Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
RFC 3666

Document Type RFC - Best Current Practice (January 2004; Errata)
Also known as BCP 76
Last updated 2013-03-02
Stream IETF
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Stream WG state (None)
Consensus Unknown
Document shepherd No shepherd assigned
IESG IESG state RFC 3666 (Best Current Practice)
Telechat date
Responsible AD Allison Mankin
IESG note Discuss comments regarding the normative status to be a BCP, need for an ENUM example in the PSTN call flows - revision was made and doc to return to IESG agenda.
Send notices to <gonzalo.camarillo@ericsson.com>, <dean.willis@softarmor.com>, <rohan@cisco.com>
Network Working Group                                        A. Johnston
Request for Comments: 3666                                           MCI
BCP: 76                                                       S. Donovan
Category: Best Current Practice                                R. Sparks
                                                           C. Cunningham
                                                             dynamicsoft
                                                              K. Summers
                                                                   Sonus
                                                           December 2003

                   Session Initiation Protocol (SIP)
          Public Switched Telephone Network (PSTN) Call Flows

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 (2003).  All Rights Reserved.

Abstract

   This document contains best current practice examples of Session
   Initiation Protocol (SIP) call flows showing interworking with the
   Public Switched Telephone Network (PSTN).  Elements in these call
   flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways.
   Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP.
   PSTN telephony protocols are illustrated using ISDN (Integrated
   Services Digital Network), ISUP (ISDN User Part), and FGB (Feature
   Group B) circuit associated signaling.  PSTN calls are illustrated
   using global telephone numbers from the PSTN and private extensions
   served on by a PBX (Private Branch Exchange).  Call flow diagrams and
   message details are shown.

Johnston, et al.         Best Current Practice                  [Page 1]
RFC 3666                  SIP PSTN Call Flows              December 2003

Table of Contents

   1.  Overview.....................................................   2
       1.1.  General Assumptions....................................   3
       1.2.  Legend for Message Flows...............................   4
       1.3.  SIP Protocol Assumptions...............................   5
   2.  SIP to PSTN Dialing..........................................   6
       2.1.  Successful SIP to ISUP PSTN call.......................   7
       2.2.  Successful SIP to ISDN PBX call........................  15
       2.3.  Successful SIP to ISUP PSTN call with overflow.........  23
       2.4.  Session established using ENUM Query...................  32
       2.5.  Unsuccessful SIP to PSTN call: Treatment from PSTN.....  38
       2.6.  Unsuccessful SIP to PSTN: REL w/Cause from PSTN........  45
       2.7.  Unsuccessful SIP to PSTN: ANM Timeout..................  49
   3.  PSTN to SIP Dialing..........................................  54
       3.1.  Successful PSTN to SIP call............................  55
       3.2.  Successful PSTN to SIP call, Fast Answer...............  62
       3.3.  Successful PBX to SIP call.............................  68
       3.4.  Unsuccessful PSTN to SIP REL, SIP error mapped to REL..  74
       3.5.  Unsuccessful PSTN to SIP REL, SIP busy mapped to REL...  76
       3.6.  Unsuccessful PSTN->SIP, SIP error interworking to tones  80
       3.7.  Unsuccessful PSTN->SIP, ACM timeout....................  84
       3.8.  Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy...  88
       3.9.  Unsuccessful PSTN->SIP, Caller Abandonment.............  91
   4.  PSTN to PSTN Dialing via SIP Network.........................  96
       4.1.  Successful ISUP PSTN to ISUP PSTN call.................  97
       4.2.  Successful FGB PBX to ISDN PBX call with overflow...... 105
   5.  Security Considerations...................................... 113
   6.  References................................................... 115
       6.1.  Normative References................................... 115
       6.2.  Informative References................................. 115
   7.  Acknowledgments.............................................. 116
   8.  Intellectual Property Statement.............................. 116
   9.  Authors' Addresses........................................... 117
   10. Full Copyright Statement..................................... 118

1.  Overview

   The call flows shown in this document were developed in the design of
   a SIP IP communications network.  They represent an example of a
   minimum set of functionality.

   It is the hope of the authors that this document will be useful for
   SIP implementers, designers, and protocol researchers alike and will
   help further the goal of a standard implementation of RFC 3261 [2].
   These flows represent carefully checked and working group reviewed
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