📄 rfc3372.txt
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Network Working Group A. Vemuri
Request for Comments: 3372 Qwest Communications
BCP: 63 J. Peterson
Category: Best Current Practice NeuStar
September 2002
Session Initiation Protocol for Telephones (SIP-T):
Context and Architectures
Status of this Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2002). All Rights Reserved.
Abstract
The popularity of gateways that interwork between the PSTN (Public
Switched Telephone Network) and SIP networks has motivated the
publication of a set of common practices that can assure consistent
behavior across implementations. This document taxonomizes the uses
of PSTN-SIP gateways, provides uses cases, and identifies mechanisms
necessary for interworking. The mechanisms detail how SIP provides
for both 'encapsulation' (bridging the PSTN signaling across a SIP
network) and 'translation' (gatewaying).
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. SIP-T for ISUP-SIP Interconnections . . . . . . . . . . . . . 4
3. SIP-T Flows . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.1 SIP Bridging (PSTN - IP - PSTN) . . . . . . . . . . . . . . . 8
3.2 PSTN origination - IP termination . . . . . . . . . . . . . . 9
3.3 IP origination - PSTN termination . . . . . . . . . . . . . . 11
4. SIP-T Roles and Behavior . . . . . . . . . . . . . . . . . . . 12
4.1 Originator . . . . . . . . . . . . . . . . . . . . . . . . . . 12
4.2 Terminator . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.3 Intermediary . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.4 Behavioral Requirements Summary . . . . . . . . . . . . . . . 15
5. Components of the SIP-T Protocol . . . . . . . . . . . . . . . 16
5.1 Core SIP . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
5.2 Encapsulation . . . . . . . . . . . . . . . . . . . . . . . . 16
5.3 Translation . . . . . . . . . . . . . . . . . . . . . . . . . 16
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RFC 3372 SIP-T September 2002
5.4 Support for mid-call signaling . . . . . . . . . . . . . . . . 17
6. SIP Content Negotiation . . . . . . . . . . . . . . . . . . . 17
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
10 References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
A. Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
B. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 21
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 23
1. Introduction
The Session Initiation Protocol (SIP [1]) is an application-layer
control protocol that can establish, modify and terminate multimedia
sessions or calls. These multimedia sessions include multimedia
conferences, Internet telephony and similar applications. SIP is one
of the key protocols used to implement Voice over IP (VoIP).
Although performing telephony call signaling and transporting the
associated audio media over IP yields significant advantages over
traditional telephony, a VoIP network cannot exist in isolation from
traditional telephone networks. It is vital for a SIP telephony
network to interwork with the PSTN.
The popularity of gateways that interwork between the PSTN and SIP
networks has motivated the publication of a set of common practices
that can assure consistent behavior across implementations. The
scarcity of SIP expertise outside the IETF suggests that the IETF is
the best place to stage this work, especially since SIP is in a
relative state of flux compared to the core protocols of the PSTN.
Moreover, the IETF working groups that focus on SIP (SIP and SIPPING)
are best positioned to ascertain whether or not any new extensions to
SIP are justified for PSTN interworking. This framework addresses
the overall context in which PSTN-SIP interworking gateways might be
deployed, provides use cases and identifies the mechanisms necessary
for interworking.
An important characteristic of any SIP telephony network is feature
transparency with respect to the PSTN. Traditional telecom services
such as call waiting, freephone numbers, etc., implemented in PSTN
protocols such as Signaling System No. 7 (SS7 [6]) should be offered
by a SIP network in a manner that precludes any debilitating
difference in user experience while not limiting the flexibility of
SIP. On the one hand, it is necessary that SIP support the
primitives for the delivery of such services where the terminating
point is a regular SIP phone (see definition in Section 2 below)
rather than a device that is fluent in SS7. However, it is also
essential that SS7 information be available at gateways, the points
Vemuri & Peterson Best Current Practice [Page 2]
RFC 3372 SIP-T September 2002
of SS7-SIP interconnection, to ensure transparency of features not
otherwise supported in SIP. If possible, SS7 information should be
available in its entirety and without any loss to trusted parties in
the SIP network across the PSTN-IP interface; one compelling need to
do so also arises from the fact that certain networks utilize
proprietary SS7 parameters to transmit certain information through
their networks.
Another important characteristic of a SIP telephony network is
routability of SIP requests - a SIP request that sets up a telephone
call should contain sufficient information in its headers to enable
it to be appropriately routed to its destination by proxy servers in
the SIP network. Most commonly this entails that parameters of a
call like the dialed number should be carried over from SS7 signaling
to SIP requests. Routing in a SIP network may in turn be influenced
by mechanisms such as TRIP [8] or ENUM [7].
The SIP-T (SIP for Telephones) effort provides a framework for the
integration of legacy telephony signaling into SIP messages. SIP-T
provides the above two characteristics through techniques known as
'encapsulation' and 'translation' respectively. At a SIP-ISUP
gateway, SS7 ISUP messages are encapsulated within SIP in order that
information necessary for services is not discarded in the SIP
request. However, intermediaries like proxy servers that make
routing decisions for SIP requests cannot be expected to understand
ISUP, so simultaneously, some critical information is translated from
an ISUP message into the corresponding SIP headers in order to
determine how the SIP request will be routed.
While pure SIP has all the requisite instruments for the
establishment and termination of calls, it does not have any baseline
mechanism to carry any mid-call information (such as the ISUP INF/INR
query) along the SIP signaling path during the session. This mid-
call information does not result in any change in the state of SIP
calls or the parameters of the sessions that SIP initiates. A
provision to transmit such optional application-layer information is
also needed.
Vemuri & Peterson Best Current Practice [Page 3]
RFC 3372 SIP-T September 2002
Problem definition: To provide ISUP transparency across SS7-SIP
interworking
SS7-SIP Interworking Requirements SIP-T Functions
==================================================================
Transparency of ISUP Encapsulation of ISUP in the
Signaling SIP body
Routability of SIP messages with Translation of ISUP information
dependencies on ISUP into the SIP header
Transfer of mid-call ISUP signaling Use of the INFO Method for mid-
messages call signaling
Table 1: SIP-T features that fulfill PSTN-IP inter-connection
Requirements
While this document specifies the requirements above, it provide
mechanisms to satisfy them - however, this document does serve as an
framework for the documents that do provide these mechanisms, all of
which are referenced in Section 5.
Note that many modes of signaling are used in telephony (SS7 ISUP,
BTNUP, Q.931, MF etc.). This document focuses on SS7 ISUP and aims
to specify the behavior across ISUP-SIP interfaces only. The scope
of the SIP-T enterprise may, over time, come to encompass other
signaling systems as well.
2. SIP-T for ISUP-SIP Interconnections
SIP-T is not a new protocol - it is a set of mechanisms for
interfacing traditional telephone signaling with SIP. The purpose of
SIP-T is to provide protocol translation and feature transparency
across points of PSTN-SIP interconnection. It intended for use where
a VoIP network (a SIP network, for the purposes of this document)
interfaces with the PSTN.
Using SIP-T, there are three basic models for how calls interact with
gateways. Calls that originate in the PSTN can traverse a gateway to
terminate at a SIP endpoint, such as an IP phone. Conversely, an IP
phone can make a call that traverses a gateway to terminate in the
PSTN. Finally, an IP network using SIP may serve as a transit
network between gateways - a call may originate and terminate in the
PSTN, but cross a SIP-based network somewhere in the middle.
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RFC 3372 SIP-T September 2002
The SS7 interfaces of a particular gateway determine the ISUP
variants that that gateway supports. Whether or nor a gateway
supports a particular version of ISUP determines whether it can
provide feature transparency while terminating a call.
The following are the primary agents in a SIP-T-enabled network.
o PSTN (Public Switched Telephone Network): This refers to the
entire interconnected collection of local, long-distance and
international phone companies. In the examples below, the term
Local Exchange Carrier (LEC) is used to denote a portion (usually,
a regional division) of the PSTN.
o IP endpoints: Any SIP user agent that can act as an originator or
recipient of calls. Thus, the following devices are classified as
IP endpoints:
* Gateways: A telephony gateway provides a point of conversion
between signaling protocols (such as ISUP and SIP) as well as
circuit-switch and packet-switched audio media. The term Media
Gateway Controller (MGC) is also used in the examples and
diagrams in this document to denote large-scale clusters of
decomposed gateways and control logic that are frequently
deployed today. So for example, a SIP-ISUP gateway speaks ISUP
to the PSTN and SIP to the Internet and is responsible for
converting between the types of signaling, as well as
interchanging any associated bearer audio media.
* SIP phones: The term used to represent all end-user devices
that originate or terminate SIP VoIP calls.
* Interface points between networks where administrative policies
are enforced (potentially middleboxes, proxy servers, or
gateways).
o Proxy Servers: A proxy server is a SIP intermediary that routes
SIP requests to their destinations. For example, a proxy server
might direct a SIP request to another proxy, a gateway or a SIP
phone.
Vemuri & Peterson Best Current Practice [Page 5]
RFC 3372 SIP-T September 2002
********************
*** ***
* *
* ------- *
* |proxy| *
* ------- *
|----| |----|
/|MGC1| VoIP Network |MGC2|\
/ ---- ---- \
SS7 / * * \ SS7
/ * ------- * \
/ * |proxy| * \
-------- * ------- * ---------
| LEC1 | ** ** | LEC2 |
-------- ********************* ---------
Figure 1: Motivation for SIP-T in ISUP-SIP interconnection
In Figure 2 a VoIP cloud serves as a transit network for telephone
calls originating in a pair of LECs, where SIP is employed as the
VoIP protocol used to set up and tear down these VoIP calls. At the
edge of the depicted network, an MGC converts the ISUP signals to SIP
requests, and sends them to a proxy server which in turn routes
calls on other MGCs. Although this figure depicts only two MGCs,
VoIP deployments would commonly have many such points of
interconnection with the PSTN (usually to diversify among PSTN rate
centers). For a call originating from LEC1 and be terminating in
LEC2, the originator in SIP-T is the gateway that generates the SIP
request for a VoIP call, and the terminator is the gateway that is
the consumer of the SIP request; MGC1 would thus be the originator
and MGC2, the terminator. Note that one or more proxies may be used
to route the call from the originator to the terminator.
In this flow, in order to seamlessly integrate the IP network with
the PSTN, it is important to preserve the received SS7 information
within SIP requests at the originating gateway and reuse this SS7
information when signaling to the PSTN at the terminating gateway.
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