Voice and Video Conferencing Fundamentals
暫譯: 語音與視頻會議基礎知識
Scott Firestone, Thiya Ramalingam, Steve Fry
- 出版商: Cisco Press
- 出版日期: 2007-03-01
- 售價: $3,070
- 貴賓價: 9.5 折 $2,917
- 語言: 英文
- 頁數: 408
- 裝訂: Paperback
- ISBN: 1587052687
- ISBN-13: 9781587052682
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商品描述
Description
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**
*語音與視訊會議基礎*
設計、開發、選擇、部署及支援先進的基於 IP 的音訊與視訊會議系統。
Scott Firestone, Thiya Ramalingam, Steve Fry
隨著音訊與視訊會議迅速進入主流，客戶和最終用戶對性能、可靠性、可擴展性和安全性提出了前所未有的要求。在《語音與視訊會議基礎》中，三位領先的專家系統性地介紹了當今最先進會議系統的原則、技術和協議。了解如何使用這些概念和技術來提供統一的、具存在感的服務，整合語音、視訊、電話、網路和互聯網，並促進突破性的商業協作。
作者首先清晰、簡明地概述了當前的語音與視訊會議，包括系統組件、操作模式、端點、功能和用戶互動。接著，他們闡明了會議架構，提供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訊編解碼器、媒體控制、SIP 和 H.323 協議及應用、視訊會議中的嘴型同步、安全性等。全書中，作者利用他們作為 Cisco® 技術領導者的豐富經驗，展示如何避免在規劃、部署和管理中出現的最常見陷阱。
《語音與視訊會議基礎》適合所有參與音訊或視訊會議的專業人士：網路和系統管理員、工程師、技術經理以及 Cisco 解決方案合作夥伴。不論您參與設計、開發、選擇、實施、管理或支援，您都會找到成功所需的深入知識。
**Scott Firestone** 擁有麻省理工學院的計算機科學碩士學位，自 1992 年以來設計視訊會議和語音產品，獲得五項專利。**Thiya Ramalingam** 是 Cisco 統一通信組織的工程經理，擁有計算機工程碩士學位和聖荷西州立大學的 MBA 學位。**Steve Fry** 是 Cisco 統一通信組織的技術領導者，過去幾年專注於設計和開發電話和會議產品。
- 徹底了解基於 IP 網路的音訊和視訊會議基礎
- 為會議應用架構設計最佳性能和可靠性的網路
- 利用視訊架構的新進展，從新興編解碼器到分散式實現
- 了解 SIP 和 H.323 的比較，以及何時使用各自
- 優化音訊與視訊之間的同步
- 在不妥協性能或連接性的情況下保護會議流量
- 學習如何評估供應商並做出更好的購買決策
本書是 Cisco Press® 基礎系列的一部分。本系列的書籍向網路專業人士介紹新的網路技術，涵蓋網路拓撲、示範部署概念、協議和管理技術。
**目錄**
前言 xviii
導言 xix
第 1 章 會議服務概述 3
會議類型 3
臨時會議 4
無需預約的會議 5
預定會議 6
語音與視訊會議組件 9
視訊會議模式 11
語音啟動會議 11
持續存在會議 13
講座模式和輪流會議 15
端點類型 16
桌面會議系統 16
會議室系統 16
遠端會議系統 16
視訊控制：遠端攝影機控制 17
文字覆蓋 18
摘要 18
第 2 章 會議系統設計與架構 21
會議系統的組件 21
用戶介面 23
會議控制 25
控制平面 26
媒體平面 27
會議架構 37
集中式架構 37
分散式架構 38
全網狀網路 40
進階會議場景 41
點對點到多點通話的升級 41
講座模式會議 41
小組模式會議 42
發言控制 42
視訊混合與切換場景 42
摘要 43
參考文獻 43
第 3 章 視訊壓縮基礎 45
評估視訊質量、比特率和信噪比 45
視訊來源格式 47
配置文件和級別 47
幀率、形狀因子和佈局 47
標準和高清 48
顏色格式 49
視訊編碼基礎 52
預處理 52
後處理 54
編碼器概述 55
混合編碼 72
混合解碼器 72
P 幀 74
混合編碼器 74
預測迴路 76
運動估計 77
B 幀 82
參數的預測迴路 86
錯誤恢復 88
可擴展分層編解碼器 91
信噪比和空間可擴展性 93
時間可擴展性 95
切換幀 99
視訊編解碼器 100
視訊流層次結構 100
宏塊 101
支持高清的編解碼器 102
摘要 102
參考文獻 103
第 4 章 媒體控制與傳輸 105
