Pub. Date:
Cisco Press
SIP Trunking

SIP Trunking

by Christina Hattingh
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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.

Christina Hattingh, member of the technical staff in the Cisco Access Routing Technology Group (ARTG), has been involved with Cisco VoIP technologies from their inception and continues to consult and deliver training in these areas. Darryl Sladden, a Cisco Senior Product Manager, has been a key architect of the Cisco Unified Border Element and the Cisco SIP Trunking strategy as well as a key contributor to the AS5000 product, and several other Cisco VoIP technologies. ATM Zakaria Swapan, Cisco ARTG member of technical staff, has been a key contributor to the Cisco SIP development, Cisco Unified Border Element, VoIP Gateway, Secure Unified Communications, Wireless Voice, QoS & Call Admission Control and several other VoIP technologies.

• 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.

Product Details

ISBN-13: 9781587059445
Publisher: Cisco Press
Publication date: 03/04/2010
Series: Networking Technology: IP Communications Series
Pages: 324
Product dimensions: 7.60(w) x 9.20(h) x 1.00(d)

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


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

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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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