Network Working Group                                      D. Malas, Ed.
Request for Comments: 5486                                     CableLabs
Category: Informational                                    D. Meyer, Ed.
                                                             March 2009


 Session Peering for Multimedia Interconnect (SPEERMINT) Terminology

Status of This Memo

  This memo provides information for the Internet community.  It does
  not specify an Internet standard of any kind.  Distribution of this
  memo is unlimited.

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  document authors.  All rights reserved.

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  than English.

Abstract

  This document defines the terminology that is to be used in
  describing Session PEERing for Multimedia INTerconnect (SPEERMINT).










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Table of Contents

  1. Introduction ....................................................2
  2. SPEERMINT Context ...............................................3
  3. General Definitions .............................................4
     3.1. Signaling Path Border Element ..............................4
     3.2. Data Path Border Element ...................................4
     3.3. Session Establishment Data .................................4
     3.4. Call Routing ...............................................5
     3.5. PSTN .......................................................5
     3.6. IP Path ....................................................5
     3.7. Peer Network ...............................................5
     3.8. Service Provider ...........................................5
     3.9. SIP Service Provider .......................................6
  4. Peering .........................................................6
     4.1. Layer 3 Peering ............................................6
     4.2. Layer 5 Peering ............................................6
          4.2.1. Direct Peering ......................................7
          4.2.2. Indirect Peering ....................................7
          4.2.3. On-Demand Peering ...................................7
          4.2.4. Static Peering ......................................7
     4.3. Functions ..................................................7
          4.3.1. Signaling Function ..................................7
          4.3.2. Media Function ......................................8
          4.3.3. Look-Up Function ....................................8
          4.3.4. Location Routing Function ...........................8
  5. Federations .....................................................8
  6. Security Considerations .........................................9
  7. Acknowledgments .................................................9
  8. Informative References .........................................10

1.  Introduction

  The term "Voice over IP Peering" (VoIP Peering) has historically been
  used to describe a wide variety of practices pertaining to the
  interconnection of service provider networks and to the delivery of
  Session Initiation Protocol (SIP [2]) call termination over those
  interconnections.

  The discussion of these interconnections has at times been confused
  by the fact that the term "peering" is used in various contexts to
  describe interconnection at different levels in a protocol stack.
  Session Peering for Multimedia Interconnect focuses on how to
  identify and route real-time sessions (such as VoIP calls) at the
  session layer, and it does not (necessarily) cover the exchange of
  packet-routing data or media sessions.  In particular, "layer 5
  network" is used here to refer to the interconnection between SIP




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  servers, as opposed to interconnection at the IP layer ("layer 3").
  The term "peering" will be used throughout the remainder of the
  document for the purpose of indicating a layer 5 interconnection.

  This document introduces standard terminology for use in
  characterizing real-time session peering.  Note however, that while
  this document is primarily targeted at the VoIP peering case, the
  terminology described here is applicable to those cases in which
  service providers peer using SIP signaling (defined as SIP Service
  Providers; see Section 3.9) for non-voice or quasi-real-time
  communications like instant messaging.

  The remainder of this document is organized as follows: Section 2
  provides the general context for the Session PEERing for Multimedia
  INTerconnect working group (SPEERMINT).  Section 3 provides the
  general definitions for real-time, SIP-based communication, with
  initial focus on the VoIP peering case, and Section 4 defines the
  terminology describing the various forms of peering.  Finally,
  Section 5 introduces the concept of federations.

2.  SPEERMINT Context

  SPEERMINT provides a peering framework that leverages the building
  blocks of existing IETF-defined protocols such as SIP [2] and ENUM
  [4].  While the SPEERMINT working group describes the use of these
  protocols in peering, it does not redefine how these protocols use
  input or output variables necessary for creating Session
  Establishment Data (SED) or the methods in which this data will be
  used during the peering process.  See Section 3.3 for additional
  detail on SED and its principal variables such as Uniform Resource
  Identifiers (URIs) [3] and telephone numbers of E.164 numbers [5].
  For example, while the SPEERMINT working group is not limited to the
  use of telephone numbers, an E.164 number may be used as a key in an
  E.164-to-URI mapping using ENUM.  This mapping involves looking up
  Naming Authority Pointer (NAPTR) records in the DNS, and results in a
  SIP URI.  The process for deriving this information has already been
  defined in [4], but is used as a building block for SPEERMINT SED, on
  which the subsequent call routing is based.  Note that the call-
  routing step does not depend on the presence of an E.164 number.
  Indeed, the URI resulting from an ENUM query may no longer even
  contain numbers of any type.  In particular, the SIP URI can be
  advertised in various other ways, such as on a web page.

  Finally, note that the term "call" is being used here in the most
  general sense, i.e., call routing and session routing are used
  interchangeably.





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3.  General Definitions

3.1.  Signaling Path Border Element

  A signaling path border element (SBE) is located on the
  administrative border of a domain through which inter-domain session
  layer messages will flow.  It typically provides signaling functions
  such as protocol inter-working (for example, H.323 to SIP), identity
  and topology hiding, and Session Admission Control for a domain.

3.2.  Data Path Border Element

  A data path border element (DBE) is located on the administrative
  border of a domain through which flows the media associated with an
  inter-domain session.  It typically provides media-related functions
  such as deep packet inspection and modification, media relay, and
  firewall-traversal support.  The DBE may be controlled by the SBE.

3.3.  Session Establishment Data

  Session Establishment Data, or SED, is the data used to route a call
  to the next hop associated with the called domain's ingress point.  A
  domain's ingress point might, for example, be the location derived
  from various types of DNS records (NAPTR, SRV, or A record) [1] that
  resulted from the resolution of the SIP URI.

  More specifically, the SED is the set of parameters that the outgoing
  SBEs need to complete the call, and may include:

     o  A destination SIP URI

     o  A SIP proxy or ingress SBE to send the INVITE to, including:

        -  Fully Qualified Domain Name (FQDN)

        -  Port

        -  Transport Protocol (UDP [8], TCP [9], and TLS [7])

     o Security parameters, including:

        -  TLS certificate to use

        -  TLS certificate to expect

        -  TLS certificate verification setting





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     o  Optional resource control parameters such as:

        -  Limits on the total number of call initiations to a peer

        -  Limits on SIP transactions per second

3.4.  Call Routing

  Call routing is the set of processes and rules used to route a call
  and any subsequent mid-dialog SIP requests to their proper (SIP)
  destination.  More generally, call routing can be thought of as the
  set of processes and rules that are used to route a real-time session
  to its termination point.

3.5.  PSTN

  The term "PSTN" refers to the Public Switched Telephone Network.  In
  particular, the PSTN refers to the collection of interconnected,
  circuit-switched, voice-oriented public telephone networks, both
  commercial and government-owned.  In general, PSTN terminals are
  addressed using E.164 numbers; however, various dial-plans (such as
  emergency services dial-plans) may not directly use E.164 numbers.

3.6.  IP Path

  For the purposes of this document, an IP path is defined to be a
  sequence of zero or more IP router hops.

3.7.  Peer Network

  This document defines a peer network as the set of SIP user agents
  (UAs) (customers) that are associated with a single administrative
  domain and can be reached via some IP path.  Note that such a peer
  network may also contain end-users who are located on the PSTN (and
  hence may also be interconnected with the PSTN) as long as they are
  also reachable via some IP path.

3.8.  Service Provider

  A Service Provider (SP) is defined to be an entity that provides
  layer 3 (IP) transport of SIP signaling and media packets.  Example
  services may include, but are not limited to, Ethernet Private Line
  (EPL), Frame Relay, and IP Virtual Private Network (VPN).  An example
  of this may be an Internet Service Provider (ISP).







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3.9.  SIP Service Provider

  A SIP Service Provider (SSP) is an entity that provides session
  services utilizing SIP signaling to its customers.  In the event that
  the SSP is also a function of the SP, it may also provide media
  streams to its customers.  Such an SSP may additionally be peered
  with other SSPs.  An SSP may also interconnect with the PSTN.  An SSP
  may also be referred to as an Internet Telephony Service Provider
  (ITSP).  While the terms ITSP and SSP are frequently used
  interchangeably, this document and other subsequent SIP peering-
  related documents should use the term SSP.  SSP more accurately
  depicts the use of SIP as the underlying layer 5 signaling protocol.

4.  Peering

  While the precise definition of the term "peering" is the subject of
  considerable debate, peering in general refers to the negotiation of
  reciprocal interconnection arrangements, settlement-free or
  otherwise, between operationally independent service providers.

  This document distinguishes two types of peering, layer 3 peering and
  layer 5 peering, which are described below.

4.1.  Layer 3 Peering

  Layer 3 peering refers to interconnection of two service providers'
  networks for the purposes of exchanging IP packets that are destined
  for one (or both) of the peer's networks.  Layer 3 peering is
  generally agnostic to the IP payload, and is frequently achieved
  using a routing protocol such as the Border Gateway Protocol (BGP)
  [6] to exchange the required routing information.

  An alternate, perhaps more operational, definition of layer 3 peering
  is that two peers exchange only customer routes, and hence any
  traffic between peers terminates on one of the peers' networks or the
  peer's customer's network.

4.2.  Layer 5 Peering

  Layer 5 (session) peering refers to interconnection of two SSPs for
  the purposes of routing real-time (or quasi-real-time) call signaling
  between their respective customers using SIP methods.  Such peering
  may be direct or indirect (see Section 4.2.1 and Section 4.2.2
  below).  Note that media streams associated with this signaling (if
  any) are not constrained to follow the same set of IP paths.






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4.2.1.  Direct Peering

  Direct peering describes those cases in which two SSPs peer without
  using an intervening layer 5 network.

4.2.2.  Indirect Peering

  Indirect, or transit, peering refers to the establishment of either a
  signaling and media path or a signaling path alone via one (or more)
  layer 5 transit network(s).  In this case, it is generally required
  that a trust relationship is established between the originating SSP
  and the transit SSP on one side, and between the transit SSP and the
  termination SSP on the other side.

4.2.3.  On-Demand Peering

  SSPs are said to peer on-demand when they are able to exchange SIP
  traffic without any pre-association prior to the origination of a
  real-time transaction (like a SIP message) between the domains.  Any
  information that needs to be exchanged between domains in support of
  peering can be learned through a dynamic protocol mechanism.  On-
  demand peering can occur as direct or indirect.

4.2.4.  Static Peering

  SSPs are said to peer statically when pre-association between
  providers is required for the initiation of any real-time
  transactions (like a SIP message).  Static peering can occur as
  direct or indirect.  An example of static peering is a federation.
  Each of the peers within the federation must first agree on a common
  set of rules and guidelines for peering, thus pre-associating with
  each other prior to initiating session requests.

4.3.  Functions

  The following are terms associated with the functions required for
  peering.

4.3.1.  Signaling Function

  The Signaling Function (SF) performs routing of SIP requests for
  establishing and maintaining calls, and to assist in the discovery or
  exchange of parameters to be used by the Media Function (MF).  The SF
  is a capability of SIP processing elements such as SIP proxies, SBEs,
  and user agents.






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4.3.2.  Media Function

  The Media Function (MF) performs media-related functions such as
  media transcoding and media security implementation between two SSPs.
  The MF is a capability of SIP-session-associated media end-points
  such as DBEs and user agents.

4.3.3.  Look-Up Function

  The Look-Up Function (LUF) determines for a given request the target
  domain to which the request should be routed.  An example of an LUF
  is an ENUM [4] look-up or a SIP INVITE request to a SIP proxy
  providing redirect responses for peers.

  In some cases, some entity (usually a 3rd party or federation)
  provides peering assistance to the originating SSP by providing this
  function.  The assisting entity may provide information relating to
  direct (Section 4.2.1) or indirect (Section 4.2.2) peering as
  necessary.

4.3.4.  Location Routing Function

  The Location Routing Function (LRF) determines for the target domain
  of a given request the location of the SF in that domain, and
  optionally develops other SED required to route the request to that
  domain.  An example of the LRF may be applied to either example in
  Section 4.3.3.  Once the ENUM response or SIP 302 redirect is
  received with the destination's SIP URI, the LRF must derive the
  destination peer's SF from the FQDN in the domain portion of the URI.

  In some cases, some entity (usually a 3rd party or federation)
  provides peering assistance to the originating SSP by providing this
  function.  The assisting entity may provide information relating to
  direct (Section 4.2.1) or indirect (Section 4.2.2) peering as
  necessary.

5.  Federations

  A federation is a group of SSPs that agree to exchange calls with
  each other via SIP and who agree on a set of administrative rules for
  such calls (settlement, abuse-handling, etc.) and specific rules for
  the technical details of the peering.

  The following provides examples of rules that the peers of a
  federation may agree to and enforce upon all participants:

     o  Common domain for all federation peers (e.g.,
        [email protected])



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     o  Codec rules (e.g., all peers must use the ITU-T G.711 codec
        [10])

     o  Authentication preference (e.g., all peers must use TLS when
        requesting to establish sessions with other peers)

     o  Transport protocol (e.g., all peers must use TCP)

     o  SIP address resolution mechanisms (e.g., all peers must use
        ENUM for resolving telephone numbers to SIP URIs)

  Finally, note that an SSP can be a member of:

     -  No federation (e.g., the SSP has only bilateral peering
        agreements)

     -  A single federation

     -  Multiple federations

  Also, an SSP can have any combination of bilateral and multilateral
  (i.e., federated) peers.

6.  Security Considerations

  This document introduces no new security considerations.  However, it
  is important to note that session peering, as described in this
  document, has a wide variety of security issues that should be
  considered in documents addressing both protocol and use-case
  analysis.

7.  Acknowledgments

  Many of the definitions were gleaned from detailed discussions on the
  SPEERMINT, ENUM, and SIPPING mailing lists.  Scott Brim, John Elwell,
  Mike Hammer, Eli Katz, Gaurav Kulshreshtha, Otmar Lendl, Jason
  Livingood, Alexander Mayrhofer, Jean-Francois Mule, Jonathan
  Rosenberg, David Schwartz, Richard Shockey, Henry Sinnreich, Richard
  Stastny, Hannes Tschofenig, Adam Uzelac, and Dan Wing all made
  valuable contributions to early versions of this document.  Patrik
  Faltstrom also made many insightful comments to early versions of
  this document.









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8.  Informative References

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

  [2]   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.

  [3]   Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
        Four: The Uniform Resource Identifiers (URI)", RFC 3404,
        October 2002.

  [4]   Faltstrom, P. and M. Mealling, "The E.164 to Uniform Resource
        Identifiers (URI) Dynamic Delegation Discovery System (DDDS)
        Application (ENUM)", RFC 3761, April 2004.

  [5]   International Telecommunications Union, "The International
        Public Telecommunication Numbering Plan", ITU-T Recommendation
        E.164, February 2005.

  [6]   Rekhter, Y., Ed., Li, T., Ed., and S. Hares, Ed., "A Border
        Gateway Protocol 4 (BGP-4)", RFC 4271, January 2006.

  [7]  Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
        Protocol Version 1.2", RFC 5246, August 2008.

  [8]  Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
        1980.

  [9]  Postel, J., "DoD standard Transmission Control Protocol", RFC
        761, January 1980.

  [10]  ITU Recommendation G.711 (11/88) - Pulse code modulation (PCM)
        of voice frequencies.

Authors' Addresses

  Daryl Malas (editor)
  CableLabs
  858 Coal Creek Circle
  Louisville, CO  80027
  USA
  EMail: [email protected]

  David Meyer (editor)
  EMail: [email protected]



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