RFC3372
Created by: dbruce,Last modification on Thu 28 of Jul, 2005 [13:42 UTC]
RFC 3372: Session Initiation Protocol for Telephones
Abstract
The popularity of gateways that interwork between the PSTN (Public
Switched Telephone Network) and SIP networks has motivated the
publication of a set of common practices that can assure consistent
behavior across implementations. This document taxonomizes the uses
of PSTN-SIP gateways, provides uses cases, and identifies mechanisms
necessary for interworking. The mechanisms detail how SIP provides
for both 'encapsulation' (bridging the PSTN signaling across a SIP
network) and 'translation' (gatewaying).
Switched Telephone Network) and SIP networks has motivated the
publication of a set of common practices that can assure consistent
behavior across implementations. This document taxonomizes the uses
of PSTN-SIP gateways, provides uses cases, and identifies mechanisms
necessary for interworking. The mechanisms detail how SIP provides
for both 'encapsulation' (bridging the PSTN signaling across a SIP
network) and 'translation' (gatewaying).
RFC 3372 - Sesseion Initiation Protocol for Telephones
This is a "Best Practices" document that describes a methodology for encapsulating ISUP (ISDN User Part) messages within the SIP protocol. This facilitates the transmittion of call setup components between SIP gateways without loss of signalling components not normally used in a SIP only environment.
components
RFC 3261: Session Initiation ProtocolRFC 3302: MIME media types for ISUP and QSIG objects
RFC 2633: S/MIME Version 3 Message Specification - new MIME type - application/isup
RFC 2976: SIP INFO method
RFC 2327: SDP: Session Description Protocol
RFC 3219: TRIP: Telephone Routing over Internet Protocol
Comments