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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                                Level 3 Communications
  Intended status: Informational                         D. Meyer, Ed.
  Expires: February 2008                               August 10, 2007


                           SPEERMINT Terminology
                  draft-ietf-speermint-terminology-10.txt


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

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............................................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


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      3.6. IP Path...................................................6
      3.7. Peer Network..............................................6
      3.8. Service Provider..........................................6
      3.9. SIP Service Provider......................................6
      3.10. Internet Telephony Service Provider......................6
   4. Peering........................................................6
      4.1. Layer 3 Peering...........................................7
      4.2. Layer 5 Peering...........................................7
         4.2.1. Direct Peering.......................................7
         4.2.2. Indirect Peering.....................................7
         4.2.3. Assisted Peering.....................................7
         4.2.4. On-demand Peering....................................8
         4.2.5. Static Peering.......................................8
      4.3. Functions.................................................8
         4.3.1. Look-Up Function.....................................8
         4.3.2. Location Function....................................8
         4.3.3. Signaling Function...................................8
         4.3.4. Media Function.......................................8
   5. Federations....................................................8
   6. Acknowledgments...............................................10
   7. Security Considerations.......................................10
   8. IANA Considerations...........................................10
   9. Normative References..........................................10
   Author's Addresses...............................................12
   Intellectual Property Statement..................................12
   Disclaimer of Validity...........................................12
   Copyright Statement..............................................13
   Acknowledgment...................................................13

1. Introduction

   The term "Voice over IP Peering" (VoIP Peering) has   historically
   been used to describe a wide variety of aspects 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
   relate to 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
   application layer, and it does not (necessarily) involve 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").
   Finally, the terms "peering" and "interconnect" are used
   interchangeably throughout this document.



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   This document introduces standard terminology for use in
   characterizing real-time session interconnection. Note however, that
   while this document is primarily targeted at the VoIP interconnect
   case, the terminology described here is applicable to those cases in
   which service providers interconnect using SIP signaling for non-
   voice or quasi-real-time communications.

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

2.  SPEERMINT Context

   Figure 1 depicts the general session interconnect context. Note that
   vertical axis in this figure describes the layering of identifiers,
   while the horizontal lines indicate working group scope. While the
   SPEERMINT working group is not limited (or coupled in any way) to the
   use of E.164 numbers, in the case shown here an E.164 number [5] is
   used as a key in an E.164 to Uniform Resource Identifier (URI)
   mapping (ENUM [4]). That URI is in turn used to retrieve a Naming
   Authority Pointer (NAPTR) record [3], which is in turn resolved into
   a SIP URI.  Call routing is based on the resulting SIP URI. Note that
   the call routing step does not depend on the presence of an E.164
   number. Indeed, the resulting SIP URI may no longer even contain any
   numbers of any type. In particular, the SIP URI can be advertised in
   various other ways, such as on a web page.

           E.164 number <--- Peer Discovery
                |
                | <--- ENUM lookup of NAPTR in Domain Name System (DNS)
                |
                |
                | ENUM Working Group Scope
           =====+====================================================
                | SPEERMINT Working Group Scope
           Lookup
           Discovery <--- Session Establishment Data (SED)
           SIP URI
                |
                | <--- Federation Detection, Policy
                |      Lookup, and Service Location
                |
                |
           Hostname <--- Addressing and session establishment
                |
                | SPEERMINT Working Group Scope


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           =====+====================================================
                | Out of scope for the SPEERMINT Working Group
                |
                | <--- Lookup of A and AAAA in DNS
                |
           IP address
                |
                | <--- Routing protocols, ARP [14] etc.
                |
           MAC-address

                  Figure 1: Session Interconnect Context

   Note that in Figure 1, Session Establishment Data (SED), is the data
   used to route a call to the called domain's ingress point (see
   Section 3.3 for additional detail).

   As illustrated in Figure 1, the ENUM Working Group is primarily
   concerned with the acquisition of SED while the SPEERMINT Working
   Group is focused on the use of such SED in routing session signaling
   requests. Importantly, the SED can be derived from ENUM (i.e., an
   E.164 DNS entry) or via any other mechanism available to theuser.
   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) provides signaling functions
   such as protocol inter-working (for example, H.323 to SIP), identity
   and topology hiding, and Call Admission Control (CAC) for a domain.
   Such an SBE is frequently (but need not be) deployed on a domain's
   border.

3.2. Data Path Border Element

   A data path border element (DBE) provides media-related functions
   such as deep packet inspection and modification, media relay, and
   firewall support under SBE control. As was the case with the SBE, a
   DBE is frequently deployed on a domain's border.

3.3. Session Establishment Data

   Session Establishment Data, or SED, is the data used to route a call
   to the called domain's ingress point. A domain's ingress point can be
   thought of as the location pointed to by the SRV record [1] that
   resulted from the resolution of the SED (i.e., a SIP URI).


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   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 to send the INVITE to, including

          o  Fully Qualified Domain Name (FQDN)

          o  Port

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

     . 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 calls to a peer

          o  Limits on SIP transactions/second

          o  Limits on the total amount of bandwidth used on a peering
             link

          o  In addition, lower layer parameters (such as Differentiated
             Services Code Point (DSCP) [8] markings on SIP and/or media
             packets) might also be included.

3.4. Call Routing

   Call routing is the set of processes, rules, and SED used to route a
   call to its proper (SIP) destination.  More generally, call routing
   can be thought of as the set of processes, rules and SED 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, may not directly use E.164 numbers.


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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 agent
   servers (UASs) [2] and SIP user agent clients (UACs) [2] (customers)
   that are controlled by 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 (or SP) is defined to be an entity that provides
   layer 3 (IP) transport of SIP signaling and media packets.  An
   example of this may be an Internet Service Provider (ISP).

3.9. SIP Service Provider

   A SIP Service Provider (or SSP) is an entity that provides
   application level transport of 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 a service provider may
   additionally be interconnected with other service providers; that is,
   it may "peer" with other service providers. A SSP may also
   interconnect with the PSTN.

3.10. Internet Telephony Service Provider

   An Internet Telephony Service Provider, or ITSP, is a synonym for
   SSP. While the terms ITSP and SSP are frequently used
   interchangeably, the SPEERMINT working group uses the term SSP.

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.





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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 one of the peer's network.

4.2. Layer 5 Peering

   Layer 5 (Session) peering refers to interconnection of two service
   providers for the purposes of routing real-time (or quasi-real time)
   call signaling between their respective customers using SIP methods.
   Such interconnection 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 service providers
   interconnect 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)
   referral or transit network(s). In this case it is generally required
   that a trust relationship is established between the originating
   service provider and the transit network on one side, and the transit
   network and the termination network on the other side.

4.2.3. Assisted Peering

   In this case, some entity (usually a 3rd party or federation)
   provides peering assistance to either the originating or terminating
   SSP by providing one or more functions (see Section 4.3) assisting in
   the routing of SIP requests, the establishment of SIP dialogs, and
   sessions between one or more SSP's.  The assisting entity may provide
   information relating to direct (Section 4.2.1) or indirect (Section
   4.2.2) peering as necessary.





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4.2.4. On-demand Peering

   Service providers are said to peer on-demand when they are able to
   exchange 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, indirect, or assisted.

4.2.5. Static Peering

   Service providers 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, indirect, or assisted.

4.3. Functions

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

4.3.1. Look-Up Function

   The Look-Up Function (LUF) provides a mechanism for querying an
   internal and/or external database, which maintains a list of SIP user
   names and associated peering domains.

4.3.2. Location Function

   The Location Function (LF) develops call routing data (CRD) by
   discovering the Signaling Function (SF), and SF's reachable host (IP
   Address and port).

4.3.3. Signaling Function

   The SF performs routing of SIP messages, to optionally perform
   termination and re-initiation of the call, and to assist in the
   discovery/exchange of parameters to be used by the Media Function
   (MF).

4.3.4. Media Function

   The MF performs media related functions such as media transcoding and
   media security implementation between two SSPs.

5. Federations

   A federation is a group of SSPs which agree to receive calls from
   each other via SIP, and who agree on a set of administrative rules


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   for such calls (settlement, abuse-handling, ...) and the specific
   rules for the technical details of the interconnection.

   A federation may provide some or all of the following functionality:

     . Common static policies

          o  Routing

          o  Domain

          o  Location

          o  Next hop

          o  Network-to-Network Interface (NNI)

     . Common dynamic policies

          o  Congestion control

          o  Codec preference

          o  Authentication preference

          o  Quality monitoring capabilities (e.g. RTP Control Protocol
             (RTCP) [12], RTCP Extended Reports (RTCP XR) [13])

     . Policy management (enforcement)

          o  Ad-hoc

          o  Published in the DNS, or

          o  Policy might also be managed by a federation entity

     . A federated ENUM root

     . Address resolution mechanisms

     . Session signaling (via federation policy)

     . Media streams (via federation policy)

     . Federation security policies

     . Interconnection policies

     . Other layer 2 and layer 3 policies


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     . Security parameters

     . Optional resource control parameters

     Finally, note that a SSP can be a member of

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

          o  A single federation

          o  Multiple federations

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

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, 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, and contributed
   the basis of Figure 1.

7. Security Considerations

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

8. IANA Considerations

   This document creates no new requirements on IANA namespaces [7].

9. Normative 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.


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   [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, 02 2005.

   [6]   Rekhter, Y. and T. Li, "A Border Gateway Protocol 4 (BGP-4)",
         RFC 1771, March 1995.

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

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

   [9]   Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC
         2246, January 1999.

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

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

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

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

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











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Author's Addresses

   Daryl Malas
   Level 3 Communications LLC
   1025 Eldorado Blvd.
   Broomfield, CO 80021
   USA
   Email: daryl.malas@level3.com

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


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   This document and the information contained herein are provided on an
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   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.





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Copyright Statement

   Copyright (C) The IETF Trust (2007).

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

Acknowledgment

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







































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