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VoIP and unified communications : internet telephony and the future voice network / William A. Flanagan.

By: Flanagan, William A [author.].
Contributor(s): IEEE Xplore (Online Service) [distributor.] | Wiley [publisher.].
Material type: materialTypeLabelBookPublisher: Hoboken, New Jersey : Wiley, c2012Distributor: [Piscataqay, New Jersey] : IEEE Xplore, [2012]Description: 1 PDF (xv, 298 pages) : illustrations.Content type: text Media type: electronic Carrier type: online resourceISBN: 9781118166048.Subject(s): Internet telephonyGenre/Form: Electronic books.Additional physical formats: Print version:: No titleDDC classification: 004.69/5 Online resources: Abstract with links to resource Also available in print.
Contents:
-- Preface xiii -- Acknowledgments xv -- 1 IP Technology Disrupts Voice Telephony 1 -- 1.1 Introduction to the Public Switched Telephone Network 1 -- 1.2 The Digital PSTN 2 -- 1.3 The Packet Revolution in Telephony 8 -- 1.3.1 Summary of Packet Switching 9 -- 1.3.2 Link Capacity: TDM versus Packets 11 -- 1.3.3 VoIP and (3z(BThe Cloud(3y(B 13 -- IN SHORT: Reading Network Drawings 14 -- 2 Traditional Telephones Still Set Expectations 17 -- 2.1 Availability: How the Bell System Ensured Service 18 -- 2.2 Call Completion 19 -- 2.3 Sound Quality: Encoding for Recognizable Voices 20 -- 2.4 Low Latency 23 -- 2.5 Call Setup Delays 24 -- 2.6 Impairments Controlled: Echo, Singing, Distortion, Noise 25 -- 3 From Circuits to Packets 27 -- 3.1 Data and Signaling Preceded Voice 27 -- 3.1.1 X.25 Packet Data Service 27 -- 3.1.2 SS7: PSTN Signaling on Packets 28 -- 3.1.3 ISDN 29 -- 3.2 Putting Voice into Packets 30 -- 3.2.1 Voice Encoding 31 -- 3.2.2 Dicing and Splicing Voice Streams 32 -- 3.2.3 The Latency Budget 33 -- 4 Packet Transmission and Switching 37 -- 4.1 The Physical Layer: Transmission 39 -- IN SHORT: The Endian Wars 40 -- 4.2 Data Link Protocols 41 -- 4.3 IP, the Network Protocol 43 -- 4.4 Layer 4 Transport Protocols 47 -- 4.4.1 Transmission Control Protocol 47 -- 4.4.2 User Datagram Protocol 50 -- 4.4.3 Stream Control Transmission Protocol 51 -- 4.5 Higher Layer Processes 54 -- 4.5.1 RTP 54 -- 4.5.2 RTCP 57 -- 4.5.3 Multiplexing RTP and RTCP on One UDP Port 58 -- 4.5.4 RTP Mixers and Translators 59 -- 4.5.5 Layered Encoding 60 -- 4.5.6 Profiles for Audio and Video Conferences 60 -- 4.5.7 Security via Encryption 61 -- IN SHORT: Public Key Infrastructure (PKI) 62 -- 4.6 Saving Bandwidth 64 -- 4.6.1 Voice Compression 64 -- 4.6.2 Header Compression 66 -- 4.6.3 Silence Suppression, VAD 67 -- 4.6.4 Sub-Packet Multiplexing 69 -- 4.6.5 Protocol and Codec Selection 70 -- 4.7 Differences: Circuit versus Packet Switched 71 -- 4.7.1 Power to the Desktop Phone 71 -- 4.7.2 Phone as Computer and Computer as Phone 72.
4.7.3 Length of a Phone Line 72 -- 4.7.4 Scaling to Large Size 75 -- 4.7.5 Software Ownership and Licenses 75 -- 5 VoIP Signaling and Call Processing 77 -- 5.1 What Packet Voice and UC Systems Share 78 -- 5.2 Session Initiation Protocol (SIP) 80 -- 5.2.1 SIP Architecture 81 -- 5.2.2 SIP Messages 88 -- 5.2.3 SIP Header Fields and Behaviors 94 -- 5.3 Session Description Protocol 101 -- IN SHORT: ABNF 104 -- 5.4 Media Gateway Control Protocol 107 -- 5.4.1 MGW Functions 107 -- 5.4.2 MGW Connection Model 110 -- 5.4.3 Megaco Procedures 112 -- 5.4.4 Megaco Details 115 -- 5.4.5 Signaling Conversion 119 -- 5.4.6 Voice Transcoding 119 -- 5.5 H.323 120 -- 5.5.1 H.323 Architecture 121 -- 5.5.2 Gatekeeper 123 -- 5.5.3 Gateway 126 -- 5.5.4 Terminal 126 -- 5.5.5 Multipoint Control Unit 127 -- 5.5.6 Call Procedures 128 -- 5.6 Directory Services 134 -- 5.6.1 Domain Name Service (DNS) 134 -- 5.6.2 ENUM 135 -- 6 VoIP and Unified Communications Define the Future 139 -- 6.1 Voice as Before, with Additions 139 -- 6.2 Legacy Services to Keep and Improve with VoIP 140 -- 6.2.1 Flexible Call Routing and 800 Numbers 141 -- 6.2.2 Call on Hold 141 -- 6.2.3 Call Transfer 142 -- 6.2.4 Call Forwarding 142 -- 6.2.5 Audio Conferencing 142 -- 6.2.6 Video Conferencing 143 -- 6.2.7 Local Number Portability 144 -- 6.2.8 Direct Inward Dialing, Dialed Number Indication 144 -- 6.2.9 CallMessage Waiting 145 -- 6.2.10 Call Recording 146 -- 6.2.11 Emergency Calling (E911) 146 -- 6.2.12 Tracking IP Phone Locations for E911 150 -- 6.3 Facsimile Transmission 153 -- 6.3.1 Facsimile on the PSTN 153 -- 6.3.2 Real-Time Fax over IP: Fax Relay or T.38 155 -- 6.3.3 Store-and-Forward Fax Handling 160 -- 6.3.4 IP Faxing over the PSTN 161 -- 6.4 Phone Features Added with VoIPUC 162 -- 6.4.1 Presence 163 -- 6.4.2 Forking 163 -- 6.4.3 Voicemail1/4eMail 163 -- 6.4.4 SMS Integration 164 -- 6.4.5 Instant Messaging 165 -- 6.4.6 Webinar Broadcasts 168 -- 6.4.7 Telepresence 168 -- 6.4.8 More UC Features to Consider 168 -- 7 How VoIP and UC Impact the Network 171.
7.1 Space, Power, and Cooling 171 -- 7.2 Priority for Voice, Video, Fax Packets 172 -- 7.3 Packets per Second 174 -- 7.4 Bandwidth 174 -- 7.5 Security Issues 175 -- 7.5.1 Eavesdropping and vLAN Hopping 176 -- 7.5.2 Access Controls for Users and Connections 176 -- 7.5.3 Modems 177 -- 7.5.4 DNS Cache Poisoning 177 -- IN SHORT: Earliest Instance of DNS Cache Poisoning 179 -- 7.5.5 Toll Fraud 179 -- 7.5.6 Pay-per-Call Scams 179 -- 7.5.7 Vishing 180 -- 7.5.8 SIP ScanningSPIT 180 -- 7.5.9 Opening the Firewall to Incoming Voice 181 -- 7.6 First Migration Steps While Keeping Legacy Equipment 181 -- 7.6.1 Circuit-Switched PBX 182 -- 7.6.2 Digital Phones 182 -- 7.6.3 Analog Phones and FX Service 183 -- 7.6.4 Facsimile Machines 184 -- 7.6.5 Modems 185 -- 8 Interconnections to Global Services 187 -- 8.1 Media Gateways 188 -- 8.2 SIP Trunking 192 -- 8.3 Operating VoIP Across Network Address Translation 196 -- 8.3.1 Failures of SIP, SDP (Signaling) 199 -- 8.3.2 Failures of RTP (Media) 199 -- 8.3.3 Solutions 200 -- 8.3.4 STUN: Session Traversal Utilities for NAT 201 -- 8.3.5 TURN: Traversal Using Relays around NAT 204 -- 8.3.6 ICE: Interactive Connectivity Establishment 206 -- 8.4 Session Border Controller 207 -- 8.4.1 Enterprise SBC 209 -- 8.4.2 Carrier SBC 210 -- 8.5 Supporting Multiple-Carrier Connections 212 -- 8.6 Mobility and Wireless Access 213 -- 8.6.1 VoIP on Wireless LANsWi-Fi 213 -- 8.6.2 Integration of Wi-Fi and Cellular Services 214 -- 8.6.3 Packet Voice on Mobile Broadband: WiMAX, LTE 214 -- 8.6.4 Radio over VoIP 215 -- IN SHORT: E&M Voice Signaling 216 -- 9 Network Management for VoIP and UC 217 -- 9.1 Starting Right 218 -- 9.1.1 Acceptance Testing 219 -- 9.1.2 Configuration Management and Governance 220 -- 9.1.3 Privilege Setting 220 -- 9.2 Continuous Monitoring and Management 221 -- 9.2.1 NMS Software 222 -- 9.2.2 Simple Network Management Protocol 223 -- 9.2.3 Web Interface 224 -- 9.2.4 Server Logging 224 -- 9.2.5 Software Maintenance 225 -- 9.2.6 Quality of ServiceExperience Monitoring 225.
9.2.7 Validate Adjustments and Optimization 226 -- 9.3 Troubleshooting and Repair 226 -- 9.3.1 Methods 226 -- 9.3.2 Software Tools 228 -- 9.3.3 Test Instruments 229 -- 10 Cost Analysis and Payback Calculation 231 -- 11 Examples of Hardware and Software 237 -- 11.1 IP Phones 237 -- 11.2 Gateways 240 -- 11.3 Session Border Controllers 242 -- 11.4 Call-Switching Servers 244 -- 11.4.1 IP PBX 246 -- 11.4.2 Conference BridgesControllers 248 -- 11.4.3 Call Recorder 250 -- 11.5 Hosted VoIPUC Service 251 -- 11.6 Management SystemsWorkstations 252 -- 12 Appendixes 253 -- 12.1 Acronyms and Definitions 253 -- 12.2 Reference Documents 268 -- 12.2.1 RFCs 268 -- 12.2.2 ITU Recommendations 272 -- 12.2.3 Other Sources 272 -- 12.3 Message and Error Codes 274 -- Index 277.
Summary: "The new generation of voice services and telephony will be based on packet networks rather than TDM transmission and switching. This book addresses the evolution of telephony to Voice over IP (VoIP) and Unified Communications (UC), bringing email, voice mail, fax, and telephone services to one user interface. Concise and to the point, this text tells readers what they need to know to deal with vendors, network engineers, data center gurus, and top management with the confidence and clear understanding of how things really work. It serves as a useful tool for engineers just entering the field, as well as for experienced engineers and technical managers who want to deal effectively with sales people"-- Provided by publisher.
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Includes index.

-- Preface xiii -- Acknowledgments xv -- 1 IP Technology Disrupts Voice Telephony 1 -- 1.1 Introduction to the Public Switched Telephone Network 1 -- 1.2 The Digital PSTN 2 -- 1.3 The Packet Revolution in Telephony 8 -- 1.3.1 Summary of Packet Switching 9 -- 1.3.2 Link Capacity: TDM versus Packets 11 -- 1.3.3 VoIP and (3z(BThe Cloud(3y(B 13 -- IN SHORT: Reading Network Drawings 14 -- 2 Traditional Telephones Still Set Expectations 17 -- 2.1 Availability: How the Bell System Ensured Service 18 -- 2.2 Call Completion 19 -- 2.3 Sound Quality: Encoding for Recognizable Voices 20 -- 2.4 Low Latency 23 -- 2.5 Call Setup Delays 24 -- 2.6 Impairments Controlled: Echo, Singing, Distortion, Noise 25 -- 3 From Circuits to Packets 27 -- 3.1 Data and Signaling Preceded Voice 27 -- 3.1.1 X.25 Packet Data Service 27 -- 3.1.2 SS7: PSTN Signaling on Packets 28 -- 3.1.3 ISDN 29 -- 3.2 Putting Voice into Packets 30 -- 3.2.1 Voice Encoding 31 -- 3.2.2 Dicing and Splicing Voice Streams 32 -- 3.2.3 The Latency Budget 33 -- 4 Packet Transmission and Switching 37 -- 4.1 The Physical Layer: Transmission 39 -- IN SHORT: The Endian Wars 40 -- 4.2 Data Link Protocols 41 -- 4.3 IP, the Network Protocol 43 -- 4.4 Layer 4 Transport Protocols 47 -- 4.4.1 Transmission Control Protocol 47 -- 4.4.2 User Datagram Protocol 50 -- 4.4.3 Stream Control Transmission Protocol 51 -- 4.5 Higher Layer Processes 54 -- 4.5.1 RTP 54 -- 4.5.2 RTCP 57 -- 4.5.3 Multiplexing RTP and RTCP on One UDP Port 58 -- 4.5.4 RTP Mixers and Translators 59 -- 4.5.5 Layered Encoding 60 -- 4.5.6 Profiles for Audio and Video Conferences 60 -- 4.5.7 Security via Encryption 61 -- IN SHORT: Public Key Infrastructure (PKI) 62 -- 4.6 Saving Bandwidth 64 -- 4.6.1 Voice Compression 64 -- 4.6.2 Header Compression 66 -- 4.6.3 Silence Suppression, VAD 67 -- 4.6.4 Sub-Packet Multiplexing 69 -- 4.6.5 Protocol and Codec Selection 70 -- 4.7 Differences: Circuit versus Packet Switched 71 -- 4.7.1 Power to the Desktop Phone 71 -- 4.7.2 Phone as Computer and Computer as Phone 72.

4.7.3 Length of a Phone Line 72 -- 4.7.4 Scaling to Large Size 75 -- 4.7.5 Software Ownership and Licenses 75 -- 5 VoIP Signaling and Call Processing 77 -- 5.1 What Packet Voice and UC Systems Share 78 -- 5.2 Session Initiation Protocol (SIP) 80 -- 5.2.1 SIP Architecture 81 -- 5.2.2 SIP Messages 88 -- 5.2.3 SIP Header Fields and Behaviors 94 -- 5.3 Session Description Protocol 101 -- IN SHORT: ABNF 104 -- 5.4 Media Gateway Control Protocol 107 -- 5.4.1 MGW Functions 107 -- 5.4.2 MGW Connection Model 110 -- 5.4.3 Megaco Procedures 112 -- 5.4.4 Megaco Details 115 -- 5.4.5 Signaling Conversion 119 -- 5.4.6 Voice Transcoding 119 -- 5.5 H.323 120 -- 5.5.1 H.323 Architecture 121 -- 5.5.2 Gatekeeper 123 -- 5.5.3 Gateway 126 -- 5.5.4 Terminal 126 -- 5.5.5 Multipoint Control Unit 127 -- 5.5.6 Call Procedures 128 -- 5.6 Directory Services 134 -- 5.6.1 Domain Name Service (DNS) 134 -- 5.6.2 ENUM 135 -- 6 VoIP and Unified Communications Define the Future 139 -- 6.1 Voice as Before, with Additions 139 -- 6.2 Legacy Services to Keep and Improve with VoIP 140 -- 6.2.1 Flexible Call Routing and 800 Numbers 141 -- 6.2.2 Call on Hold 141 -- 6.2.3 Call Transfer 142 -- 6.2.4 Call Forwarding 142 -- 6.2.5 Audio Conferencing 142 -- 6.2.6 Video Conferencing 143 -- 6.2.7 Local Number Portability 144 -- 6.2.8 Direct Inward Dialing, Dialed Number Indication 144 -- 6.2.9 CallMessage Waiting 145 -- 6.2.10 Call Recording 146 -- 6.2.11 Emergency Calling (E911) 146 -- 6.2.12 Tracking IP Phone Locations for E911 150 -- 6.3 Facsimile Transmission 153 -- 6.3.1 Facsimile on the PSTN 153 -- 6.3.2 Real-Time Fax over IP: Fax Relay or T.38 155 -- 6.3.3 Store-and-Forward Fax Handling 160 -- 6.3.4 IP Faxing over the PSTN 161 -- 6.4 Phone Features Added with VoIPUC 162 -- 6.4.1 Presence 163 -- 6.4.2 Forking 163 -- 6.4.3 Voicemail1/4eMail 163 -- 6.4.4 SMS Integration 164 -- 6.4.5 Instant Messaging 165 -- 6.4.6 Webinar Broadcasts 168 -- 6.4.7 Telepresence 168 -- 6.4.8 More UC Features to Consider 168 -- 7 How VoIP and UC Impact the Network 171.

7.1 Space, Power, and Cooling 171 -- 7.2 Priority for Voice, Video, Fax Packets 172 -- 7.3 Packets per Second 174 -- 7.4 Bandwidth 174 -- 7.5 Security Issues 175 -- 7.5.1 Eavesdropping and vLAN Hopping 176 -- 7.5.2 Access Controls for Users and Connections 176 -- 7.5.3 Modems 177 -- 7.5.4 DNS Cache Poisoning 177 -- IN SHORT: Earliest Instance of DNS Cache Poisoning 179 -- 7.5.5 Toll Fraud 179 -- 7.5.6 Pay-per-Call Scams 179 -- 7.5.7 Vishing 180 -- 7.5.8 SIP ScanningSPIT 180 -- 7.5.9 Opening the Firewall to Incoming Voice 181 -- 7.6 First Migration Steps While Keeping Legacy Equipment 181 -- 7.6.1 Circuit-Switched PBX 182 -- 7.6.2 Digital Phones 182 -- 7.6.3 Analog Phones and FX Service 183 -- 7.6.4 Facsimile Machines 184 -- 7.6.5 Modems 185 -- 8 Interconnections to Global Services 187 -- 8.1 Media Gateways 188 -- 8.2 SIP Trunking 192 -- 8.3 Operating VoIP Across Network Address Translation 196 -- 8.3.1 Failures of SIP, SDP (Signaling) 199 -- 8.3.2 Failures of RTP (Media) 199 -- 8.3.3 Solutions 200 -- 8.3.4 STUN: Session Traversal Utilities for NAT 201 -- 8.3.5 TURN: Traversal Using Relays around NAT 204 -- 8.3.6 ICE: Interactive Connectivity Establishment 206 -- 8.4 Session Border Controller 207 -- 8.4.1 Enterprise SBC 209 -- 8.4.2 Carrier SBC 210 -- 8.5 Supporting Multiple-Carrier Connections 212 -- 8.6 Mobility and Wireless Access 213 -- 8.6.1 VoIP on Wireless LANsWi-Fi 213 -- 8.6.2 Integration of Wi-Fi and Cellular Services 214 -- 8.6.3 Packet Voice on Mobile Broadband: WiMAX, LTE 214 -- 8.6.4 Radio over VoIP 215 -- IN SHORT: E&M Voice Signaling 216 -- 9 Network Management for VoIP and UC 217 -- 9.1 Starting Right 218 -- 9.1.1 Acceptance Testing 219 -- 9.1.2 Configuration Management and Governance 220 -- 9.1.3 Privilege Setting 220 -- 9.2 Continuous Monitoring and Management 221 -- 9.2.1 NMS Software 222 -- 9.2.2 Simple Network Management Protocol 223 -- 9.2.3 Web Interface 224 -- 9.2.4 Server Logging 224 -- 9.2.5 Software Maintenance 225 -- 9.2.6 Quality of ServiceExperience Monitoring 225.

9.2.7 Validate Adjustments and Optimization 226 -- 9.3 Troubleshooting and Repair 226 -- 9.3.1 Methods 226 -- 9.3.2 Software Tools 228 -- 9.3.3 Test Instruments 229 -- 10 Cost Analysis and Payback Calculation 231 -- 11 Examples of Hardware and Software 237 -- 11.1 IP Phones 237 -- 11.2 Gateways 240 -- 11.3 Session Border Controllers 242 -- 11.4 Call-Switching Servers 244 -- 11.4.1 IP PBX 246 -- 11.4.2 Conference BridgesControllers 248 -- 11.4.3 Call Recorder 250 -- 11.5 Hosted VoIPUC Service 251 -- 11.6 Management SystemsWorkstations 252 -- 12 Appendixes 253 -- 12.1 Acronyms and Definitions 253 -- 12.2 Reference Documents 268 -- 12.2.1 RFCs 268 -- 12.2.2 ITU Recommendations 272 -- 12.2.3 Other Sources 272 -- 12.3 Message and Error Codes 274 -- Index 277.

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"The new generation of voice services and telephony will be based on packet networks rather than TDM transmission and switching. This book addresses the evolution of telephony to Voice over IP (VoIP) and Unified Communications (UC), bringing email, voice mail, fax, and telephone services to one user interface. Concise and to the point, this text tells readers what they need to know to deal with vendors, network engineers, data center gurus, and top management with the confidence and clear understanding of how things really work. It serves as a useful tool for engineers just entering the field, as well as for experienced engineers and technical managers who want to deal effectively with sales people"-- Provided by publisher.

Also available in print.

Mode of access: World Wide Web

Description based on PDF viewed 10/24/2017.

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