SIP Telephony Call Flow Examples
draft-johnston-sip-call-flows-00
Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
---|---|---|---|
Authors | Steve Donovan , Robert Sparks , Kevin Summers , Chris Cunningham | ||
Last updated | 1999-10-21 | ||
RFC stream | (None) | ||
Intended RFC status | (None) | ||
Formats | |||
Stream | Stream state | (No stream defined) | |
Consensus boilerplate | Unknown | ||
RFC Editor Note | (None) | ||
IESG | IESG state | Expired | |
Telechat date | (None) | ||
Responsible AD | (None) | ||
Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
This document gives examples of SIP IP Telephony call flows. Elements in these call flows include SIP User Agents and Clients, SIP Proxy and Redirect Servers, and Gateways to the PSTN (Public Switch Telephone Network). Scenarios include SIP Registration, SIP to SIP calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway via SIP. Call flow diagrams and message details are shown. PSTN telephony protocols are illustrated using SS7 (Signaling System 7), ISDN (Integrated Services Digital Network) and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and from private extensions served on a PBX (Private Branch Exchange).
Authors
Steve Donovan
Robert Sparks
Kevin Summers
Chris Cunningham
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)