RFC 3372 (rfc3372) - Page 2 of 23
Session Initiation Protocol for Telephones (SIP-T): Context and Architectures
Alternative Format: Original Text Document
RFC 3372 SIP-T September 2002 5.4 Support for mid-call signaling . . . . . . . . . . . . . . . . 17 6. SIP Content Negotiation . . . . . . . . . . . . . . . . . . . 17 7. Security Considerations . . . . . . . . . . . . . . . . . . . 19 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20 9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20 10 References . . . . . . . . . . . . . . . . . . . . . . . . . . 20 A. Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21 B. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 21 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22 Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 23 1. Introduction The Session Initiation Protocol (SIP [1]) is an application-layer control protocol that can establish, modify and terminate multimedia sessions or calls. These multimedia sessions include multimedia conferences, Internet telephony and similar applications. SIP is one of the key protocols used to implement Voice over IP (VoIP). Although performing telephony call signaling and transporting the associated audio media over IP yields significant advantages over traditional telephony, a VoIP network cannot exist in isolation from traditional telephone networks. It is vital for a SIP telephony network to interwork with the PSTN. The popularity of gateways that interwork between the PSTN and SIP networks has motivated the publication of a set of common practices that can assure consistent behavior across implementations. The scarcity of SIP expertise outside the IETF suggests that the IETF is the best place to stage this work, especially since SIP is in a relative state of flux compared to the core protocols of the PSTN. Moreover, the IETF working groups that focus on SIP (SIP and SIPPING) are best positioned to ascertain whether or not any new extensions to SIP are justified for PSTN interworking. This framework addresses the overall context in which PSTN-SIP interworking gateways might be deployed, provides use cases and identifies the mechanisms necessary for interworking. An important characteristic of any SIP telephony network is feature transparency with respect to the PSTN. Traditional telecom services such as call waiting, freephone numbers, etc., implemented in PSTN protocols such as Signaling System No. 7 (SS7 [6]) should be offered by a SIP network in a manner that precludes any debilitating difference in user experience while not limiting the flexibility of SIP. On the one hand, it is necessary that SIP support the primitives for the delivery of such services where the terminating point is a regular SIP phone (see definition in Section 2 below) rather than a device that is fluent in SS7. However, it is also essential that SS7 information be available at gateways, the points Vemuri & Peterson Best Current Practice
