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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Colin Perkins is a research assistant professor at the University of Southern California Information Sciences Institute, where his research interests include scaling Internet multimedia conferencing to support very large distributed meetings and to very high quality. From 1996 to 2000, he was a research fellow with the Department of Computer Science, University College, London, where he conducted research into advanced VoIP and IP-based videoconferencing technologies, and developed one of the earliest RTP teleconferencing implementations. He is co-chair of the Audio/Video Transport and Multiparty Multimedia Session Control working groups of the IETF, and has authored several RFC standards relating to RTP. He holds a Ph.D. in electronic engineering from the University of York.
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.
Organization of the Book
The book is logically divided into four parts: Part I, Introduction to Networked Multimedia, introduces the problem space, provides background, and outlines the properties of the Internet that affect audio/video transport:
The next five chapters, which constitute Part II, Media Transport Using RTP, discuss the basics of the Real-time Transport Protocol. You will need this information to design and build a tool for voice-over-IP, streaming music or video, and so on.
Part III, Robustness, discusses 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 the audio will sound a lot better, and the pictures will be smoother and less susceptible to corruption, if you apply them.
Finally, Part IV, Advanced Topics, describes various techniques that have more specialized use. Many implementations do not use these features, but they can significantly improve performance in some cases:
This book describes audio/video transport over IP networks in considerable detail. It assumes some basic familiarity with IP net-work programming and the operation of network protocol stacks, and it 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.
Several classes of readers 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 webcasting 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 it builds on those standards to show how a system is built. This book does not discuss signaling (for example, SIP, RTSP, or H.323), which 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, at either a graduate or an 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 that constitute 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 they 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 behavior of the network, and how the network can be engineered to suit those applications better. This book includes extensive discussion of the most common network behavior (and how applications can adapt to it), the needs of congestion control, and the security implications of real-time audio/video traffic.
In summary, 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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