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SIP Trunks

by: Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan

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On-line Price: TBAHardcoverpackage, 360

Retail Price: TBA

Publisher: CISCO PRESS,30.04.10

Category: Cisco Level:

ISBN: 1587059444
ISBN13: 9781587059445

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Features and Benefits


Everything enterprise decision-makers, consultants, and service providers need to know to implement cost-effective, flexible SIP trunking

# Identifies the real benefits of SIP trunking, debunks the myths, and helps enterprises objectively assess the costs and potential ROI
# Shows how to evaluate service provider SIP trunk offerings and structure effective RFPs
# Guides decision-makers in planning a long-term migration to SIP trunking
# Presents network design considerations and implementation best practices

Table of Contents


  Introduction xix

Part I: From TDM Trunking to SIP Trunking

Chapter 1 Overview of IP Telephony 1

  History of IP Telephony 1

  Basic Components of IP Telephony 2

          Microphones and Speakers 2

          Digital Signal Processors 3

  Comparing VoIP Signaling Protocols 4

  Call Control Elements of IP Telephony 5

          Other Physical Components of IP Telephony 5

          IP Phones 6

          IP-PBX 6

          Ethernet Switches 6

          Non-IP Phone IP Telephony Devices 6

          WAN Connectivity Device 6

          Voice Gateways 7

          Supplementary Services 9

  Summary 10

Chapter 2 Trends in IP Telephony 11

  Major Trends in IP Communications 12

  Enterprise IP Communications Endpoints 13

          Desktop Handset Trends 15

          Enterprise Softphone IP Phone Trends 16

          Enterprise WiFi IP Phone Trends 17

          Cellular Phone Trends Within Enterprises and Their Effects on SIP Trunking 18

  Endpoint Trends in Enterprises and Their Effects on SIP Trunk 19

  Feature Trends in SIP Trunking Within the Enterprise 20

  Feature Trends in SIP Trunking Between Enterprises 22

  Feature Trends in SIP Trunk for PSTN Access 24

  Feature Trends in Advanced SIP Trunking Features from

  Service Providers 26

  Feature Trends for Call Centers Services from SIP Trunk Providers 28

  Summary 30

Chapter 3 Transitioning to SIP Trunks 31

  Phase I: Assess the Current State of Trunking 33

  Phase II: Determining the Priority of the Project 34

  Phase III: Gather Information from the Local SPs 35

  Phase IV: Conducting a Pilot Implementation of SIP Trunks for PSTN Access 35

  Phase V: Transitioning a Live Department to SIP Trunks 37

  Phase VI: Transition to SIP Trunking for Call Center Locations 38

  Phase VII: Transition to SIP Trunking at Headquarters Locations 39

  Phase VIII: Transition to SIP Trunking of Branch Locations 40

  Phase IX: Transition Any Remaining Trunk to SIP Trunking 41

  Phase X: Post Project Assessment 41

  Summary 43

Chapter 4 Cost Analysis 45

  Capital Costs 46

          Cost of Installation 47

          Cost of Equipment 47

          Border Element Chassis Cost 48

          Port Cost 48

          Digital Signal Processor (DSP) Cost 48

          Software License Cost 49

  Monthly Recurring Costs 49

          Port/Line Charge 49

          Bandwidth Charge 50

          Service Level Agreement Charge 50

  Cost of Usage 51

          Pay as You Use 51

          Bundled Offer 51

          Burstable Shared Trunks 52

          Cost of Spike Calls 53

  Cost of Security 53

  Cost of Expertise/Knowledge 54

  Other Areas of Costs and Savings 54

  Summary 55

  Further Reading 55

Part II: Planning Your Network for SIP Trunking

Chapter 5 Components of SIP Trunks 57

  SP Network Components 57

          SP Network-Edge Session Border Controllers 58

          SP Network-Call Agent 59

          SP Network-Billing Server 61

          SP Network-IP Network Infrastructure 62

          SP Network-Customer Premise Equipment 64

          SP Network-Media Gateways (Voice and Video) 66

          SP Network-Legally Required Supplementary Services Systems/Legal Intercept and Emergency Services 68

          SP Network-Enhanced Services 70

          SP Network-Peering Session Border Controllers 71

          SP Network-Monitoring Equipment 74

  Enterprise Network Components 75

          Enterprise Networks-SP Interconnecting Session Border Controllers 76

          Enterprise Network: IP Network Infrastructure 77

          Enterprise Network-Enterprise Session Management 77

          Enterprise Networks-Application Interconnection Session Border Controller 78

          Enterprise Networks-Intercompany Media Engine 79

  Summary 79

Chapter 6 SIP Trunking Models 81

  Understanding the Traditional PSTN Gateway Connection Model 82

  Choosing a SIP Trunking Model 83

          Types of Calls Carried by the SIP Trunk 83

          Single or Multiple Physical Entry Points 84

          International Call Access 84

          Physical Termination of Traffic into Your Network 84

  Centralized Model 84

  Distributed Model 85

  Hybrid Model 86

  Considering Trade-Offs with the Centralized and Distributed Models 88

          DID Number Portability 88

          Regional or Geographic Boundaries 89

          Regulatory Considerations 90

          Containing Oversubscription 90

          Quality of Service (QoS) Considerations 91

          Bandwidth Provisioning 91

          Latency Implications 91

          Operational and Equipment Implications 92

          Cost 92

          High Availability 93

          Emergency Call Routing 93

          Dial Plan and Call Routing Considerations 94

          IP Addressing 95

  Understanding the Centralized Model with Direct Media Model 96

  Summary 97

Chapter 7 Design and Implementation Considerations 101

  Geographic and Regulatory Considerations 102

  IP Connectivity Options 102

          Physical Delivery and Connectivity 103

          IP Addressing 104

  Dial Plans and Call Routing 104

          Porting Phone Numbers to SIP Trunks 105

          Emergency Calls 105

  Supplementary Services 106

          Voice Calls 106

          Voice Mail 107

          Transcoding 107

          Mobility 108

  Network Demarcation 108

          Service Provider UNI Compliance 109

          Codec Choice 109

          Fault Isolation 110

          Statistics 110

          Billing 111

          QoS Marking 111

  Security Considerations 112

          SIP Trunk Levels of Security Exposure 113

          Access Lists (ACL) 114

          Hostname Validation 115

          NAT and Topology Hiding 116

          Firewalls 116

          Security Protection at the SIP Protocol Level 119

                  SIP Listening Port 120

                  Transport Layer Security (TLS) 120

                  Back-to-Back User Agent (B2BUA) 121

                  SIP Normalization 121

                  Digit Manipulation 122

                  SIP Privacy Methods 122

          Registration and Authentication 122

          Toll Fraud 123

          Signaling and Media Encryption 124

  Session Management, Call Traffic Capacity, Bandwidth

          Control, and QoS 124

          Trunk Provisioning 125

          Bandwidth Adjustments and Consumption 125

          Call Admission Control (CAC) 125

                  Limiting Calls per Dial-Peer 126

                  Global Call Admission Control 126

          Quality of Service (QoS) 127

                  Traffic Marking 127

                  Delay and Jitter 128

                  Echo 128

                  Congestion Management 128

          Voice-Quality Monitoring 129

  Scalability and High Availability 130

Local and Geographical SIP Trunk Redundancy 131

          Border Element Redundancy 132

                  In-Box Hardware Redundancy 132

                  Box-to-Box Hardware Redundancy (1+1) 132

                  Clustering (N+1) 133

          Load Balancing 133

                  Service Provider Load Balancing 134

                  Domain Name System (DNS) 134

                  CUCM Route Groups and Route Lists 135

                  Cisco Unified SIP Proxy 135

          PSTN TDM Gateway Failover 136

  SIP Trunk Capacity Engineering 137

  SIP Trunk Monitoring 138

  Summary 139

  Further Reading 139

Chapter 8 Interworking 141

  Protocols 142

          Applications 142

          Endpoints 143

          Service Provider SIP Trunk Interworking-SP UNI 143

          SIP Normalization 145

  Media 148

          DTMF 148

                  DTMF Relay 148

                  DTMF Relay Methods 149

                  DTMF Relay Conversion 150

          Codecs 150

                  Payload Types 151

                  Codec Filtering or Stripping 152

                  Transcoding 153

                  Transrating 154

          Fax and Modem Traffic 155

                  T.38 as a Fax Method for SIP Trunks 155

                  Fax Pass-Through as a Fax Method for SIP Trunks 155

                  Modem Traffic 155

  Encryption Interworking 156

  Summary 158

  Further Reading 158

Chapter 9 Questions to Ask of a Service Provider Offering and an SBC Vendor 161

  Technical Requirements 161

          Session Management 162

                  Signaling/Media Protocol 162

                  Operational Modes Support 162

                  SIP Features 163

                  SIP Methods 166

                  IETF and General SIP Support 167

                  Session Timers 168

                  Quality of Service 168

          Interworking Support 169

                  Codecs Support 169

                  SIP to H.323 Interworking Support 170

                  Other Interworking Support 171

          Demarcation 171

                  Topology Hiding 171

                  NAT Traversal 172

                  Session Routing 172

                  Accounting and Billing 172

          Security 173

                  Privacy 173

                  Firewall Integration 174

                  Threat Protection 174

                  Policy 174

                  Access Control 175

          Operations and Management 175

                  Event/Alarm Management 176

                  Configuration Management 176

                  Performance Management 176

                  Security Management 176

                  Fault Management 176

                  Other Questions about Operations and Management 177

          System Specification 178

          Performance/Sizing 178

                  Availability 179

                  Load Balancing 179

                  Performance 180

  Delivery, Documentation, and Support 180

  Delivery 181

          Documentation and Training 182

          Support 182

  Quality 183

          Quality Assurance 184

          Certification 185

  Business 185

          Bidder Background 186

          Bidder References 188

  Cost 188

  Summary 189

  Further Reading 189

Part III: Deploying SIP Trunks

Chapter 10 Deployment Scenarios 191

  Enterprise SIP Trunk for PSTN Access 191

          Cisco UCM SIP to an AT&T FlexReach SIP Trunk 192

          CUCM to a Verizon SIP Trunk 197

          Cisco UCM H.323 Interconnect 202

          Sharing a SIP Trunk Across the Enterprise 204

          Contact Center SIP Trunk Interconnect 206

  SMB SIP Trunk for PSTN Access 212

  Additional Deployment Variations 223

          CUBE with SRST 224

          CUBE Transcoding 225

          CUBE with Integrated Cisco IOS Firewall 227

          CUBE with Tcl Scripting 229

          CUBE Using SIP TLS to CUCM 232

          Telepresence Business-to-Business Interconnect 233

          Miscellaneous Helpful Configurations 235

                  Collocated MTP 236

                  SIP IP Address Bind 236

                  SIP Out-of-Dialog OPTIONS Ping 237

                  Multiple Codecs Outbound from CUCM on a SIP Trunk 237

                  SIP Header Manipulation 238

                  Dual Digit Drop 239

                  SIP Registration 239

                  SIP Transport Choices 239

                  QoS Remarking 240

                  SIP User Agent Parameters 240

  Troubleshooting 240

  Summary 241

  Further Reading 241

Chapter 11 Deployment Steps and Best Practices 243

  Deployment Steps 244

          Planning 244

                  Cost Analysis 245

                  Assess Traffic Volumes and Patterns 245

                  Assess Network Design Implications 246

                  Emergency Call Policy 246

                  Define Production User Community Phases 246

                  Define the User Community to Pilot 247

                  Evaluate Future New Services 247

                  Assess Security Implications 248

          Evaluating a SIP Trunk Offering 248

                  Assess SIP Trunk Provider Offerings 249

                  Determine the Availability of TDM-Equivalent Features 249

                  Determine Geographic Coverage 249

                  Assess DID Porting Realities 249

                  Determine Call Load Balancing and Failover Routing 251

                  Determine Emergency Call Handling 251

                  Determine the Physical Delivery of the SIP Trunk 251

                  Determine Network Demarcation 252

          Agree on Monitoring and Troubleshooting Procedures 252

          Pilot Trial 252

                  Define Clear Success Criteria 253

                  Assess Organizational Responsibility 253

                  Determine the Length of the Trial 253

                  Install and Configure the Service 254

                  Define a Clear Test Plan and Execute the Test Plan 254

                  Start Using the SIP Trunk for the Pilot User Community 255

          Production Service 256

  Best Practices 256

          Providers 256

          Deployment 257

          Network Design 257

          Protocols and Codecs 258

          Cisco Unified Communications Manager (CUCM) 259

          SBC Best Practices 260

          Security 261

          Redundancy 261

  Summary 262

Chapter 12 Case Studies 263

  Enterprise Connecting to a Service Provider 263

          Creating Different Route Groups 267

          MTP Configuration 267

          Interconnect Between H.323 and SIP 270

          DTMF Interworking 271

          Dial-Peer Configurations Example 272

          Call Admission Control 274

  Distributed SIP Trunking to Connect PSTN 274

          Enterprise Architecture 275

          Bank Requirements 276

          SP Requirements 277

          Configurations 277

                  CUCM Configuration 277

                  CUBE Configuration 290

  Summary 295

Chapter 13 Future of Unified Communications 297

  Meaning of UC 298

  Components of UC 298

  UC Today 299

  UC Is Anytime, Anyplace, Anywhere 300

  Mobility Provides Access Anytime 301

  Telepresence: the Future of Presence 302

  UC in Healthcare 303

  Journey Ahead 304

          Longer-Term Technological Changes 304

          IPv6 and Its Effect on the Future of UC 307

          The Power of Revolution: The Greening of Unified

          Communications 308

  Summary 308

Index 311

9781587059445, TOC, 1/28/10

About the Authors


Christina Hattingh is a member of the technical staff in the Access Routing Technology Group (ARTG) of Cisco. The ARTG router product portfolio, including the Cisco 2800, 3800, 2900, and 3900 Series integrated services routers and their predecessors, was one of the first Cisco platforms to converge voice and data starting in the late 1990s by offering TDM voice interfaces, WAN interfaces, and critical QoS features. Over time sophisticated call control and routing elements were integrated into the router-based platform making stand-alone VoIP deployments and wide inter-vendor VoIP network interoperability possible. In this role, Christina trains Cisco sales staff and customers and consults widely on voice network deployment and design. She is a long-time speaker of the Cisco Networkers and CiscoLive conferences. Christina holds a graduate degree in mathematical statistics and computer science from the University of Pretoria, South Africa.

Darryl Sladden is a product manager at Cisco and has been with Cisco for more than ten years. Currently, Darryl is a member of the ARTG at Cisco. The ARTG responsibilities include the Cisco ISR and ISR G2, AS5000, and the Cisco Unified Border Element (CUBE). Darryl has been a key contributor to the AS5000 product, CUBE, and several other VoIP technologies at Cisco for several years. The CUBE and the AS5000 product lines are widely used by service providers and enterprise customers as border elements between SIP, H.323, and TDM networks. Darryl has worked with many service provider and enterprise customers who use CUBE to implement SIP Trunks into both Cisco Unified Communications Manager (CUCM) and Cisco Unified Communications Manager Express (CUCME) solutions. Darryl has a degree in mathematics from the University of Waterloo and holds a patent in the use of voice-based network management, and several other patents are under consideration.

ATM Zakaria Swapan is a member of the technical staff in the ARTG at Cisco. The ARTG responsibilities include the Cisco 2800, 3800, 2900, and 3900 Series integrated services routers and the CUBE. ATM has been a key contributor to SIP, Secure Unified Communications, Wireless Voice, Network Intelligence, Network Virtualization, RSVP, and many other developments. ATM has also worked with service providers and enterprise customers who use CUBE to implement SIP Trunks into both CUCM and CUCME solutions. ATM holds an M.S. degree in computer science from Texas A&M University and a B.S. degree in computer science and engineering from Bangladesh University of Engineering and Technology (BUET).