Sip Trunking

Sip Trunking

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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(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.show more

Product details

  • Paperback | 360 pages
  • 187 x 232 x 22.86mm | 590g
  • Pearson Education (US)
  • Cisco Press
  • Indianapolis, United States
  • English
  • 1587144417
  • 9781587144417

About Christina Hattingh

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).show more

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, 1/28/10show more