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BCP 76, RFC 3666 on Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
A new Request for Comments is now available in online RFC libraries.
Title: Session Initiation Protocol (SIP)
Public Switched Telephone Network (PSTN) Call Flows
Author(s): A. Johnston, S. Donovan, R. Sparks, C. Cunningham,
Status: Best Current Practice
Date: December 2003
Mailbox: email@example.com, firstname.lastname@example.org,
SeeAlso: BCP 76
I-D Tag: draft-ietf-sipping-pstn-call-flows-02.txt
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.
This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.
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.
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Authors, for further information.
Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute
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