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SPEERMINT Working Group                                        J-F. Mule
Internet-Draft                                                 CableLabs
Intended status: Best Current                               July 9, 2007
Practice
Expires: January 10, 2008


       SPEERMINT Requirements for SIP-based VoIP Interconnection
                draft-ietf-speermint-requirements-02.txt

Status of this Memo

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   This Internet-Draft will expire on January 10, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2007).













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Abstract

   This memo defines Best Current Practices for session peering between
   SIP Service Providers for voice or other types of multimedia traffic
   exchanges.  In its current state, 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,
   presence and instant messaging interconnects.  It is intended to
   become best current practices based on the use cases discussed in the
   SPEERMINT working group.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  General Requirements . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Scope  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.2.  Session Peering Points . . . . . . . . . . . . . . . . . .  5
     3.3.  Session Establishment Data (SED) . . . . . . . . . . . . .  6
       3.3.1.  User Identities and SIP URIs . . . . . . . . . . . . .  7
       3.3.2.  URI Reachability . . . . . . . . . . . . . . . . . . .  7
     3.4.  Other Considerations . . . . . . . . . . . . . . . . . . .  8
   4.  Signaling and Media Guidelines for Session Peering . . . . . . 10
     4.1.  Protocol Specifications  . . . . . . . . . . . . . . . . . 10
     4.2.  Minimum set of SIP-SDP-related requirements  . . . . . . . 10
     4.3.  Media-related Requirements . . . . . . . . . . . . . . . . 11
     4.4.  Requirements for Presence and Instant Messaging  . . . . . 11
     4.5.  Security Requirements  . . . . . . . . . . . . . . . . . . 13
       4.5.1.  Security in today's VoIP networks  . . . . . . . . . . 13
       4.5.2.  Signaling Security and TLS Considerations  . . . . . . 13
       4.5.3.  Media Security . . . . . . . . . . . . . . . . . . . . 14
   5.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 16
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 18
   Appendix A.  Policy Parameters for Session Peering . . . . . . . . 21
     A.1.  Categories of Parameters and Justifications  . . . . . . . 21
     A.2.  Summary of Parameters for Consideration in Session
           Peering Policies . . . . . . . . . . . . . . . . . . . . . 24
   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 traffic according to a policy.  It is
   assumed that these sessions use the Session Initiation Protocol (SIP)
   protocol to enable peering between two or more actors.  The actors of
   SIP session peering are called SIP Service Providers (SSPs) and they
   are typically represented by users, user groups such as enterprises
   or real-time collaboration service communities, or other service
   providers offering voice or multimedia services.

   Common terminology for SIP session peering is defined
   ([I-D.ietf-speermint-terminology]) and a reference architecture is
   described in [I-D.ietf-speermint-architecture].  As the traffic
   exchanged using SIP as the session establishment protocol increases
   between parties, a number of use cases have been exposed by users of
   SIP services and various other actors for how session level peering
   has been or could be deployed based on the reference architecture
   ([I-D.ietf-speermint-voip-consolidated-usecases]) .

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

   This document describes guidelines and requirements that are intended
   to become Best Current Practices for session peering (direct,
   indirect or assisted).  These requirements are also independent of
   the type of media exchanged by the parties and should be 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 specific requirements for VoIP, presence and
   instant messaging interconnects.
   It is not the goal of this document to mandate any particular use of
   any IETF protocols to establish session peering.  However, when
   protocol mechanisms are used, the document aims at providing
   guidelines or best current practices on how they should be
   implemented, configured or configurable in order to facilitate
   session peering.

   Finally, a list of parameters for the definition of a session peering
   policy is provided in an informative appendix.  It should be
   considered as an example of the information a SIP 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",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in RFC 2119
   [RFC2119].

   This memo makes use of the following terms and acronyms defined in
   [I-D.ietf-speermint-terminology]: SIP Service Provider (SSP),
   Signaling Path Border Element (SBE), Data Path Border Element (DBE),
   Session Establishment Data (SED), Layer 3 and Layer 5 peering,
   session peering, federation, etc.  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 sections define general guidelines and requirements
   applicable to session peering for multimedia sessions.

3.1.  Scope

   SSPs desiring to establish session peering relationships have to
   reach an agreement on numerous aspects.
   This document only addresses best current practice for certain
   aspects of a session peering agreement, including the declaration,
   advertisement and management of ingress and egress for session
   signaling and media, information and conventions related to the
   Session Establishment Data (SED), the security requirements each peer
   may enforce on its network to accept and secure session exchanges,
   and, the format and necessary details to determine the minimum set of
   parameters required to achieve SIP and SDP interoperability.
   Several other aspects of session peering are critical to reach a
   successful agreement but they are considered out of scope of the
   SPEERMINT working group and not addressed in this document.  They
   include aspects such as 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),
   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 primary focus of this document
   is on the requirements applicable to the boundaries of Layer 5 SIP
   networks: SIP UA or end-device requirements are also considered out
   of scope.

   The informative Appendix A lists parameters that SPPs should consider
   when discussing the technical aspects of SIP session peering.

3.2.  Session Peering Points

   For session peering to be scalable and operationally manageable by
   SSPs, maximum flexibility should be given for how signaling path and
   media path border elements are declared, dynamically advertised and
   updated.  Indeed, in any session peering environment, there is a need
   for a SIP Service Provider to declare or dynamically advertise the
   SIP and media entities that will face the peer's network.

   An SSP SHOULD declare the signaling border elements responsible for
   egress and ingress points so called Signaling Path Border Elements
   (SBEs).  If the SSP also provides media streams to its users, an SSP
   SHOULD declare the media border elements responsible for egress and
   ingress points so called Signaling Path Data Elements (SDEs); if such



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   an SSP relies on STUN servers ([RFC3489]) and STUN Relay extensions
   to permit the traversal of NAT devices, the SSP SHOULD declare those
   STUN servers as part of its SDEs.  It is RECOMMENDED that SSPs use
   DNS and provide one or more domain names to be used with [RFC3263] to
   locate SBEs.
   An SPP SHOULD also indicate if some restrictions exist on the type of
   media traffic the SIP entities acting as SBEs are capable of
   establishing.  Ingress and egress SBE points MAY be peer-dependent,
   and/or media-dependent.  An SSP SHOULD be able to accomodate
   multiple, media-dependent ingress points from a peer's network.  The
   mechanisms recommended for the declaration and advertisement of SBE
   and SDE entities MUST allow for peer and media variability.

   Motivations:
   While there could be one single Signaling Path Border Element (SBE)
   in some SSP networks that communicates with all SIP peer networks, an
   SSP may choose to have one or more SBEs for receiving incoming SIP
   session requests (ingress signaling points), and one or more SBEs for
   outgoing SIP session requests (egress signaling points).  Ingress and
   egress signaling points may be distinct SIP entities and could be
   media-dependent.  Some providers deploy SIP entities specialized for
   voice, real-time collaboration, etc.  For example, within an SSP
   network, some SBEs may be dedicated for certain types of media
   traffic due to specific SIP extensions required for certain media
   types (e.g.  SIMPLE, the SIP MESSAGE Method for Instant Messaging
   [RFC3428] or the Message Sessions Relay Protocol (MSRP)).

   An SSP SHOULD communicate how authentication of the peer's SBEs will
   occur (see the security requirements for more details).  The use of
   access control lists based on fixed IP addresses or fixed IP sub-nets
   of the SBEs is NOT RECOMMENDED as it does not scale: it not only
   involves an error-prone manual process to configure access control
   lists but it also prevents peers from dynamically making IP network
   addressing changes to their SBE egress points without advertising
   those changes "manually".

3.3.  Session Establishment Data (SED)

   The Session Establishment Data (SED) is defined as the data used to
   route a call or SIP session to the called domain's ingress point
   ([I-D.ietf-speermint-terminology]).  Given that SED is the set of
   parameters that the outgoing SBEs need to complete the session
   establishment, some information must be shared between SSPs on
   special requirements or conventions required for a successful session
   establishment.  The following paragraphs capture the recommended best
   practices for the SED data.





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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, RFC 3986 [RFC3986]) and SIP URIs
   SHOULD be preferred over tel URIs (RFC 3966 [RFC3966]).
   The use of DNS domain names and hostnames is RECOMMENDED in SIP URIs
   and they MUST 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, an SSP SHOULD NOT need to be aware of
   which individual user identities are valid within a peer's domain.

   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.  As described in
   [I-D.draft-elwell-speermint-enterprise-usecases], in some use cases
   for enterprise to enterprise peering (even if a transit SSP is
   involved), it should be possible to use user identity URIs that do
   not map to E.164 numbers, e.g. for presence, instant messaging and
   even for voice.

   Motivations:
   When SSP support voice, 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].  The recommendations defined in
   [RFC3824] SHOULD be followed by implementers when using E.164 numbers
   with SIP.  Furthermore, it is commonly assumed 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.

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.




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

   Motivations:
   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 peers.

3.4.  Other Considerations

   The considerations listed below were gathered early on in the
   SPEERMINT working group as part of discussions to define the scope of
   the working group.  They are left here but have been re-written
   without requirements verbs for the most part.

   o  Session peering should be independent of lower layers.
      The mechanisms used to establish session peering should
      accommodate diverse supporting lower layers.  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:
      Mechanisms developed for session peering should be flexible and
      extensible to cover existing and future session peering models.
      It is also recommended that SSP 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 an SPP 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 an SSP's session-dependent
      or session independent policies as defined in
      [I-D.ietf-sipping-session-policy-framework].

   o  Administrative and Technical Policies:
      Various types of policy information may need to be discovered or
      exchanged in order to establish session peering.  At a minimum, a



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      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, see examples in Section
      Appendix A.

      Motivations:
      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").


































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4.  Signaling and Media Guidelines for Session Peering

   This section provides some guidelines for maximizing SIP-based
   interconnections between SSPs.  It should be considered as the
   minimal set of requirements to be implemented to perform SIP
   interconnects for presence, IM, or VoIP.

4.1.  Protocol Specifications

   A detailed list of SIP and SDP RFCs the session peers' SBEs must
   conform to must be provided by SSPs.  It is NOT RECOMMENDED to rely
   on Internet-Drafts for commercial SIP interconnects, but if
   applicable, a list of supported or required IETF Internet-Drafts
   SHOULD be provided.  Such specifications SHOULD include protocol
   implementation compliance statements, indicate the minimal extensions
   that MUST be supported, and the full details on what options and
   protocol features MUST be supported, MUST NOT be supported or MAY be
   supported.  This specification SHOULD include a high-level
   description of the services that are expected to be supported by the
   peering relationship and it MAY include sample message flows.

4.2.  Minimum set of SIP-SDP-related requirements

   The main objective of SIP interconnects being the establishment of
   successful SIP calls between peer SSPs, this section provides some
   guidelines for the minimum set of SIP specifications that SHOULD be
   supported by SBEs.

   The Core SIP Specifications as defined in [RFC3261] and
   [I-D.ietf-sip-hitchhikers-guide] MUST be supported by Signaling Path
   Border Elements (SBEs) 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 SIP
   interconnects.  The Hitchkiker's guide include specific sections for
   voice, instant message and presence.

   Furthermore, SBE implementers MUST follow the recommendations
   contained in RFC 3261 regarding the use of the Supported and Require
   headers.  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 SSP target domain basis in order
   to match a network peer's policy and to maximize interoperability.

   In the cases of indirect or assisted peering, it is also important
   that an adequate level of SIP message transparency is available.  In
   particular, the intermediary SBE MUST NOT modify or remove
   information in the SIP or SDP parameters beyond what is required for
   the purpose of call routing.  In particular, intermediary SBE SHOULD



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   NOT:

   o  Remove SIP header lines, SIP header fields and SIP message bodies
      that are intended for the destination SBE, or the called SIP UA
      irrespective of whether or not those header lines or parameters
      are understood by the intermediary SBE;

   o  Modify header fields and bodies in a way that may break any
      integrity protection.

4.3.  Media-related Requirements

   SSPs engaged in session peering SHOULD support of compatible codecs.
   An SSP domain policy SHOULD specify media-related parameters that
   their user's SIP entities support or that the SSP authorizes in its
   domain's policy.  Direct media exchange between the SSPs' user
   devices is preferred and media transcoding SHOULD be avoided by
   proposing commonly agreed codecs.  SSPs SHOULD discuss mechanisms
   employed for IPv4-IPv6 translation of media, as well as solutions
   used for NAT traversal such as ICE [I-D.ietf-ice] and STUN
   ([RFC3489]).

   Motivations: The media capabilities of an SSP's network are either a
   property of the SIP end-devices, SIP applications, 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 media codecs for SIP sessions
   between SSPs is outside the scope of SPEERMINT.  A list of media-
   related policy parameters are provided in the informative Appendix A.

   For media related security guidance, please refer to Section
   Section 4.5.

4.4.  Requirements for Presence and Instant Messaging

   This section lists some presence and Instant Messaging requirements
   defined in [I-D.presence-im-requirements] and authored by A. Houri,
   E. Aoki and S. Parameswar.  Credits must go to A. Houri, E. Aoki and
   S. Parameswar.

   It was requested to integrate [I-D.presence-im-requirements] into
   this draft since some of the requirements are generic and non
   specific to any application type.  In particular, requirements
   numbered PRES-IM-REQ-001, PRES-IM-REQ-002, PRES-IM-REQ-010, PRES-IM-
   REQ-011, PRES-IM-REQ-015 and PRES-IM_REQ-017 are covered by
   guidelines provided in other parts of this document.




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   The numbering of the requirements is as defined in the above
   mentioned ID.  It is expected that as more discussions occur and
   consensus is achieved in the working group, those requirements will
   be renumbered or re-written in the mindset of a BCP document.  The
   following list describes requirements for presence and instant
   messaging session peering:

   o  From (PRES-IM-REQ-003, PRES-IM-REQ-004 and PRES-IM-REQ-005): 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
      public or private (personal) list of presentities.

   o  From (PRES-IM-REQ-006, PRES-IM-REQ-007, PRES-IM-REQ-008 and PRES-
      IM-REQ-009): The mechanisms recommended for Instant Messaging
      message exchanges between SSPs MUST allow a user of SSP's
      community to communicate with users of the other SSP community via
      their local community using various methods, including 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, or sending a file.

   o  PRES-IM-REQ-012: 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.

   o  PRES-IM-REQ-013: Privacy Sharing Security - 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. 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.

   o  PRES-IM-REQ-014: 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  PRES-IM-REQ-016: Mappings - A lot of the early deployments of SIP
      based presence and IM gateways are deployed in front of legacy



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      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.

4.5.  Security Requirements

4.5.1.  Security in today's VoIP networks

   In today's SIP deployments, various approaches exist to secure
   exchanges between SIP Service Providers.  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 for SIP by some categories of SSPs: TLS in
   the enterprise networks for applications such as VoIP and secure
   Instant Messaging or in service provider networks for Instant
   Messaging and presence applications, IPsec and L2/L3 VPNs in some SSP
   networks where there is a desire to secure all signaling and media
   traffic at or below the IP layer.  Media level security is not widely
   used today between providers for media transported using the Real-
   Time Protocol (RTP) , even though it is in use in few deployments
   where the privacy of voice and other RTP media is critical.
   A security threat analysis provides guidance for VoIP session peering
   ([I-D.draft-niccolini-speermint-voipthreats]).  More discussions
   based on this threat analysis and use cases is required in the
   working group to define best current practices that this document, or
   a separate memo should recommend for both signaling and media
   security.

4.5.2.  Signaling Security and TLS Considerations

   The Transport Layer Security (TLS) is a standard way to secure
   signaling between SIP entities.  TLS can be used in direct peering to
   mutually authenticate SSPs and provide message confidentiality and
   integrity protection.  The remaining paragraphs explore how TLS could
   be deployed and used between 2 SSPs to secure SIP exchanges.  The
   intent is to capture what two SSPs should discuss and agree on in
   order to establish TLS connections for SIP session peering.

      1.  SSPs SHOULD agree on one or more Certificate Authorities (CAs)
      to trust for securing session peering exchanges.
      Motivations:
      An SSP should have control over which root CAs it trusts for SIP
      communications.  This may imply creating a certificate trust list



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      and including the peer's CA for each authorized domain.  In the
      case of a federation, 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].  Federations supporting
      [RFC4474] MUST specify the CA(s) permitted to issue certificates
      of the authentication service.

      3.  SIP and SBE 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
      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 and SBE 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 [I-D.ietf-sip-certs].

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

4.5.3.  Media Security

   Media security for session peering is as important as signaling
   security, especially for SSPs that want to continue to meet commonly
   assumed privacy and confidentiality requirements outside their
   networks.  Media can be secured using secure media transport
   protocols (e.g. secure RTP or sRTP).  The issues of key management



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   protocols for sRTP are being raised in IETF and this continues to be
   an area where requirements definition and protocol work is ongoing.
   More consensus is required outside SPEERMINT before best current
   practices can emerge.  See media security requirements for SIP
   sessions ([I-D.ietf-wing-media-security-requirements]) and its
   references for more details.  Some of these scenarios may be
   applicable to interdomain SSP session peering.












































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5.  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, 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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6.  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 4.5.












































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7.  References

7.1.  Normative References

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

7.2.  Informative References

   [I-D.Malas-sip-performance]
              Malas, D., "SIP End-to-End Performance Metrics", May 2007.

   [I-D.draft-elwell-speermint-enterprise-usecases]
              Elwell, J. and B. Rodrig, "Use cases for Enterprise
              Peering using the Session Initiation Protocol",
              draft-elwell-speermint-enterprise-usecases-00.txt (work in
              progress), February 2007.

   [I-D.draft-niccolini-speermint-voipthreats]
              Niccolini, S. and E. Chen, "VoIP Security Threats relevant
              to SPEERMINT", March 2007.

   [I-D.gurbani-sip-domain-certs]
              Gurbani, V., Jeffrey, A., and S. Lawrence, "Domain
              Certificates in the Session Initiation Protocol (SIP)",
              draft-gurbani-sip-domain-certs-05 (work in progress),
              June 2007.

   [I-D.ietf-ice]
              Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", June 2007.

   [I-D.ietf-sip-certs]
              Jennings, C., Peterson, J., and J. Fischl, "Certificate
              Management Service for The Session Initiation Protocol
              (SIP)", May 2007.

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

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




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   [I-D.ietf-speermint-architecture]
              Penno et al., R., "SPEERMINT Peering Architecture",
              April 2007.

   [I-D.ietf-speermint-terminology]
              Meyer, R. and D. Malas, "SPEERMINT Terminology",
              July 2007.

   [I-D.ietf-speermint-voip-consolidated-usecases]
              Uzelac et al., A., "VoIP SIP Peering Use Cases",
              June 2007.

   [I-D.ietf-wing-media-security-requirements]
              Wing, D., Fries, S., and H. Tschofenig, "Requirements for
              a Media Security Key Management Protocol",
              draft-wing-media-security-requirements (work in progress),
              June 2007.

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

   [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.

   [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.

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

   [RFC3428]  Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.,
              and D. Gurle, "Session Initiation Protocol (SIP) Extension
              for Instant Messaging", RFC 3428, December 2002.

   [RFC3455]  Garcia-Martin, M., Henrikson, E., and D. Mills, "Private



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              Header (P-Header) Extensions to the Session Initiation
              Protocol (SIP) for the 3rd-Generation Partnership Project
              (3GPP)", RFC 3455, January 2003.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 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.

   [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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Appendix A.  Policy Parameters for Session Peering

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

   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.

   o  IP Network Connectivity:
      Session peers must define how the IP network connectivity between
      their respective SBEs and SDEs.  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.




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      *  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
      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.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



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      defined in the VoIP Metrics Report Block, in Section 4.7 of
      [RFC3611].

   o  Security:
      A 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.

      *  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



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         Section 4.5.

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 SDEs where applicable, including
         STUN Relay servers if present

   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

   Copyright (C) The IETF Trust (2007).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

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