Session Peering for Multimedia Interconnect (SPEERMINT) Terminology
RFC 5486
| Document | Type | RFC - Informational (March 2009; No errata) | |
|---|---|---|---|
| Authors | David Meyer , Daryl Malas | ||
| Last updated | 2015-10-14 | ||
| Replaces | draft-ietf-speermint-reqs-and-terminology | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 5486 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Jon Peterson | ||
| Send notices to | (None) | ||
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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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.,
bob@peer1.federation.example.com)
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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: d.malas@cablelabs.com
David Meyer (editor)
EMail: dmm@1-4-5.net
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