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SPEERMINT Working Group                                        J-F. Mule
Internet-Draft                                                 CableLabs
Intended status: Informational                         February 25, 2008
Expires: August 28, 2008


       SPEERMINT Requirements for SIP-based VoIP Interconnection
                draft-ietf-speermint-requirements-04.txt

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   This Internet-Draft will expire on August 28, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2008).














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Abstract

   A number of use cases have been described for session peering of
   voice, presence, instant messaging and other types of multimedia
   traffic.  This memo captures some of the requirements identified by
   these use case scenarios.  It is intended to become an informational
   document linking the use cases to potential protocol solutions.


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 . . . . . . . . . . . . . . . .  8
       3.3.1.  User Identities and SIP URIs . . . . . . . . . . . . .  8
       3.3.2.  URI Reachability . . . . . . . . . . . . . . . . . . .  9
     3.4.  Other Considerations . . . . . . . . . . . . . . . . . . . 10
   4.  Considerations and Requirements for Session Peering of
       Presence and Instant Messaging . . . . . . . . . . . . . . . . 12
   5.  Security Requirements  . . . . . . . . . . . . . . . . . . . . 14
     5.1.  Security in SIP networks in the context of session
           peering  . . . . . . . . . . . . . . . . . . . . . . . . . 14
     5.2.  Security Requirements for the Lookup and Location
           Routing Data . . . . . . . . . . . . . . . . . . . . . . . 14
     5.3.  Hop-by-hop Security for SIP Signaling and TLS
           Considerations . . . . . . . . . . . . . . . . . . . . . . 15
     5.4.  End-to-End Media Security  . . . . . . . . . . . . . . . . 16
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 17
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 18
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 20
   Appendix A.  Policy Parameters for Session Peering . . . . . . . . 23
     A.1.  Categories of Parameters and Justifications  . . . . . . . 23
     A.2.  Summary of Parameters for Consideration in Session
           Peering Policies . . . . . . . . . . . . . . . . . . . . . 26
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 27
   Intellectual Property and Copyright Statements . . . . . . . . . . 28









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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
   ([I-D.ietf-speermint-terminology]) and a reference architecture is
   described in [I-D.ietf-speermint-architecture].  A number of use
   cases have been exposed by users of SIP services and various other
   actors describing how layer-5 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 an 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 that have been
   derived from the working group discussions.  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.  In the
   case where some requirements are media-specific, we define them in a
   separate section.

   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 RFC 2119
   [RFC2119].

   This document also reuses the SIP 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 SPPs may consider
   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.  Session Peering Points

   For session peering to be scalable and operationally manageable,
   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 entities



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   that will face the peer's network.  The media or data path border
   elements are typically signaled dynamically in the session
   description.

   The use cases defined
   ([I-D.ietf-speermint-voip-consolidated-usecases]) catalog the various
   session peering points 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.  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).

   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 media border elements
   responsible for egress and ingress data path border elements (DBEs),
   if applicable.  While some SPPs 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 and media 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?
      For advertising media-dependent SBEs, solutions exist as long as
      URIs are protocol-dependent URIs.  A protocol-dependent URI like a
      SIP URI can be mapped to more than one types of media.  It should
      be noted that some URIs like the IM URI are abstract ([RFC3428])
      and need to be translated to protocol dependent URIs.  It is also
      not possible to know what media is supported by the SIP SBE before
      initiating a query by using mechanisms like [RFC3263].

   The following example provides some additional motivations for the
   above requirement on advertising media-dependent SBEs to peers.
   In large multi-service SIP networks, an SSP chooses to have several
   SBEs for receiving incoming SIP session requests (ingress SBEs), and
   several SBEs for outgoing SIP session requests (egress SBEs).  In
   order to facilitate the operations, feature management, and
   maintenance of its SBEs, the SSP opts for having distinct SBEs for
   voice, real-time collaboration, etc.  Some SBEs are therefore
   dedicated to exchanging 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)).  Note that this example is
   applicable to some enterprise networks where IP voice traffic hits
   different SIP gateways and voice servers (e.g.  IP-PBX) than Instant
   Messaging and real-time collaboration servers (e.g. real-time
   collaboration and IM server supporting SIMPLE and XMPP).

   In the use cases provided as part of direct and indirect scenarios,



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   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)
   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 SIP Route.

      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
   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



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   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
      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.



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   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.

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 have not been updated in this revision of
   the draft.

   o  It is assumed that session peering is 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].

   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
      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



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      (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
      incompatible policy parameters will cause the session to fail
      (this was originally referred to as "no blocked calls").












































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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

   Session peering does bring a new environment in which security
   requirements should be analyzed but the fundamental mechanisms for
   securing SIP and media exchanges remain applicable (see Section 26.2
   of [RFC3261].  The issues are less in the mechanisms that do exist
   and can be used to mitigate threats than they are in getting two SSPs
   to agree on which ones to use.

   This section first provides a broad picture of the various mechanisms
   used today in the context of SIP session peering.  We then describe
   security considerations for the three types of information flows
   described in the use cases: the data queried from the Lookup or
   Location Routing Functions, data exchanged in the SIP signaling
   between SSPs (directly and indirectly), and media.

5.1.  Security in SIP networks in the context of session peering

   In today's SIP deployments, various approaches exist to secure
   exchanges between SIP Service Providers.  Lookup, signaling and media
   security are the three primary topics for consideration in most
   deployments.
   A number of transport, network and session-level mechanisms are used
   for SIP by some categories of SSPs.  TLS is used in the enterprise
   networks for applications such as VoIP and secure Instant Messaging
   and session-level security is used end-to-end for some instant
   messaging systems or in service provider networks for Instant
   Messaging and presence applications.  At the network-level, IPsec and
   L2/L3 VPNs are widely used 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 between providers is not widely used today 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 session peering
   ([I-D.draft-niccolini-speermint-voipthreats]).  More discussions
   based on this threat analysis and use cases continue to be required
   in the working group to define what hop-by-hop or end-to-end security
   requirements are necessary in the context of session peering.

5.2.  Security Requirements for the Lookup and Location Routing Data

   The Look-Up Function (LUF) and Location Routing Function (LRF) are
   defined in [I-D.ietf-speermint-terminology].  They provide a
   mechanism for determining for a given request the target domain to
   which the request should be routed, and SED required to route the



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   request to that domain.

   Requirement #15:
   The protocols used for the LUF and LRF must allow the look-up and SED
   data to be exchanged securely (authentication and encryption services
   should be provided).

   Notes on the solution space: ENUM, SIP and proprietary protocols are
   typically used today for accessing these functions.

5.3.  Hop-by-hop Security for SIP Signaling and TLS Considerations

   Given the direct and indirect peering uses cases referenced in the
   previous sections of this document, hop-by-hop security between two
   SSPs using Transport Layer Security (TLS) is desirable.

   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.  One or more Certificate Authorities (CAs) should be agreed
      between SSPs for securing session peering exchanges.
      Alternatively, self-signed certificates may also be used.

      Motivations:
      An SSP should have control over which root CAs it trusts for SIP
      communications.  This may imply creating a certificate trust list
      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 an SSP
      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.
      There are various use cases of direct peering where there is no
      pre-established trust relationship that can rely on self-signed
      certificates.

      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] and CA(s) must specify the CA(s) permitted to issue
      certificates of the authentication service.



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      3.  SIP entities and SBEs 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 as
      defined in [RFC3261].

      4.  The following SIP and TLS protocol parameters should be agreed
      upon as part of session peering policies: the version of TLS
      supported by SIP entities and SBEs (TLSv1, TLSv1.1), the SIP TLS
      port (default 5061), the server-side session timeout (default 300
      seconds), the list of supported or recommended ciphersuites, the
      list of trusted root CAs if applicable or whether self-signed
      certs are acceptable.

      5.  SIP entities and SBEs involved in the session establishment
      over TLS must verify and validate the client certificates.  See
      section 9 and 9.3 of [I-D.ietf-sip-certs].

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

5.4.  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.

   o  Requirement #16:
      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, the above requirement is also met given the security features
   of these protocols.







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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

   None.
















































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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-malas-performance-metrics]
              Malas, D., "SIP End-to-End Performance Metrics",
              December 2007.

   [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-01.txt (work in
              progress), November 2007.

   [I-D.ietf-sip-certs]
              Jennings, C., Peterson, J., and J. Fischl, "Certificate
              Management Service for The Session Initiation Protocol
              (SIP)", draft-ietf-sip-certs-05.txt (work in progress),
              January 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-sipping-session-policy]
              Hilt, V. and G. Camarillo, "A Session Initiation Protocol
              (SIP) Event Package for Session-Specific Session
              Policies", draft-ietf-sipping-policy-package-04.txt (work
              in progress), August 2007.



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   [I-D.ietf-speermint-architecture]
              Penno et al., R., "SPEERMINT Peering Architecture",
              draft-ietf-speermint-architecture-04.txt (work in
              progress), August 2007.

   [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.

   [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-05.txt
              (work in progress), February 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.

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

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




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   [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.

   [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 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 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.



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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.draft-malas-performance-metrics]; 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:
      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.

      *  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 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

   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 (2008).

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   contained in BCP 78, and except as set forth therein, the authors
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