RFC 3372: Session Initiation Protocol for Telephones


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).

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.


RFC 3261: Session Initiation Protocol
RFC 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

See Also:


Created by: dbruce, Last modification: Thu 28 of Jul, 2005 (13:42 UTC)
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