3.1 Introduction

This chapter describes the Session Initiation Protocol (SIP) in some detail. The anticipation is that an SIP will play a key role in 3G VoIP networks based on IPv6; hence, the coverage we allocate to this topic. An SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols. Besides basic VoIP telephony, SIP can support presence/proximity, multimodal and collaborative communications.

We stressed in Chapter 2 the importance of signaling. Signaling is a critical constituent element of a truly-global VoIP construct that is able to support public telephony. SIP can in principle be leveraged to this end. This (relatively) new protocol is described in a series of IETF RFCs, as shown in Table 3.1: