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E-raamat: Guide to Voice and Video over IP: For Fixed and Mobile Networks

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Guide to Voice and Video over IP: For Fixed and Mobile NetworksSuurem pilt
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This book covers advances in speech and video compression, assessment and monitoring of VoIP quality and next-generation network architecture for multimedia services. Offers case studies with step-by-step instructions, worked examples and chapter-end problems.

This book presents a review of the latest advances in speech and video compression, computer networking protocols, the assessment and monitoring of VoIP quality, and next generation network architectures for multimedia services. The book also concludes with three case studies, each presenting easy-to-follow step-by-step instructions together with challenging hands-on exercises. Features: provides illustrative worked examples and end-of-chapter problems; examines speech and video compression techniques, together with speech and video compression standards; describes the media transport protocols RTP and RTCP, as well as the VoIP signalling protocols SIP and SDP; discusses the concepts of VoIP quality of service and quality of experience; reviews next-generation networks based on the IP multimedia subsystem and mobile VoIP; presents case studies on building a VoIP system based on Asterisk, setting up a mobile VoIP system based on Open IMS and Android mobile, and analysing VoIP protocols and quality.

Arvustused

From the reviews:

This book guides readers through the technologies and applications of voice and video over Internet protocol (VoIP). It is a comprehensive work based on the teaching and research experiences of the authors, and an excellent resource for understanding and practicing key techniques and tools for VoIP. This book could serve as a textbook for students or a reference for practitioners. (Hari Vishwakarma, Computing Reviews, July, 2013)

1 Introduction
1(16)
1.1 Overview of VoIP
1(2)
1.2 How VoIP Works and Factors That Affect Quality
3(1)
1.3 VoIP Tools
4(5)
1.3.1 Microsoft's Lync
4(1)
1.3.2 Skype
5(2)
1.3.3 Google Talk
7(1)
1.3.4 X-Lite
8(1)
1.4 VoIP Trend
9(3)
1.5 VoIP Protocol Stack and the Scope of the Book
12(2)
1.6 Summary
14(3)
2 Speech Compression
17(36)
2.1 Introduction
17(1)
2.2 Speech Compression Basics
18(7)
2.2.1 Speech Signal Digitisation
18(3)
2.2.2 Speech Waveform and Spectrum
21(2)
2.2.3 How Is Human Speech Produced?
23(2)
2.3 Speech Compression and Coding Techniques
25(11)
2.3.1 Waveform Compression Coding
26(2)
2.3.2 Parametric Compression Coding
28(3)
2.3.3 Hybrid Compression Coding---Analysis-by-Synthesis
31(4)
2.3.4 Narrowband to Fullband Speech Audio Compression
35(1)
2.4 Standardised Narrowband to Fullband Speech/Audio Codecs
36(9)
2.4.1 ITU-T G.711 PCM and G.711.1 PCM-WB
36(1)
2.4.2 ITU-T G.726 ADPCM
37(1)
2.4.3 ITU-T G.728 LD-CELP
38(1)
2.4.4 ITU-T G.729 CS-ACELP
38(1)
2.4.5 ITU-T G.723.1 MP-MLQ/ACELP
39(1)
2.4.6 ETSI GSM
39(1)
2.4.7 ETSI AMR
40(1)
2.4.8 IETF's iLBC
41(1)
2.4.9 Skype/IETF's SILK
41(1)
2.4.10 ITU-T G.722 ADPCM-WB
42(1)
2.4.11 ITU-T G.722.1 Transform Coding
43(1)
2.4.12 ETSI AMR-WB and 1TU-T G.722.2
44(1)
2.4.13 ITU-T G.719 Fullband Audio Coding
44(1)
2.4.14 Summary of Narrowband to Fullband Speech Codecs
45(1)
2.5 Illustrative Worked Examples
45(3)
2.5.1 Question 1
45(2)
2.5.2 Question 2
47(1)
2.5.3 Question 3
47(1)
2.6 Summary
48(1)
2.7 Problems
49(4)
3 Video Compression
53(20)
3.1 Introduction to Video Compression
53(2)
3.2 Video Compression Basics
55(3)
3.2.1 Digital Image and Video Colour Components
55(1)
3.2.2 Colour Sub-sampling
56(1)
3.2.3 Video Resolution and Bandwidth Requirement
57(1)
3.3 Video Compression Techniques
58(1)
3.4 Lossless Video Compression
58(1)
3.5 Lossy Video Compression
59(4)
3.5.1 Predictive Coding
59(1)
3.5.2 Quantisation
60(1)
3.5.3 Transform Coding
61(1)
3.5.4 Interframe Coding
62(1)
3.6 Video Coding Standards
63(6)
3.6.1 H+120
63(1)
3.6.2 H.261
63(1)
3.6.3 MPEG 1 & 2
64(3)
3.6.4 H.263
67(1)
3.6.5 MPEG-4
68(1)
3.6.6 H.264
68(1)
3.6.7 Highly Efficiency Video Coding (HEVC)
69(1)
3.7 Illustrative Worked Examples
69(2)
3.7.1 Question 1
69(1)
3.7.2 Question 2
70(1)
3.7.3 Question 3
71(1)
3.8 Summary
71(1)
3.9 Problems
72(1)
4 Media Transport for VoIP
73(28)
4.1 Media Transport over IP Networks
73(1)
4.2 TCP or UDP?
74(2)
4.3 Real-Time Transport Protocol---RTP
76(12)
4.3.1 RTP Header
76(2)
4.3.2 RTP Header for Voice Call Based on Wireshark
78(2)
4.3.3 RTP Payload and Bandwidth Calculation for VoIP
80(3)
4.3.4 Illustrative Worked Example
83(1)
4.3.5 RTP Header for Video Call Based on Wireshark
84(4)
4.4 RTP Control Protocol---RTCP
88(8)
4.4.1 RTCP Sender Report and Example
89(2)
4.4.2 RTCP Receiver Report and Example
91(1)
4.4.3 RTCP Source Description and Example
92(2)
4.4.4 RTCP BYE Packet and Example
94(1)
4.4.5 Extended RTCP Report---RTCP XR for VoIP Metrics
95(1)
4.5 Compressed RTP---cRTP
96(2)
4.5.1 Basic Concept of Compressed RTP---cRTP
96(2)
4.5.2 Illustrative Worked Example
98(1)
4.6 Summary
98(1)
4.7 Problems
99(2)
5 VoIP Signalling---SIP
101(22)
5.1 What is Session Initiation Protocol?
101(5)
5.1.1 SIP Network Elements
102(1)
5.1.2 User Agent
103(1)
5.1.3 Proxy Server
104(1)
5.1.4 Redirect Server
105(1)
5.1.5 Registrar
105(1)
5.1.6 Location Server
106(1)
5.2 SIP Protocol Structure
106(7)
5.2.1 SIP Message Format
107(6)
5.3 Session Descriptions Protocol
113(4)
5.3.1 Session Description
114(1)
5.3.2 Time Description
115(1)
5.3.3 Media Description
115(1)
5.3.4 Attributes
116(1)
5.3.5 Example of SDP Message from Wireshark
117(1)
5.4 SIP Messages Flow
117(3)
5.4.1 Session Establishment
118(2)
5.5 Summary
120(1)
5.6 Problems
120(3)
6 VoIP Quality of Experience (QoE)
123(40)
6.1 Concept of Quality of Service (QoS)
123(12)
6.1.1 What is Quality of Service (QoS)?
123(1)
6.1.2 QoS Metrics and Measurements
124(1)
6.1.3 Network Packet Loss and Its Characteristics
125(5)
6.1.4 Delay, Delay Variation (Jitter) and Its Characteristics
130(5)
6.2 Quality of Experience (QoE) for VoIP
135(3)
6.2.1 What is Quality of Experience (QoE)?
135(1)
6.2.2 Factors Affect Voice Quality in VoIP
136(1)
6.2.3 Overview of QoE for Voice and Video over IP
137(1)
6.3 Subjective Speech Quality Assessment
138(3)
6.4 Objective Speech Quality Assessment
141(7)
6.4.1 Comparison-Based Intrusive Objective Test (Full-Reference Model)
141(4)
6.4.2 Parameter-Based Measurement: E-Model
145(2)
6.4.3 A Simplified and Applicable E-Model
147(1)
6.5 Subjective Video Quality Assessment
148(2)
6.6 Objective Video Quality Assessment
150(5)
6.6.1 Full-Reference (FR) Video Quality Assessment
150(3)
6.6.2 Reduced-Reference (RR) Video Quality Assessment
153(1)
6.6.3 No-Reference Video Quality Assessment
154(1)
6.7 Illustrative Worked Examples
155(3)
6.7.1 Question 1
155(1)
6.7.2 Question 2
156(1)
6.7.3 Question 3
157(1)
6.8 Summary
158(1)
6.9 Problems
159(4)
7 IMS and Mobile VoIP
163(30)
7.1 What Is IP Multimedia Subsystem?
163(14)
7.1.1 What Do We Need IMS for?
163(1)
7.1.2 IMS Architecture
164(3)
7.1.3 IMS Elements
167(6)
7.1.4 IMS Services
173(1)
7.1.5 IMS Signalling and Bearer Traffic Interfaces
174(3)
7.2 Mobile Access Networks
177(11)
7.2.1 Cellular Standards
178(1)
7.2.2 The GSM Standard
179(2)
7.2.3 The UMTS Standard
181(5)
7.2.4 Long-Term Evolution
186(2)
7.3 Summary
188(1)
7.4 Problems
189(4)
8 Case Study 1---Building Up a VoIP System Based on Asterisk
193(22)
8.1 What is Asterisk?
193(4)
8.1.1 Channel Modules
194(1)
8.1.2 Codec Translator Modules
194(1)
8.1.3 Application Modules
195(1)
8.1.4 File Format Modules
195(1)
8.1.5 Installing Asterisk
196(1)
8.2 What Is X-Lite 4
197(4)
8.2.1 Using X-Lite
197(4)
8.3 Voice and Video Injection Tools
201(2)
8.3.1 Manycam Video Injection Tool
201(1)
8.3.2 Virtual Audio Cable Injection Tool
202(1)
8.4 Lab Scenario
203(2)
8.5 Adding SIP Phones
205(1)
8.6 Configuring Dial Plans
206(1)
8.7 Configuring DAHDI Channels
207(1)
8.8 Starting and Stopping Asterisk
208(1)
8.9 Setup SIP Phone
208(1)
8.10 Making Voice Calls Between SIP Phones
209(2)
8.11 Making Video Calls Between SIP Phones
211(1)
8.12 Making Voice Calls Between SIP and Analogue Phones
211(1)
8.13 Problems
212(3)
9 Case Study 2---VoIP Quality Analysis and Assessment
215(22)
9.1 What Is Wireshark
215(7)
9.1.1 Live Capture and Offline Analysis
215(1)
9.1.2 Three-Pane Packet Browser
216(2)
9.1.3 VoIP Analysis
218(4)
9.2 Wireshark Familiarization
222(1)
9.3 Introduction to Netem and tc Commands
223(2)
9.3.1 Adding qdisc
224(1)
9.3.2 Changing and Deleting qdisc
224(1)
9.4 Lab Scenario
225(1)
9.4.1 Challenges
225(1)
9.5 SIP Registration
226(1)
9.5.1 Challenges
226(1)
9.6 SIP Invite
227(3)
9.6.1 Challenges
227(3)
9.7 VoIP Messages Flow
230(2)
9.7.1 Challenges
230(2)
9.8 VoIP Quality Assessment: Packet Losses
232(1)
9.8.1 Challenges
232(1)
9.9 VoIP Quality Assessment: Delay Variation
233(1)
9.9.1 Challenges
234(1)
9.10 Problems
234(3)
10 Case Study 3---Mobile VoIP Applications and IMS
237(28)
10.1 What Is Open Source IMS Core
237(9)
10.1.1 The Main Features of OSIMS Core P-CSCF
238(1)
10.1.2 The Main Features of OSIMS Core I-CSCF
239(1)
10.1.3 The Main Features of OSIMS Core S-CSCF
240(1)
10.1.4 The Main Features of OSIMS Core FHoSS
241(1)
10.1.5 Installation and Configuration of OSIMS Core
242(4)
10.2 What Is Android
246(7)
10.2.1 Android Smart Phone Market Share
248(1)
10.2.2 Android Architecture
248(1)
10.2.3 The History of Android
249(1)
10.2.4 IMSDroid IMS Client
250(3)
10.3 Lab Scenario
253(7)
10.3.1 Configuring IMSDroid
254(1)
10.3.2 Adding OSIMS Core Subscribers
255(5)
10.4 Making Voice and Video Calls
260(1)
10.4.1 Placing a Call
260(1)
10.4.2 In Call Screen
261(1)
10.5 Problems
261(4)
Index 265
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