La Nueva Telefonía

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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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

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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.

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RFC 3372 SIP-T September 2002

Problem definition: To provide ISUP transparency across SS7-SIP

interworking

SS7-SIP Interworking Requirements SIP-T Functions

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Transparency of ISUP Encapsulation of ISUP in the

Signaling

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