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Versions: (draft-ietf-speermint-reqs-and-terminology) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 RFC 5486

  SPEERMINT Working Group                                D. Malas, Ed.
  Internet-Draft                                             CableLabs
  Intended status: Informational                         D. Meyer, Ed.
  Expires: May 2009                                    November 7, 2008


                           SPEERMINT Terminology
                  draft-ietf-speermint-terminology-17.txt


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   This Internet-Draft will expire on May 7, 2009.

Abstract

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

Table of Contents


   1. Introduction...................................................2
   2. SPEERMINT Context..............................................3
   3. General Definitions............................................3
      3.1. Signaling Path Border Element.............................3
      3.2. Data Path Border Element..................................4
      3.3. Session Establishment Data................................4
      3.4. Call Routing..............................................5
      3.5. PSTN......................................................5


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      3.6. IP Path...................................................5
      3.7. Peer Network..............................................5
      3.8. Service Provider..........................................5
      3.9. SIP Service Provider......................................5
   4. Peering........................................................6
      4.1. Layer 3 Peering...........................................6
      4.2. Layer 5 Peering...........................................6
         4.2.1. Direct Peering.......................................6
         4.2.2. Indirect Peering.....................................6
         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.......................................7
         4.3.3. Look-Up Function.....................................7
         4.3.4. Location Routing Function............................8
   5. Federations....................................................8
   6. Acknowledgments................................................9
   7. Security Considerations........................................9
   8. IANA Considerations............................................9
   9. Informative References.........................................9
   Author's Addresses...............................................11
   Intellectual Property Statement..................................11
   Disclaimer of Validity...........................................11
   Copyright Statement..............................................12
   Acknowledgment...................................................12

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
   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


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   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 which 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 telephone numbers
   of E.164 numbers [5] and Uniform Resource Identifiers (URI) [3].  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.

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



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   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, and 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:

     . A destination SIP URI

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

          o  Fully Qualified Domain Name (FQDN)

          o  Port

          o  Transport Protocol (UDP [9], TCP [10], and TLS [11])

     . Security Parameters, including

          o  TLS certificate to use

          o  TLS certificate to expect

          o  TLS certificate verification setting

     . Optional resource control parameters such as

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

          o  Limits on SIP transactions per second






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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, which 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; various dial-plans (such as emergency
   services dial-plans), however, some applications may not directly use
   E.164 numbers.

3.6. IP Path

   For 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 too, Ethernet Private Line
   (EPL), Frame Relay, and IP VPN.  An example of this may be an
   Internet Service Provider (ISP).

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


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   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 which 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 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 the peer's network 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.

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 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



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   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 a SIP proxy, SBE,
   and user agent.

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 a DBE and user agent.

4.3.3. Look-Up Function

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



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

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

     . Common domain for all federation peers (e.g.
        bob@peer1.federation.example.com)

     . Codec rule (e.g. all peers must use the ITU-T G.711 codec [15])

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

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

     . SIP address resolution mechanisms (e.g. all peers must use ENUM
        for resolving TNs to SIP URIs)

     Finally, note that an SSP can be a member of

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


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          o  A single federation

          o  Multiple federations

     and an SSP can have any combination of bi-lateral and multi-
     lateral (i.e., federated) peers.

6. Acknowledgments

   Many of the definitions were gleaned from detailed discussions on the
   SPEERMINT, ENUM, and SIPPING mailing lists. Scott Brim, 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, Dan Wing, John Elwell, and Adam Uzelac all made valuable
   contributions to early versions of this document. Patrik Faltstrom
   also made many insightful comments to early versions of this draft.

7. 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.

8. IANA Considerations

   This document has no IANA actions.

9. 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.




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   [5]   International Telecommunications Union, "The International
         Public Telecommunication Numbering Plan", ITU-T Recommendation
         E.164, 02 2005.

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

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

   [8]   Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
         Extended Reports (RTCP XR)", RFC 3611, November 2003.

   [9]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs", BCP 26, RFC 2434, October
         1998.

   [10]  Dierks, T. and E. Rescorla, "The TLS Protocol Version 1.1", RFC
         4346, April 2006.

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

   [12]  Postel, J., "DoD Standard Transmission Control Protocol", RFC
         761, January 1980.

   [13]  Plummer, David C., "An Ethernet Address Resolution Protocol",
         RFC 826, November 1982.

   [14]  Babiarz, J., Chan, K., and F. Baker, "Configuration Guidelines
         for DiffServ Service Classes", RFC 4594, August 2006.

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
















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Author's Addresses

   Daryl Malas
   CableLabs
   858 Coal Creek Circle
   Louisville, CO  80027
   USA
   Email: d.malas@cablelabs.com

   David Meyer
   Email: dmm@1-4-5.net


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Copyright Statement

   Copyright (C) The IETF Trust (2008).

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

Acknowledgment

   Funding for the RFC Editor function is currently provided by the
   Internet Society.







































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