Introduction

This book describes the protocols, standards, and architecture of systems that deliver real-time voice, music, and video over IP networks, such as the Internet. These systems include voice-over-IP, telephony, teleconferencing, streaming video, and webcasting applications. The book focuses on media transport: how to deliver audio and video reliably across an IP network, how to ensure high quality in the face of network problems, and how to ensure that the system is secure.

The book adopts a standards-based approach, based around the Real-time Transport Protocol (RTP) and its associated profiles and payload formats. It describes the RTP framework, how to build a system that uses that framework, and extensions to RTP for security and reliability.

Many media codecs are suitable for use with RTP ”for example, MPEG audio and video; ITU H.261 and H.263 video; G.711, G.722, G.726, G.728, and G.729 audio; and industry standards such as GSM, QCELP, and AMR audio. RTP implementations typically integrate existing media codecs, rather than developing them specifically . Accordingly, this book describes how media codecs are integrated into an RTP system, but not how media codecs are designed.

Call setup, session initiation, and control protocols, such as SIP, RTSP, and H.323, are also outside the scope of this book. Most RTP implementations are used as part of a complete system, driven by one of these control protocols. However, the interactions between the various parts of the system are limited, and it is possible to understand media transport without understanding the signaling. Similarly, session description using SDP is not covered, because it is part of the signaling.

Resource reservation is useful in some situations, but it is not required for the correct operation of RTP. This book touches on the use of resource reservation through both the Integrated Services and the Differentiated Services frameworks, but it does not go into details.

That these areas are not covered in this book does not mean that they are unimportant. A system using RTP will use a range of media codecs and will employ some form of call setup, session initiation, or control. The way this is done depends on the application, though: The needs of a telephony system are very different from those of a webcasting application. This book describes only the media transport layer that is common to all those systems.