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.

WILLIAM A. FLANAGAN is President and founder of Flanagan Consulting. With three decades of telecommunications experience, Mr. Flanagan is an expert in voice and data technologies, products, markets, and customers. His network designs have solved problems for enterprises, government agencies, and carriers.

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 "The Cloud" / 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 Call/Message 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 VoIP/UC / 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 Scanning/SPIT / 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 LANs/Wi-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 Service/Experience 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 Bridges/Controllers / 248
11.4.3 Call Recorder / 250
11.5 Hosted VoIP/UC Service / 251
11.6 Management Systems/Workstations / 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