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    SIP Trunking

    3.5 7

    by Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan


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    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).

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    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

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    The first complete guide to planning, evaluating, and implementing high-value SIP trunking solutions

     

    Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there’s a key missing link: most businesses still connect to their service providers via old-fashioned, inflexible TDM trunks. Now, three Cisco® experts show how to use Session Initiation Protocol (SIP) trunking to eliminate legacy interconnects and gain the full benefits of end-to-end VoIP.

     

    Written for enterprise decision-makers, network architects, consultants, and service providers, this book demystifies SIP trunking technology and trends and brings unprecedented clarity to the transition from TDM to SIP interconnects. The authors separate the true benefits of SIP trunking from the myths and help you systematically evaluate and compare service provider offerings. You will find detailed cost analyses, including guidance on identifying realistic, achievable savings.

     

    SIP Trunking also introduces essential techniques for optimizing network design and security, introduces proven best practices for implementation, and shows how to apply them through a start-to-finish case study.

     

    • Discover the advanced Unified Communications solutions that SIP trunking facilitates
    • Systematically plan and prepare your network for SIP trunking
    • Generate effective RFPs for SIP trunking
    • Ask service providers the right questions—and make sense of their answers
    • Compare SIP deployment models and assess their tradeoffs
    • Address key network design issues, including security, call admission control, and call flows
    • Manage SIP/TDM interworking throughout the transition

     

    This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.

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