RFC - 3666

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

Original:	ftp://ftp.isi.edu/in-notes/rfc3666.txt
Authors:	A. Johnston [MCI], S. Donovan [dynamicsoft], R. Sparks [dynamicsoft], C. Cunningham [dynamicsoft], K. Summers [Sonus]
Date:	December 2003
Category:	Best Current Practice [ BCP-76 ]

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Referred by:	3 RFC
Refers to:	10 RFC

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Status

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.

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