SIP Trunking
Published on 11. March 2010
Book
Hardback
360 pages
978-1-58705-944-5 (ISBN)
Description
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 (R) 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 (R) 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.
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 (R) 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 (R) 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.
More details
Language
English
Place of publication
Indianapolis
United States
Publishing group
Pearson Education (US)
Target group
Professional and scholarly
Dimensions
Height: 231 mm
Width: 194 mm
Thickness: 27 mm
Weight
740 gr
ISBN-13
978-1-58705-944-5 (9781587059445)
Copyright in bibliographic data and cover images is held by Nielsen Book Services Limited or by the publishers or by their respective licensors: all rights reserved.
Schweitzer Classification
Persons
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).
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).
Content
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
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