RFC 3372 (rfc3372) - Page 2 of 23

Session Initiation Protocol for Telephones (SIP-T): Context and Architectures

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



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