Internet DRAFT - draft-ietf-speermint-terminology

draft-ietf-speermint-terminology



  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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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 

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Acknowledgment 

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   Internet Society. 

    



































 
 
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