SIP Trunking

SIP Trunking

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Product Details

ISBN-13: 9781587059476
Publisher: Pearson Education
Publication date: 02/18/2010
Series: Networking Technology: IP Communications
Sold by: Barnes & Noble
Format: NOOK Book
Pages: 360
File size: 4 MB

About the Author

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).

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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SIP Trunking 3.6 out of 5 based on 0 ratings. 7 reviews.
Boudville More than 1 year ago
What's so great about SIP and why should you [ie. your company] migrate to it? The book explains at various levels the reasons. Succinctly, you should look at Chapter 4, which has an easily understood thread. One main reason is simply to reduce toll charges for long distance, international and local access calls. If you have already informally used VoIP, then going to SIP is essentially a corporate equivalent of moving to it. To some readers, the best reason for SIP is what you can then avoid dealing with your local phone company, which the book characterises as often inflexible and giving poor service. Typically, SIP leads to a more efficient use of your Internet bandwidth. When there are few [incoming or outgoing] calls using SIP, then that "unused" bandwidth is available for general Internet access by your users and by visitors to your website. The book also briefly mentions Power of Ethernet, where a device using this has just one cable that carries both an Internet connection and power. Eliminating the extra power cable can be useful. Unfortunately, the book is marred by a poorly edited first chapter. This chapter was hastily written and not proofread. The ITU is not the Internal [sic] Telecommunications Union. While IEFT in one section is meant to be IETF. Sadly, on page 2, we see nonsense like "200 to 2000 khz", "20 kz to 20,000 hkz" and "20 kz to 20,000 kHz". The chapter will be most readers' initial impression of the book, and this is just sloppy.
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