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RTP: Audio and Video for the Internet

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RTP: Audio and Video for the Internet

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Description

  • Copyright 2003
  • Edition: 1st
  • Book
  • ISBN-10: 0-672-32249-8
  • ISBN-13: 978-0-672-32249-5

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:

  • RTP's goals, design philosophy, and relationships with other protocols
  • The psychology of human perception in the design of media delivery systems
  • RTP data transfer and control protocols, including framing, loss detection, reception quality feedback, and membership control
  • Media playout, timing, and synchronization, including lip synchronization
  • Mitigating network problems: error concealment, error correction, and congestion control
  • Optimizing performance over low-speed links: header compression, multiplexing, and tunneling
  • Integrating leading media codecs and standards into RTP systems
  • Securing RTP sessions: encryption, authentication, and the new secure RTP profile for wireless networks
  • Extensive references and practical examples throughout


  • 0672322498B05092003

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    Table of Contents



    Preface.


    Acknowledgements.

    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.

    ITU Standards.

    Audio/Video Streaming.

    A Snapshot of RTP.

    Related Standards.

    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.

    Packet Duplication.

    Packet Corruption.

    Network Transit Time.

    Acceptable Packet Sizes.

    Effects of Multicast.

    Effects of Network Technologies.

    Conclusions about Measured Characteristics.

    Effects of Transport Protocols.

    UDP/IP.

    TCP/IP.

    Requirements for Audio/Video Transport in Packet Networks.

    Benefits of Packet-Based Audio/Video.

    Summary.

    II. MEDIA TRANSPORT USING RTP.

    3. The Real-Time Transport Protocol.

    Fundamental Design Philosophies of RTP.

    Application-Level Framing.

    The End-to-End Principle.

    Achieving Flexibility.

    Standard Elements of RTP.

    The RTP Specification.

    RTP Profiles.

    RTP Payload Formats.

    Optional Elements.

    Related Standards.

    Call Setup and Control.

    Session Description.

    Quality of Service.

    Future Standards Development.

    Summary.

    4. RTP Data Transfer Protocol.

    RTP Sessions.

    Preface

    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. 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.

    Organization of the Book

    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:

  • Chapter 1 "Introduction" gives a brief introduction to the Real-time Transport Protocol, outlines the relationship between RTP and other standards, and describes the scope of the book.
  • Chapter 2 "Audio/video communication over packet networks" describes the unique environment provided by IP networks, and how this affects packet audio/video applications.
  • 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.

  • Chapter 3 "The Real-time Transport Protocol" introduces RTP and related standards, describes how they fit together, and outlines the design philosophy underpinning the protocol.
  • Chapter 4 "RTP data transfer protocol" is a detailed description of the transport protocol, used to convey audio/visual data over IP networks.
  • Chapter 5 "RTP control protocol" describes the control protocol, provides reception quality feedback, membership control and synchronisation.
  • Chapter 6 "Media playout and timing" explains how a receiver can reconstruct the audio/visual data, and play it out to the user with correct timing.
  • Chapter 7 "Lip synchronisation" addresses a related problem: how to synchronise media streams, for example to get lip synchronisation.
  • 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.

  • Chapter 8 "Error concealment" addresses the issue of concealing errors caused by incomplete reception, describing several techniques a receiver can use to hide network problems.
  • Chapter 9 "Error correction" describes techniques that can be used to repair damaged media streams, where the sender and receiver cooperate in repairing the media stream.
  • Chapter 10 "Congestion control" discusses the way the Internet responds to congestion, and how this affects the design of audio/video applications.
  • The final section describes a number of techniques, which have more specialised use. Many implementations do not use these features, but they can give a significant performance increase in some cases:
  • Chapter 11 "Header compression" outlines a technique that can significantly increase the efficiency of RTP on low-speed network links, such as dialup modems or cellular radio links.
  • Chapter 12 "Multiplexing/tunnelling" presents alternatives to header compression, that work by combining several media streams into one. Again, the intent is to improve efficiency on low-speed links.
  • Chapter 13 "Security" describes how encryption and authentication technology can be used to protect RTP sessions; it also describes common security and privacy issues.
  • Intended Audience

    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:

  • Engineers: The primary audience is those building Voice-over-IP applications, teleconferencing systems, and streaming media and web-casting applications. This book is a guide to the design and implementation of the media engine of such systems. It should be read in conjunction with the relevant technical standards, and builds from those standards to show how a system is built. This book does not discuss signalling (e.g. SIP, RTSP or H.323)—that is a separate subject worthy of a book in its own right—instead it talks in detail about media transport, and how to achieve good quality audio and smooth motion video over IP networks.
  • Students: The book can be read as an accompaniment to a course in network protocol design or telecommunications, either at graduate or advanced undergraduate level. Familiarity with IP networks, and layered protocol architectures, is assumed. The unique aspects of protocols for real-time audio/video transport are highlighted, as are the differences from a typical layered system model. The cross-disciplinary nature of the subject is highlighted, in particular the relation between the psychology of human perception and the demands of robust media delivery.
  • Researchers: Academics and industrial researchers can use this book as a source of information about the standards and algorithms comprising the current state of the art in real-time audio/video transport over IP networks. Pointers to the literature are included in the references section, and will be useful starting points for those seeking further depth and areas where more research is needed.
  • Network Administrators: An understanding of the technical protocols underpinning the common streaming audio/video applications is useful for those administering computer networks; to show how those applications can affect the behaviour of the network, and how the network can be engineered to better suit those applications. This book includes extensive discussion of the network behaviour commonly seen, and how applications can adapt to it, the needs of congestion control, and the security implications of real-time audio/video traffic.
  • 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.



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