Note that media can also be carried in numerous protocols other than
   RTP such as SIP (SIP MESSAGE method), MSRP, XMPP, etc.  In these
   cases, it is desirable those those protocols offer data
   confidentiality protection at a minimum.








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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: Edwin Aoki, Scott Brim, John Elwell, Mike
   Hammer, Avshalom Houri, Richard Shocky, Henry Sinnreich, Richard
   Stastny, Patrik Faltstrom, Otmar Lendl, Daryl Malas, Dave Meyer,
   Sriram Parameswar, Jon Peterson, Jason Livingood, Bob Natale, Benny
   Rodrig, Brian Rosen, Eric Rosenfeld, Adam Uzelac and Dan Wing.
   Specials thanks go to Rohan Mahy, Brian Rosen, John Elwell for their
   initial drafts describing guidelines or best current practices in
   various environments, and to Avshalom Houri, Edwin Aoki and Sriram
   Parameswar for authoring the presence and instant messaging
   requirements.





































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7.  IANA Considerations

   This document does not register any values in IANA registries.
















































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8.  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 [RFC3261],
   and [RFC4474] are applicable to the SIP protocol exchanges.  A number
   of security considerations are also described in Section Section 5.












































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9.  References

9.1.  Normative References

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

9.2.  Informative References

   [I-D.draft-ietf-pmol-sip-perf-metrics]
              Malas, D., "SIP End-to-End Performance Metrics",
              draft-ietf-pmol-sip-perf-metrics-01.txt (work in
              progress), June 2008.

   [I-D.draft-niccolini-speermint-voipthreats]
              Niccolini, S., Chen, E., and J. Seedorf, "VoIP Security
              Threats relevant to SPEERMINT",
              draft-niccolini-speermint-voipthreats-03.txt (work in
              progress), February 2008.

   [I-D.houri-speermint-presence-im-requirements]
              Houri, A., Aoki, E., and S. Parameswar, "Presence and IM
              Requirements", May 2007.

   [I-D.ietf-avt-dtls-srtp]
              McGrew, D. and E. Rescorla, "DTLS Extensions to Establish
              Keys for SRTP", draft-ietf-avt-dtls-srtp-02.txt (work in
              progress), February 2008.

   [I-D.ietf-sip-dtls-srtp-framework]
              Fischl, J., Tschofenig, H., and E. Rescorla, "DTLS-SRTP
              Framework", draft-ietf-sip-dtls-srtp-framework-01 (work in
              progress), February 2008.

   [I-D.ietf-sip-hitchhikers-guide]
              Rosenberg, J., "A Hitchhikers Guide to the Session
              Initiation Protocol (SIP)", July 2007.

   [I-D.ietf-speermint-architecture]
              Penno et al., R., "SPEERMINT Peering Architecture",
              draft-ietf-speermint-architecture-06.txt (work in
              progress), May 2008.

   [I-D.ietf-speermint-consolidated-presence-im-usecases]
              Houri, A., Aoki, E., and S. Parameswar, "Presence &
              Instant Messaging Peering Use Cases",
              draft-ietf-speermint-consolidated-presence-im-usecases-04
              (work in progress), February 2008.



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   [I-D.ietf-speermint-terminology]
              Meyer, R. and D. Malas, "SPEERMINT Terminology",
              draft-ietf-speermint-terminology-16.txt (work in
              progress), February 2008.

   [I-D.ietf-speermint-voip-consolidated-usecases]
              Uzelac et al., A., "VoIP SIP Peering Use Cases",
              draft-ietf-speermint-voip-consolidated-usecases-08.txt
              (work in progress), May 2008.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

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

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

   [RFC3365]  Schiller, J., "Strong Security Requirements for Internet
              Engineering Task Force Standard Protocols", BCP 61,
              RFC 3365, August 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.

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

   [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



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              Initiation Protocol (SIP)", RFC 3702, February 2004.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 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.

   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security", RFC 4347, April 2006.

   [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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Appendix A.  Policy Parameters for Session Peering

   This informative section lists various types of parameters that
   should be considered by implementers when deciding what configuration
   parameters to expose to system administrators or management stations,
   as well as SSPs or federations of SSPs when discussing the technical
   aspects of a session peering policy.

   In the context of session peering, a policy can be defined as the set
   of parameters and other information needed by an SSP to exchange
   traffic with another peer.  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 an SSP's session-
   dependent or session independent policies as defined in [I-D.ietf-
   sipping-session-policy].

   Various types of policy information may need to be discovered or
   exchanged in order to establish session peering.  At a minimum, a
   policy should specify information related to session establishment
   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.

   Some aspects of session peering policies must be agreed to and
   manually implemented; they are static and are typically documented as
   part of a business contract, technical document or agreement between
   parties.  For some parameters linked to protocol support and
   capabilities, standard ways of expressing those policy parameters may
   be defined among SSP and exchanged dynamically.  For e.g., templates
   could be created in various document formats so that it could be
   possible to dynamically discover some of the domain policy.  Such
   templates could be initiated by implementers (for each software/
   hardware release, a list of supported RFCs, RFC parameters is
   provided in a standard format) and then adapted by each SSP based on
   its service description, server or device configurations and variable
   based on peer relationships.

A.1.  Categories of Parameters and Justifications

   The following list should be considered as an initial list of
   "discussion topics" to be addressed by peers when initiating a VoIP
   peering relationship.






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   o  IP Network Connectivity:
      Session peers should define how the IP network connectivity
      between their respective SBEs and DBEs.  While this is out of
      scope of session peering, SSPs must agree on a common mechanism
      for IP transport of 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 private p headers if applicable), error response
         codes, supported or recommended format of some header field
         values , etc.

      *  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].

   o  Presence and Instant Messaging: TBD

   o  Accounting:
      Methods used for call or session accounting should be specified.
      An SSP may require a peer to track session usage.  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



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      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 SSP 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.draft-ietf-pmol-sip-perf-metrics]; 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
      [RFC3611].

   o  Security:
      An SSP 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 SSP 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
         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
         peering.





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      *  Call authorization requirements and validation: the presence of
         a caller or user identity may be required by an SSP.  Indeed,
         some SSPs 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 list).

      *  Privacy requirements: an SSP 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 SSP 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
         Section 5.

A.2.  Summary of Parameters for Consideration in Session Peering
      Policies

   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

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

      *  Media transport and means to establish secure media sessions

      *  List of ingress and egress DBEs where applicable, including
         STUN Relay servers if present




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   o  SIP

      *  SIP RFCs, methods and error responses

      *  headers and header values

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

   o  Accounting

   o  Capacity Control and Performance Management: any limits on, or,
      means to measure and limit the maximum number of active calls to a
      peer or federation, maximum number of sessions and messages per
      specified unit time, maximum number of active users or subscribers
      per specified unit time, the aggregate media bandwidth per peer or
      for the federation, specified SIP signaling performance metrics to
      measure and report; media-level VoIP metrics if applicable.

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






























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Author's Address

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

   Email: jf.mule@cablelabs.com










































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Full Copyright Statement

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