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SIP Trunks

by: Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan

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Retail Price: $69.95

Publisher: CISCO PRESS,30.04.10

Category: Cisco Level:

ISBN: 1587059444
ISBN13: 9781587059445

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Features and Benefits


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Everything enterprise decision-makers, consultants, and service providers need to know to implement cost-effective, flexible SIP trunking


# Identifies the real benefits of SIP trunking, debunks the myths, and helps enterprises objectively assess the costs and potential ROI
# Shows how to evaluate service provider SIP trunk offerings and structure effective RFPs
# Guides decision-makers in planning a long-term migration to SIP trunking
# Presents network design considerations and implementation best practices



Table of Contents

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  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 Authors

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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).