BCP 76

RFC 3666

Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows, December 2003

Canonical URL:
https://www.rfc-editor.org/rfc/rfc3666.txt
File formats:
Plain TextPDF
Status:
BEST CURRENT PRACTICE
Authors:
A. Johnston
S. Donovan
R. Sparks
C. Cunningham
K. Summers
Stream:
IETF
Source:
sipping (rai)

Cite this RFC: TXT  |  XML

DOI:  10.17487/RFC3666

Discuss this RFC: Send questions or comments to sipping@ietf.org

Other actions: View Errata  |  Submit Errata  |  Find IPR Disclosures from the IETF


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.


For the definition of Status, see RFC 2026.

For the definition of Stream, see RFC 4844.


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