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RFC 6271 - Requirements for SIP-Based Session Peering

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Internet Engineering Task Force (IETF)                         J-F. Mule
Request for Comments: 6271                                     CableLabs
Category: Informational                                        June 2011
ISSN: 2070-1721

               Requirements for SIP-Based Session Peering


   This memo captures protocol requirements to enable session peering of
   voice, presence, instant messaging, and other types of multimedia
   traffic.  This informational document is intended to link the various
   use cases described for session peering to protocol solutions.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1. Introduction ....................................................2
   2. Terminology .....................................................3
   3. General Requirements ............................................3
      3.1. Scope ......................................................4
      3.2. Border Elements ............................................4
      3.3. Session Establishment Data .................................8
           3.3.1. User Identities and SIP URIs ........................8
           3.3.2. URI Reachability ....................................9
   4. Requirements for Session Peering of Presence and
      Instant Messaging ..............................................10
   5. Security Considerations ........................................12
      5.1. Security Properties for the Acquisition of Session
           Establishment Data ........................................12
      5.2. Security Properties for the SIP Signaling Exchanges .......13
      5.3. End-to-End Media Security .................................14
   6. Acknowledgments ................................................15
   7. References .....................................................15
      7.1. Normative References ......................................15
      7.2. Informative References ....................................15
   Appendix A. Policy Parameters for Session Peering .................19
     A.1. Categories of Parameters for VoIP Session Peering and
          Justifications .............................................19
     A.2. Summary of Parameters for Consideration in Session
          Peering Policies ...........................................22

1.  Introduction

   Peering at the session level represents an agreement between parties
   to exchange multimedia traffic.  In this document, we assume that the
   Session Initiation Protocol (SIP) is used to establish sessions
   between SIP Service Providers (SSPs).  SIP Service Providers are
   referred to as peers, and they are typically represented by users,
   user groups, enterprises, real-time collaboration service
   communities, or other service providers offering voice or multimedia
   services using SIP.

   A number of documents have been developed to provide background
   information about SIP session peering.  It is expected that the
   reader is familiar with the reference architecture described in
   [ARCHITECTURE], use cases for voice ([VOIP]), and instant messaging
   and presence ([RFC5344]).

   Peering at the session layer can be achieved on a bilateral basis
   (direct peering established directly between two SSPs), or on an
   indirect basis via a session intermediary (indirect peering via a
   third-party SSP that has a trust relationship with the SSPs) -- see
   the terminology document [RFC5486] for more details.

   This document first describes general requirements.  The use cases
   are then analyzed in the spirit of extracting relevant protocol
   requirements that must be met to accomplish the use cases.  These
   requirements are intended to be independent of the type of media
   exchanged such as Voice over IP (VoIP), video telephony, and instant
   messaging (IM).  Requirements specific to presence and instant
   messaging are defined in Section 4.

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

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [RFC2119].

   This document also reuses the terminology defined in [RFC5486].

   It is assumed that the reader is familiar with the Session
   Description Protocol (SDP) [RFC4566] and the Session Initiation
   Protocol (SIP) [RFC3261].  Finally, when used with capital letters,
   the term 'Authentication Service' is to be understood as defined by
   SIP Identity [RFC4474].

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, signaling
   path border elements (SBEs), and the associated protocol requirements
   for the look-up and location routing of the session establishment
   data.  The requirements applicable to SIP User Agents or related to
   the provisioning of the session data are considered out of scope.

   SIP Service Providers have to reach an agreement on numerous points
   when establishing session peering relationships.

   This document highlights only certain aspects of a session peering
   agreement.  It describes the requirements relevant to protocols in
   four areas: the declaration, advertisement and management of ingress
   and egress border elements for session signaling and media
   (Section 3.2), the information exchange related to the Session
   Establishment Data (SED, Section 3.3), specific requirements for
   presence and instant message (Section 4), and the security properties
   that may be desirable to secure session exchanges (Section 5).

   Numerous other considerations of session peering arrangements are
   critical to reach a successful agreement, but they are considered out
   of scope of this document.  They include information about SIP
   protocol support (e.g., SIP extensions and field conventions), media
   (e.g., type of media traffic to be exchanged, compatible media codecs
   and transport protocols, mechanisms to ensure differentiated quality
   of service for media), Layer 3 IP connectivity between the signaling
   and data path border elements, and accounting and traffic capacity
   control (e.g., the maximum number of SIP sessions at each ingress
   point, or the maximum number of concurrent IM or VoIP sessions).

   The informative Appendix A lists parameters that may be considered
   when discussing the technical parameters of SIP session peering.  The
   purpose of this list is to capture the parameters that are considered
   outside the scope of the protocol requirements.

3.2.  Border Elements

   For border elements to be operationally manageable, maximum
   flexibility should be given for how they are declared or dynamically
   advertised.  Indeed, in any session peering environment, there is a
   need for a SIP Service Provider to declare or dynamically advertise
   the SIP entities that will face the peer's network.  The data path
   border elements are typically signaled dynamically in the session

   The use cases defined in [VOIP] catalog the various border elements
   between SIP Service Providers; they include signaling path border
   elements (SBEs) and SIP proxies (or any SIP entity at the boundary of
   the Layer 5 network).

   o  Requirement #1:

      Protocol mechanisms MUST be provided to enable a SIP Service
      Provider 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 is
      general agreement that [RFC3263] provides a solution for
      dynamically advertising ingress SBEs in most cases of direct or
      indirect peering.  We discuss the DNS-based solution space further
      in Requirement #4 below, especially in cases where the DNS
      response varies based on who sends the query (peer-dependent

   o  Requirement #2:

      Protocol mechanisms MUST be provided to enable a SIP Service
      Provider 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 from
      where it will generate SIP calls.  The SSP accepting calls from a
      peer may wish to know from where SIP calls will originate (this
      information is typically used by the terminating SSP).

      While provisioning requirements are out of scope, some SSPs may
      find use for a mechanism to dynamically advertise or discover the
      egress SBEs of a peer.

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

   o  Requirement #3:

      Protocol mechanisms MUST be provided to allow a SIP Service
      Provider to communicate its DBEs to its peers.

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

   In some SIP networks, SSPs may expose the same border elements to all
   peers.  In other environments, it is common for SSPs to advertise
   specific SBEs and DBEs to certain peers.  This is done by SSPs to
   meet specific objectives for a given peer: routing optimization of
   the signaling and media exchanges, optimization of the latency or
   throughput based on the 'best' SBE and DBE combination, and other
   service provider policy parameters.  These are some of the reasons
   why advertisement of SBEs and DBEs may be peer dependent.

   o  Requirement #4:

      The mechanisms recommended for the declaration or advertisement of
      SBE and DBE entities MUST allow for peer variability.

      Notes on solution space:

      A simple solution is to advertise SBE entities using DNS and
      [RFC3263] by providing different DNS names to different peers.
      This approach has some practical limitations because the SIP URIs
      containing the DNS names used to resolve the SBEs may be
      propagated by users, for example, in the form of sip:user@domain.
      It is impractical to ask users to implement different target URIs
      based upon their SIP Service Provider's desire to receive incoming
      session signaling at different ingress SBEs based upon the
      originator.  The solution described in [RFC3263] and based on DNS
      to advertise SBEs is therefore under specified for this

      Other DNS mechanisms have been used extensively in other areas of
      the Internet, in particular in Content Distribution
      Internetworking to make the DNS responses vary based on the
      originator of the DNS query (see [RFC3466], [RFC3568], and
      [RFC3570]).  The applicability of such solutions for session
      peering needs further analysis.

      Finally, other techniques such as Anycast services ([RFC4786]) may
      be employed at lower layers than Layer 5 to provide a solution to
      this requirement.  For example, anycast nodes could be defined by
      SIP service providers to expose a common address for SBEs into
      DNS, allowing the resolution of the anycast node address to the

      appropriate peer-dependent service address based on the routing
      topology or other criteria gathered from the combined use of
      anycast and DNS techniques.

      Notes on variability of the SBE advertisements based on the media

      Some SSPs may have some restrictions on the type of media traffic
      their SBEs can accept.  For SIP sessions however, it is not
      possible to communicate those restrictions in advance of the
      session initiation: a SIP target may support voice-only media,
      voice and video, or voice and instant messaging communications.
      While the inability to find out whether a particular type of SIP
      session can be terminated by a certain SBE can cause session
      attempts to fail, there is consensus to not add a new requirement
      in this document.  These aspects are essentially covered by SSPs
      when discussing traffic exchange policies and are deemed out of
      scope of this document.

   In the use cases provided as part of direct and indirect peering
   scenarios, an SSP deals with multiple SIP entities and multiple SBEs
   in its own domain.  There is often a many-to-many relationship
   between the SIP proxies considered inside the trusted network
   boundary of the SSP and its signaling path border elements at the
   network boundaries.

   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 a static direct peering scenario (Figure
   2 in Section 5.2. of [VOIP]), it should be possible for the SIP proxy
   in the originating network (O-Proxy) to select the appropriate egress
   SBE (O-SBE) to reach the SIP target based on the information the
   proxy receives from the Look-Up Function (O-LUF), and/or Location
   Routing Function (O-LRF) -- message response labeled (2).  Note that
   this example also applies to the case of indirect 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 Look-Up Function (LUF) and the
      Location Routing Functions (LRF) MUST be capable of returning both
      a target URI destination and a value providing the next SIP

      Notes: solutions may exist depending on the choice of the protocol
      used between the Proxy and its LUF/LRF.  The idea is for the
      O-Proxy to be provided with the next SIP hop and the equivalent of
      one or more SIP Route header values.  If ENUM is used as a
      protocol for the LUF, the solution space is undefined.

   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: some mechanisms exist.  For example, the required use of
      SIP over TLS may be discovered via [RFC3263], and guidelines
      concerning the use of the SIPS URI scheme in SIP have been
      documented in [RFC5630].

3.3.  Session Establishment Data

   The Session Establishment Data (SED) is defined in [RFC5486] 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

   The use of DNS domain names and hostnames is recommended in SIP URIs
   and they should be resolvable on the public Internet.  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.  The host part of SIP
      URIs SHOULD contain a fully qualified domain name instead of a
      numeric IPv4 or IPv6 address.

   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 Requirements #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 Public
      Switched Telephone Network (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).


      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 (e.g., sip:, pres:, or im: URIs), it
   must be possible to determine whether the SSP domain servicing the
   URI allows for session peering, and if it does, it should be possible
   to locate and retrieve the domain's policy and SBE entities.

   For example, an originating service provider must be able to
   determine whether a SIP URI is open for direct interconnection
   without requiring an SBE to initiate a SIP request.  Furthermore,
   since each call setup implies the execution of any proposed
   algorithm, the establishment of a SIP session via peering should
   incur minimal overhead and delay, and employ caching wherever
   possible to avoid extra protocol round trips.

   o  Requirement #9:

      The mechanisms for session peering MUST allow an SBE to locate its
      peer SBE given a URI type and the target SSP domain name.

4.  Requirements for Session Peering of Presence and Instant Messaging

   This section describes requirements for presence and instant
   messaging session peering.

   Two SSPs create a peering relationship to enable their IM and
   presence users to collaborate with users on the other SSP network.
   We focus the requirements on inter-domain subscriptions to presence
   information, the exchange of messages and privacy settings, and the
   use of standard presence document formats across domains.

   Several use cases for presence and instant messaging peering are
   described in [RFC5344], a document authored by A. Houri, E. Aoki, and
   S. Parameswar.  Credits for the original content captured from these
   use cases into requirements in this section must go to them.

   o  Requirement #10:

      The mechanisms recommended for the exchange of presence
      information between SSPs SHOULD allow a user of one presence
      community to send a presence subscription request to 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 Sections 2.1 and 2.2 of [RFC5344].

   o  Requirement #11:

      The mechanisms recommended for instant messaging exchanges between
      SSPs SHOULD allow a user of one SSP's community to communicate
      with users of the other SSP community via their local community
      using the various methods.  Note that some SSPs may exercise some
      control over which methods are allowed based on service policies.
      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,

      Notes: see Sections 2.4, 2.5, and 2.6 of [RFC5344].

   o  Requirement #12:

      In some presence communities, users can define the list of
      watchers that receive presence notifications for a given
      presentity.  Such privacy settings for watcher notifications per
      presentity are typically not shared across SSPs causing multiple
      notifications to be sent for one presentity change between SSPs.

      The sharing of those privacy settings per presentity between SSPs
      would allow fewer notifications: a single notification would be
      sent per presentity and the terminating SSP would send
      notifications to the appropriate watchers according to the
      presentity's privacy information.

      The mechanisms recommended for presence information exchanges
      between SSPs SHOULD allow the sharing of some user privacy
      settings in order for users to convey the list of watchers that
      can receive notification of presence information changes on a per-
      presentity basis.

      The privacy sharing mechanism must be done with the express
      consent of the user whose privacy settings will be shared with the
      other community.  Because of the privacy-sensitive information
      exchanged between SSPs, the protocols used for the exchange of
      presence information must follow the security recommendations
      defined in Section 6 of [RFC3863].

      Notes: see Section 2.3 of [RFC5344].

   o  Requirement #13:

      It should be possible for an SSP to associate a presence document
      with a list of watchers in the peer SSP community so that the peer
      watchers can receive the presence document notifications.  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.

      The systems used to exchange presence documents between SSPs
      SHOULD allow a presence document to be delivered to one or more

      Note: The presence document and the list of authorized watchers in
      the peer SSP may be sent separately.  Also, the privacy-sharing
      mechanisms defined in Requirement #12 also apply to this

   o  Requirement #14:

      Early deployments of SIP-based presence and instant messaging
      gateways have been done in front of legacy proprietary systems
      that use different naming schemes or name values for the elements
      and properties defined in a Presence Information Data Format
      (PIDF) document ([RFC3863]).  For example, the value "Do Not
      Disturb" in one presence service may be mapped to "Busy" in

      another system for the status element.  Beyond this example of
      status values, it is important to ensure that the meaning of the
      presence information is preserved between SSPs.

      The systems used to exchange presence documents between SSPs
      SHOULD use standard PIDF documents and translate any non-standard
      value of a PIDF element to a standard one.

5.  Security Considerations

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

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

   A security threat analysis provides additional guidance for session
   peering ([VOIPTHREATS]).

5.1.  Security Properties for the Acquisition of Session Establishment

   The Look-Up Function (LUF) and Location Routing Function (LRF) are
   defined in [RFC5486].  They provide mechanisms for determining the
   SIP target address and domain the request should be sent to, and the
   associated SED to route the request to that domain.

   o  Requirement #15:

      The protocols used to query the Look-Up and Location Routing
      Functions SHOULD support mutual authentication.


      A mutual authentication service should be provided for the LUF and
      LRF protocol exchanges.  The content of the response returned by
      the LUF and LRF may depend on the identity of the requestor: the
      authentication of the LUF and LRF requests is therefore a
      desirable property.  Mutual authentication is also desirable: the
      requestor may verify the identity of the systems that provided the

      LUF and LRF responses given the nature of the data returned in
      those responses.  Authentication also provides some protection for
      the availability of the LUF and LRF against attackers that would
      attempt to launch Denial-of-Service (DoS) attacks by sending bogus
      requests causing the LUF to perform a lookup and consume

   o  Requirement #16:

      The protocols used to query the Look-Up and Location Routing
      Functions SHOULD provide support for data confidentiality and


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

   o  Notes on the solution space for Requirements #15 and #16:

      ENUM, SIP, and proprietary protocols are typically used today for
      accessing these functions.  Even though SSPs may use lower-layer
      security mechanisms to guarantee some of those security
      properties, candidate protocols for the LUF and LRF should meet
      the above requirements.

5.2.  Security Properties for the SIP Signaling Exchanges

   The SIP signaling exchanges are out of scope of this document.  This
   section describes some of the security properties that are desirable
   in the context of SIP interconnects between SSPs without formulating
   any normative requirements.

   In general, the security properties desirable for the SIP exchanges
   in an inter-domain context apply to session peering.  These include:

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

   o  providing an Authentication Service to authenticate the identity
      of connected users based on the SIP Service Provider domains (for
      both the SIP requests and the responses).

   The fundamental mechanisms for securing SIP between proxy servers
   intra- and inter-domain are applicable to session peering; refer to
   Section 26.2 of [RFC3261] for transport-layer security of SIP
   messages using TLS, [RFC5923] for establishing TLS connections
   between proxies, [RFC4474] for the protocol mechanisms to verify the
   identity of the senders of SIP requests in an inter-domain context,
   and [RFC4916] for verifying the identity of the sender of SIP

5.3.  End-to-End Media Security

   Media security is critical to guarantee end-to-end confidentiality of
   the communication between the end-users' devices, independently of
   how many direct or indirect peers are present along the signaling
   path.  A number of desirable security properties emerge from this

   The establishment of media security may be achieved along the media
   path and not over the signaling path given the indirect peering use

   For example, media carried over the Real-Time Protocol (RTP) can be
   secured using secure RTP (SRTP [RFC3711]).  A framework for
   establishing SRTP security using Datagram TLS (DTLS) [RFC4347] is
   described in [RFC5763]: it allows for end-to-end media security
   establishment using extensions to DTLS ([RFC5764]).

   It should also be noted that media can be carried in numerous
   protocols other than RTP such as SIP (SIP MESSAGE method), MSRP (the
   Message Session Relay Protocol, [RFC4975], XMPP (the Extensible
   Messaging and Presence Protocol, [RFC6120]), and many others.  Media
   may also be carried over TCP ([RFC4571]), and it can be encrypted
   over secure connection-oriented transport sessions over TLS

   A desirable security property for session peering is for SIP entities
   to be transparent to the end-to-end media security negotiations: SIP
   entities should not intervene in the Session Description Protocol
   (SDP) exchanges for end-to-end media security.

   o  Requirement #17:

      The protocols used to enable session peering MUST NOT interfere
      with the exchanges of media security attributes in SDP.  Media
      attribute lines that are not understood by SBEs MUST be ignored
      and passed along the signaling path untouched.

6.  Acknowledgments

   This document is based on the input and contributions made by a large
   number of people including: Bernard Aboba, Edwin Aoki, Scott Brim,
   John Elwell, Patrik Faltstrom, Mike Hammer, Avshalom Houri, Otmar
   Lendl, Jason Livingood, Daryl Malas, Dave Meyer, Bob Natale, Sriram
   Parameswar, Jon Peterson, Benny Rodrig, Brian Rosen, Eric Rosenfeld,
   Peter Saint-Andre, David Schwartz, Richard Shocky, Henry Sinnreich,
   Richard Stastny, and Adam Uzelac.

   Specials thanks go to Rohan Mahy, Brian Rosen, and John Elwell for
   their initial documents describing guidelines or best current
   practices in various environments, to Avshalom Houri, Edwin Aoki, and
   Sriram Parameswar for authoring the presence and instant messaging
   requirements, and to Dan Wing for providing detailed feedback on the
   Security Consideration sections.

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

   [ARCHITECTURE]  Malas, D. and J. Livingood, "Session PEERing for
                   Multimedia INTerconnect Architecture", Work
                   in Progress, February 2011.

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

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

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

   [RFC3365]       Schiller, J., "Strong Security Requirements for
                   Internet Engineering Task Force Standard Protocols",
                   BCP 61, RFC 3365, August 2002.

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

   [RFC3466]       Day, M., Cain, B., Tomlinson, G., and P. Rzewski, "A
                   Model for Content Internetworking (CDI)", RFC 3466,
                   February 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.

   [RFC3568]       Barbir, A., Cain, B., Nair, R., and O. Spatscheck,
                   "Known Content Network (CN) Request-Routing
                   Mechanisms", RFC 3568, July 2003.

   [RFC3570]       Rzewski, P., Day, M., and D. Gilletti, "Content
                   Internetworking (CDI) Scenarios", RFC 3570,
                   July 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.

   [RFC3863]       Sugano, H., Fujimoto, S., Klyne, G., Bateman, A.,
                   Carr, W., and J. Peterson, "Presence Information Data
                   Format (PIDF)", RFC 3863, August 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.

   [RFC4571]       Lazzaro, J., "Framing Real-time Transport Protocol
                   (RTP) and RTP Control Protocol (RTCP) Packets over
                   Connection-Oriented Transport", RFC 4571, July 2006.

   [RFC4572]       Lennox, J., "Connection-Oriented Media Transport over
                   the Transport Layer Security (TLS) Protocol in the
                   Session Description Protocol (SDP)", RFC 4572,
                   July 2006.

   [RFC4786]       Abley, J. and K. Lindqvist, "Operation of Anycast
                   Services", BCP 126, RFC 4786, December 2006.

   [RFC4916]       Elwell, J., "Connected Identity in the Session
                   Initiation Protocol (SIP)", RFC 4916, June 2007.

   [RFC4975]       Campbell, B., Mahy, R., and C. Jennings, "The Message
                   Session Relay Protocol (MSRP)", RFC 4975,
                   September 2007.

   [RFC5344]       Houri, A., Aoki, E., and S. Parameswar, "Presence and
                   Instant Messaging Peering Use Cases", RFC 5344,
                   October 2008.

   [RFC5411]       Rosenberg, J., "A Hitchhiker's Guide to the Session
                   Initiation Protocol (SIP)", RFC 5411, February 2009.

   [RFC5486]       Malas, D. and D. Meyer, "Session Peering for
                   Multimedia Interconnect (SPEERMINT) Terminology",
                   RFC 5486, March 2009.

   [RFC5503]       Andreasen, F., McKibben, B., and B. Marshall,
                   "Private Session Initiation Protocol (SIP) Proxy-to-
                   Proxy Extensions for Supporting the PacketCable
                   Distributed Call Signaling Architecture", RFC 5503,
                   March 2009.

   [RFC5630]       Audet, F., "The Use of the SIPS URI Scheme in the
                   Session Initiation Protocol (SIP)", RFC 5630,
                   October 2009.

   [RFC5763]       Fischl, J., Tschofenig, H., and E. Rescorla,
                   "Framework for Establishing a Secure Real-time
                   Transport Protocol (SRTP) Security Context Using
                   Datagram Transport Layer Security (DTLS)", RFC 5763,
                   May 2010.

   [RFC5764]       McGrew, D. and E. Rescorla, "Datagram Transport Layer
                   Security (DTLS) Extension to Establish Keys for the
                   Secure Real-time Transport Protocol (SRTP)",
                   RFC 5764, May 2010.

   [RFC5923]       Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse
                   in the Session Initiation Protocol (SIP)", RFC 5923,
                   June 2010.

   [RFC6076]       Malas, D. and A. Morton, "Basic Telephony SIP End-to-
                   End Performance Metrics", RFC 6076, January 2011.

   [RFC6120]       Saint-Andre, P., "Extensible Messaging and Presence
                   Protocol (XMPP): Core", RFC 6120, March 2011.

   [VOIP]          Uzelac, A. and Y. Lee, "VoIP SIP Peering Use Cases",
                   Work in Progress, April 2010.

   [VOIPTHREATS]   Seedorf, J., Niccolini, S., Chen, E., and H. Scholz,
                   "Session Peering for Multimedia Interconnect
                   (SPEERMINT) Security Threats and Suggested
                   Countermeasures", Work in Progress, March 2011.

Appendix A.  Policy Parameters for Session Peering

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

   In the context of session peering, a policy can be defined as the set
   of parameters and other information needed by an SSP to exchange
   traffic with another peer.  Some of the session policy parameters may
   be statically exchanged and set throughout the lifetime of the
   peering relationship.  Other parameters may be discovered and updated
   dynamically using some explicit protocol mechanisms.  These dynamic
   parameters may be session dependent, or they may apply over multiple
   sessions or peers.

   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, and
   Layer 3 related interconnect requirements, which are out of the scope
   of this document.

   Some aspects of session peering policies must be agreed to and
   manually implemented; they are static and are typically documented as
   part of a business contract, technical document, or agreement between
   parties.  For some parameters linked to protocol support and
   capabilities, standard ways of expressing those policy parameters may
   be defined among SSPs and exchanged dynamically.  For example,
   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
   or hardware release, the template could list supported RFCs, and the
   associated RFC parameters implemented in the given release in a
   standard format.  Each SSP would then complete the template and adapt
   its content based on its service description, the deployed server or
   device configurations and the variation of these configurations based
   on peer relationships.

A.1.  Categories of Parameters for VoIP Session Peering and

   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 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
      accomplished via private (e.g., IPVPN, IPsec, etc.) or public IP

   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), dual-tone multi-frequency (DTMF) transport voice band
         data communications (as applicable) along with the supported or
         recommended codec packetization rates, level of RTP payload
         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.

      *  Media variability at the signaling path border elements: list
         of media types supported by the various ingress points of a
         peer's network.

      *  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 [RFC5411].

   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 [RFC5503], 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 proactively 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 [RFC6076];
      a subset of these may be applicable to session peering and

      Some media-related metrics for monitoring VoIP calls have been
      defined in the VoIP Metrics Report Block, in Section 4.7 of

   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 described above.

         Finally, it is possible that some requirements be imposed on
         lower layers, but these are considered out of scope of session

      *  Call authorization requirements and validation: the presence of
         a caller or user identity may be required by 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 lists).

      *  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 constraints on the
         minimum authentication/encryption options for use in SRTP.

      *  Network-layer security parameters: this covers how IPsec
         security associations may be established, the IPsec key
         exchange mechanisms should be used, and any details on keying
         materials, the lifetime of timed security associations if
         applicable, etc.

      *  Transport-layer security parameters: this covers how TLS
         connections should be established, as described in Section 5.

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

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

   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

   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

Author's Address

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

   EMail: jf.mule@cablelabs.com


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