Session Initiation Protocol for Telephones (SIP-T): Context and Architectures
RFC 3372

Document Type RFC - Best Current Practice (September 2002; Errata)
Also known as BCP 63
Last updated 2015-10-14
Stream IETF
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IESG IESG state RFC 3372 (Best Current Practice)
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Telechat date
Responsible AD Allison Mankin
IESG note IESG passage as BCP subsequent to 27 Jun 2002 meeting.
Published: RFC 3372
Send notices to <rohan@cisco.com>, <dean.willis@softarmor.com>
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

Vemuri & Peterson        Best Current Practice                  [Page 1]
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
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