SIP Trunking

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.  

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, 1/28/10  

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).