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CurrentBEST CURRENT PRACTICEIETF streamBCP76

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

In plain English — editorial summary, not part of the RFC

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.

Document record

Document ID
RFC3666
Published
January 2004
Authors
A. Johnston; S. Donovan; R. Sparks; C. Cunningham; K. Summers
Status
BEST CURRENT PRACTICE
Stream
IETF
Area
rai
Pages
118
Also known as
BCP76

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The document record above — title, authors, date, status, stream, area, relationships, DOI and errata — is imported verbatim from the public RFC Editor index. The “in plain English” section is editorial: written by The metasystema editorial team, not part of the RFC. Where the two differ, the RFC text governs.

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canonical URL: /rfc/3666-session-initiation-protocol-sip-public-switched-telephone-network-pstn-call-flow