The Real-time Transport Protocol (RTP) provides a framework for delivery of audio and video across IP networks with unprecedented quality and reliability. In RTP: Audio and Video for the Internet, Colin Perkins, a leader of the RTP standardization process in the IETF, offers readers detailed technical guidance for designing, implementing, and managing any RTP-based system.
By bringing together crucial information that was previously scattered or difficult to find, Perkins has created an incredible resource that enables professionals to leverage RTP's benefits in a wide range of Voice-over IP (VoIP) and streaming media applications. He demonstrates how RTP supports audio/video transmission in IP networks, and shares strategies for maximizing performance, robustness, security, and privacy.
Comprehensive, exceptionally clear, and replete with examples, this book is the definitive RTP reference for every audio/video application designer, developer, researcher, and administrator.
Key coverage includes:
Download the Sample Chapter related to this title.
I. INTRODUCTION TO NETWORKED MULTIMEDIA.1. An Introduction To RTP.
A Brief History of Audio/Video Networking.
Early Packet Voice and Video Experiments.
Audio and Video on the Internet.
A Snapshot of RTP.
Overview of an RTP Implementation.
Behavior of an RTP Sender.
Behavior of an RTP Receiver.
Summary.2. Voice And Video Communication Over Packet Networks.
TCP/IP and the OSI Reference Model.
Performance Characteristics of an IP Network.
Measuring IP Network Performance.
Average Packet Loss.
Packet Loss Patterns.
Network Transit Time.
Acceptable Packet Sizes.
Effects of Multicast.
Effects of Network Technologies.
Conclusions about Measured Characteristics.
Effects of Transport Protocols.
Requirements for Audio/Video Transport in Packet Networks.
Benefits of Packet-Based Audio/Video.
II. MEDIA TRANSPORT USING RTP.3. The Real-Time Transport Protocol.
Fundamental Design Philosophies of RTP.
The End-to-End Principle.
Standard Elements of RTP.
The RTP Specification.
RTP Payload Formats.
Call Setup and Control.
Quality of Service.
Future Standards Development.
Summary.4. RTP Data Transfer Protocol.
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. This includes voice-over-IP, telephony, teleconferencing, streaming video and web-casting applications. It focuses on media transport: how to reliably deliver audio and video across an IP network, how to ensure high quality in the face of network problems, and how to ensure 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.
There are many media codecs that are suitable for use with RTP, for example MPEG audio and video, ITU H.261 and H.263 video, and G.711, G.722, G.726, G.728 and G.729 audio, and industry standards such as GSM, QCELP and AMR audio. An RTP implementation typically integrates an existing media codec, rather than developing them specifically. Accordingly, this book describes how media codecs are integrated into an RTP system, but not how media codecs are design.
Call set-up, 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 signalling. Similarly, session description using SDP is not covered, being part of the signalling.
Resource reservation is useful in some situations, but not required for correct operation of RTP. This book touches on the use of resource reservation through the both Integrated Services and Differentiated Services frameworks, but 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 set-up, 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 web-casting application. This book describes only the media transport layer, common to all those systems.
The book is logically divided into four sections: The first section introduces the problem space, provides background, and outlines the properties of the Internet that affect audio/video transport:
The next five chapters discuss the basics of the Real-time Transport Protocol. This is information you need to design and build a tool for voice-over-IP, streaming music or video, etc.
The third part of the book discusses robustness: how to make your application reliable in the face of network problems, and how to compensate for loss and congestion in the network. You can build a system without using these techniques, but it'll sound a lot better, and the pictures will be smoother and less susceptible to corruption, if you apply them.
This book describes audio/video transport over IP networks in considerable detail. It assumes some basic familiarity with IP network programming, and the operation of network protocol stacks, and builds on this knowledge to describe the features unique to audio/video transport. An extensive list of references is included, pointing readers to additional information on specific topics and to background reading material.
There are several classes of reader who might be expected to find this book useful:
This book can be used as a reference, in conjunction with the technical standards, as a study guide or as part of an advanced course on network protocol design or communication technology.
Download the Index
file related to this title.