Voice and Video Conferencing Fundamentals

Scott Firestone, Thiya Ramalingam, Steve Fry

Voice and Video Conferencing Fundamentals
  • 出版商: Cisco Press
  • 出版日期: 2007-03-01
  • 售價: $3,010
  • 貴賓價: 9.5 折 $2,860
  • 語言: 英文
  • 頁數: 408
  • 裝訂: Paperback
  • ISBN: 1587052687
  • ISBN-13: 9781587052682

    • 海外代購書籍(需單獨結帳)

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Voice and Video Conferencing Fundamentals

 

Design, develop, select, deploy, and support advanced IP-based audio and video conferencing systems

 

Scott Firestone, Thiya Ramalingam, Steve Fry

 

As audio and video conferencing move rapidly into the mainstream, customers and end users are demanding unprecedented performance, reliability, scalability, and security. In Voice and Video Conferencing Fundamentals, three leading experts systematically introduce the principles, technologies, and protocols underlying today’s state-of-the-art conferencing systems. Discover how to use these concepts and techniques to deliver unified, presence-enabled services that integrate voice, video, telephony, networks, and the Internet–and enable breakthrough business collaboration.

 

The authors begin with a clear, concise overview of current voice and video conferencing, including system components, operational modes, endpoints, features, and user interactivity. Next, they illuminate conferencing architectures, offering practical insights for designing today’s complex IP-based conferencing and collaboration systems.

 

Topics covered in this book include video codecs, media control, SIP and H.323 protocols and applications, lip synchronization in video conferencing, security, and much more. Throughout the book, the authors draw on their extensive experience as Cisco® technical leaders, showing how to avoid the most common pitfalls that arise in planning, deployment, and administration.

 

Voice and Video Conferencing Fundamentals is for every professional involved with audio or video conferencing: network and system administrators, engineers, technology managers, and Cisco solution partners alike. Whether you’re involved with design, development, selection, implementation, management, or support, you’ll find the in-depth knowledge you need to succeed.

 

Scott Firestone holds a master’s degree in computer science from MIT and has designed video conferencing and voice products since 1992, resulting in five patents. Thiya Ramalingam is an engineering manager for the Cisco Unified Communications organization. Thiya holds a master’s degree in computer engineering and an MBA degree from San Jose State University. Steve Fry, a technical leader in the Cisco Unified Communication organization, has spent the last several years designing and developing telephony and conferencing products.

 

  • Thoroughly understand the fundamentals of audio and video conferencing over IP networks
  • Architect networks for optimal performance and reliability in conferencing applications
  • Leverage new advances in video architecture, from emerging codecs to distributed implementations
  • Understand how SIP and H.323 compare, and when to use each
  • Optimize synchronization between audio and video
  • Secure conferencing traffic without compromising performance or connectivity
  • Learn how to evaluate vendors and make better buying decisions

 

This book is part of the Cisco Press® Fundamentals Series. Books in this series introduce networking professionals to new networking technologies, covering network topologies, sample deployment concepts, protocols, and management techniques.

 

 

 

Table of Contents

Foreword xviii

Introduction xix

Chapter 1 Overview of Conferencing Services 3

Conference Types 3

    Ad Hoc Conferences 4

    Reservationless Conferences 5

    Scheduled Conferences 6

Voice and Video Conferencing Components 9

Video Conferencing Modes 11

    Voice-Activated Conferences 11

    Continuous Presence Conferences 13

    Lecture Mode and Round-Robin Conferences 15

Types of Endpoints 16

    Desktop Conferencing Systems 16

    Room Conferencing Systems 16

    Telepresence Systems 16

Video Controls: Far-End Camera Control 17

Text Overlay 18

Summary 18

 

Chapter 2 Conferencing System Design and Architecture 21

Components of a Conferencing System 21

    User Interface 23

    Conference Control 25

    Control Plane 26

    Media Plane 27

Conferencing Architectures 37

    Centralized Architecture 37

    Distributed Architecture 38

    Full-Mesh Networks 40

Advanced Conferencing Scenarios 41

    Escalation of Point-to-Point-to-Multipoint Call 41

    Lecture Mode Conferences 41

    Panel Mode Conference 42

    Floor Control 42

    Video Mixing and Switching Scenarios 42

Summary 43

References 43

 

Chapter 3 Fundamentals of Video Compression 45

Evaluating Video Quality, Bit Rate, and Signal-to-Noise Ratio 45

Video Source Formats 47

    Profiles and Levels 47

    Frame Rates, Form Factors, and Layouts 47

    Standard and High Definitions 48

    Color Formats 49

Basics of Video Coding 52

    Preprocessing 52

    Post-Processing 54

    Encoder Overview 55

Hybrid Coding 72

    Hybrid Decoder 72

    P-Frames 74

    Hybrid Encoder 74

    Predictor Loop 76

    Motion Estimation 77

    B-Frames 82

    Predictor Loops for Parameters 86

    Error Resiliency 88

Scalable Layered Codecs 91

    SNR and Spatial Scalability 93

    Temporal Scalability 95

Switching Frames 99

Video Codecs 100

    Video Stream Hierarchy 100

    Macroblocks 101

    HD-Capable Codecs 102

Summary 102

References 103

 

Chapter 4 Media Control and Transport 105

Overview of RTP 105

    RTP Header 107

    RTP Port Numbers 111

    SSRC Collisions 111

    RTP Header Extensions 112

Overview of RTCP 113

    RTCP Packet Headers 113

    RTCP Sender Report 114

    RTCP Receiver Report 116

    RTCP Source Description (SDES) 117

    RTCP BYE 119

    RTCP APP 120

RTP Devices in Conference Systems 120

    RTP Translator 120

    RTP Mixer 123

    Audio Mixer 123

    Video MCU 124

    Video Switcher 124

Video Stream RTP Formats 126

    H.263 126

    H.264 133

Detecting Stream Loss 141

Summary 142

References 143

 

Chapter 5 Signaling Protocols: Conferencing Using SIP 145

SIP Overview 145

    User Agent 146

    Proxy Server 146

    Redirect Server 147

    Registrar 147

SIP Transactions and Dialogs 148

SIP Messages 149

    SIP Requests 149

    SIP Responses 152

SIP Record Routing 153

Event Subscription and Notification 154

Session Description Protocol 155

SIP Conferencing Models 157

    Conference URI 157

    Early and Delayed Offer 158

    DTMF Support 159

Ad Hoc Audio Conferencing 160

Ad Hoc Video Conferencing 162

    Video SDP Extensions 163

    Bandwidth Information in the SDP 167

    Multiple Stream Support and Grouping of Media Lines 168

    Escalation and De-escalation 169

    Media Control Support 172

Scheduled Conferences 173

    Entry IVR 174

    In-Conference Features 177

    Roll Call 177

    Hold and Resume 178

    Mute and Unmute 179

    Outdial 179

RSVP/QoS Support in Conferencing Flows 180

Summary 182

References 183

 

Chapter 6 Signaling Protocols: Conferencing Using H.323 185

H.323 Overview 185

H.323 Endpoint Aliasing 187

H.225 Call Signaling 188

    H.225 Message Format 188

    Common H.225 Message Types Used in H.323 Signaling 189

H.245 Control Protocol 191

    H.245 Messages 192

    Video-Specific H.245 Messages 202

H.323 Fast Connect Mode 204

Using the Empty Capability Set 207

    Call Hold Signaling with the Empty Capability Set 207

    Call Transfer with the Empty Capability Set 207

H.323 Device Types 208

H.323 Gatekeeper Services 209

    Required H.323 Gatekeeper Features 209

    Optional H.323 Gatekeeper Features 210

    Gatekeeper Signaling Options 211

    Gatekeeper RAS Signaling 212

    Mid-Call Bandwidth Requests 214

    Configuring a Gatekeeper in Cisco Unified CallManager 215

    Configuring Gatekeeper Support in a Cisco IOS Router 217

    H.225 Call Setup for Video Devices Using a Gatekeeper 217

Using Service Prefixes with MCUs 219

Summary 220

References 220

 

Chapter 7 Lip Synchronization in Video Conferencing 223

Understanding Lip Sync Skew 223

    Human Perceptions 223

    Measuring Skew 225

    Delay Accumulation 226

    Delays in the Network Path 228

Lip Sync Approaches 229

    Poor Man’s Lip Sync 230

    Common Reference Lip Sync 232

Understanding the Sender Side 232

    Sender Audio Path 233

    Video Source Format 235

    Sender Video Path 238

Understanding the Receive Side 241

    Audio Receiver Path 241

    Receiver Video Path 243

    Types of Playout Devices 244

RTP 244

    Canonical RTP Model 244

    RTP Time Stamps 246

    Using RTP for Buffer-Level Management 247

Correlating Timebases Using RTCP 250

    NTP 250

    Forming RTCP Packets 251

    Using RTCP for Media Synchronization 252

    Lip Sync Policy 254

Summary 255

References 255

 

Chapter 8 Security Design in Conferencing 257

Security Fundamentals 257

Threats 258

    Confidentiality Attacks 258

    Denial-of-Service Attacks 259

    Authentication and Identity Attacks 262

    Network Infrastructure Attacks 263

    Endpoint Infrastructure Attacks 266

    Server Attacks 267

Configuring Basic Security 269

Port Usage 270

    H.323 Port Usage 270

    SIP Port Usage 275

    SCCP Port Usage 275

    Preset Port Numbers 276

NAT and PAT 276

    NAT Classifications 277

    NAT Complications for VoIP Protocols 284

    NAT ALGs 285

    NAT/FW Traversal Solutions 285

Encryption Basics 299

    Symmetric Encryption 299

    Secure Hashes 299

    Asymmetric Encryption: Public Key Cryptography 300

    Nonrepudiation 309

    Key Distribution 309

IPsec and TLS for Secure Signaling 310

    IPsec 311

    TLS 311

Media Encryption 312

    security-descriptions 312

    MIKEY 313

H.323 Encryption: H.235 313

    H.235.1 314

    H.235.2 316

    H.235.3 319

    H.235.6 319

SIP Encryption 321

    SIP-Digest 321

    SCCP Encryption 324

Summary 324

References 325

 

Appendix A Video Codec Standards 327

商品描述(中文翻譯)

描述

語音和視訊會議基礎知識

設計、開發、選擇、部署和支援先進的基於IP的音頻和視頻會議系統

Scott Firestone, Thiya Ramalingam, Steve Fry

隨著語音和視訊會議迅速走入主流，客戶和最終用戶對性能、可靠性、可擴展性和安全性的要求前所未有。在《語音和視訊會議基礎知識》中，三位領先的專家系統地介紹了當今最先進的會議系統背後的原則、技術和協議。了解如何使用這些概念和技術提供統一的、支持存在的服務，將語音、視頻、電話、網絡和互聯網整合起來，實現突破性的業務協作。

作者首先清晰、簡明地概述了當前的語音和視訊會議，包括系統組件、操作模式、終端、功能和用戶互動。接下來，他們闡明了會議架構，提供了設計當今複雜的基於IP的會議和協作系統的實用見解。

本書涵蓋的主題包括視頻編解碼器、媒體控制、SIP和H.323協議和應用、視訊會議中的嘴唇同步、安全性等等。在整本書中，作者們借鑒了他們作為思科技術領導者的豐富經驗，展示了如何避免在規劃、部署和管理中常見的陷阱。

《語音和視訊會議基礎知識》適用於與音頻或視訊會議相關的每一位專業人士：網絡和系統管理員、工程師、技術經理和思科解決方案合作夥伴。無論您參與設計、開發、選擇、實施、管理還是支援，您都能找到所需的深入知識以取得成功。

Scott Firestone擁有麻省理工學院的計算機科學碩士學位，自1992年以來一直設計視訊會議和語音產品，並取得了五項專利。Thiya Ramalingam是思科統一通信組織的工程經理。Thiya擁有舊金山州立大學的計算機工程碩士學位和MBA學位。Steve Fry是思科統一通信組織的技術領導者，他在過去幾年中一直設計和開發電話和會議產品。

全面了解基於IP網絡的音頻和視訊會議的基礎知識
在會議應用中為網絡架構提供最佳性能和可靠性
利用新的視頻架構進行優化，從新興編解碼器到分佈式實現
了解SIP和H.323的比較，以及何時使用每個協議
優化音頻和視頻之間的同步
在不影響性能或連接性的情況下保護會議流量
學習如何評估v

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