How This Book Is Organized
This book is designed to be flexible and allow you to easily move between chapters to focus on topics of interest to you. However, if you do intend to read every chapter, the order in the book is an excellent sequence to use because it progresses from basic to more advanced subjects. Chapter 1 provides an overview of the components of an IP voice network. The remainder of the book is divided into three parts: “Gateways.” “Gatekeepers,” and “IP-to-IP Gateways.”
Part I, “Voice Gateways and Gatekeepers”
Chapter 1, “Gateways and Gatekeepers”—This chapter provides an overview of components of an IP voice network, including different types of gateways, such as routers, standalone devices, and switch modules, H.323 gatekeepers, and IP-to-IP gateways (IPIPGW). It contains a review of the Multiple Gateway Control Protocol (MGCP), H.323, Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP), and Real-Time Transport Protocol (RTP) protocols. It also describes different types of call agents that are used in IP voice networks. In addition, it introduces the network used throughout the book in examples and case studies.
Chapter 2, “Media Gateway Control Protocol”—This chapter covers how MGCP implements call signaling, describes call flow using MGCP, and discusses the pros and cons of the protocol and when to use it. It then covers implementing MGCP gateways—configuration of the router, addition of an MGCP gateway to CallManager, configuration of dual tone multifrequency (DTMF) relay, some security features, and troubleshooting of MGCP gateways.
Chapter 3, “H.323”—This chapter discusses the implemention of call signaling with the H.323 protocol, some of the protocols that are part of the H.323 suite, call flow using H.323, and the pros and cons of the protocol and when to use it. It then covers implementing H.323 gateways—configuring the router, using toll bypass, adding a gateway to CallManager, configuring DTMF relay, allowing H.323-to-H.323 connections, configuring both H.323 and MGCP on a gateway, using H.323 security features, and troubleshooting.
Chapter 4, “Session Initiation Protocol”—This chapter describes how SIP implements call signaling, describes the various SIP functions and how they participate in the call flow, and lists the pros and cons of the protocol and when to use it. It covers implementing SIP gateways—configuring the router, using SIP toll bypass, adding the gateway to CallManager, using SIP security features, allowing SIP-to-H.323 connections, and troubleshooting SIP gateways.
Chapter 5, “Circuit Options”—This chapter examines the various analog and digital circuits that gateways use to connect a VoIP network to the public switched telephone network (PSTN). It looks at the different signaling types, features, and uses of each type of circuit; issues such as echo cancellation; and selection and sizing of the circuit.
Chapter 6, “Connecting to the PSTN”—PSTN connections are examined in more depth in this chapter. The analog coverage includes a discussion on configuring and troubleshooting Foreign Exchange Office (FXO), direct inward dial (DID), and Centralized Automated Message Accounting (CAMA). The digital coverage includes explanations on configuring and troubleshooting T1 and E1 PRI, BRI, and channel-associated signaling (CAS) connections. The chapter also discusses caller ID and name delivery considerations.
Chapter 7, “Connecting to PBXs”—This chapter covers the configuration of analog PBX trunks using Ear and Mouth (E&M) and FXO interfaces, and digital PBX trunks using CAS, PRI, and BRI interfaces. It discusses Transparent Common Channel Signaling (T-CCS) and Q Signaling (QSIG).
Chapter 8, “Connecting to an IP WAN”—This chapter covers reasons for connecting a VoIP network to an IP WAN, QoS considerations, use of the modular quality of service command-line interface (MQC) to provide the needed quality of service, and QoS over an MPLS network. Handling faxes and modems in a VoIP network are discussed. This chapter also describes the need for securing voice traffic over a WAN by using SRTP and V3PN, the implications of NAT on voice traffic, and the use of firewalls with VoIP.
Chapter 9, “Dial Plans”—This chapter covers dial plan design and implementation. It discusses building a scalable dial plan, handling overlapping number ranges, and implementing a dial plan on gateways and call agents. It helps you understand the different types of dial peers and how they affect call routing by examining the way that the gateway selects dial peer matches.
Chapter 10, “Digit Manipulation”—This chapter discusses various ways for a gateway to control called and calling phone numbers, including digit stripping, forwarding only a specified number of digits, prefixing digits, and expanding numbers. It also covers voice translation rules, use of regular expressions, voice translation profiles, control of the calling-line identification information, and verification and troubleshooting of digit manipulation.
Chapter 11, “Influencing Path Selection”—This chapter covers several different ways of influencing route selection, including hunt groups, trunk groups, and tail end hop-off. The chapter examines call admission control (CAC) techniques—local CAC using maximum connections and local voice busyout; measurement-based CAC using IP SLA, PSTN fallback, and advanced local voice busyout; and resource-based CAC using Resource Reservation Protocol (RSVP) and RSVP agent, and using CAC with gatekeepers. It also discusses considerations when using POTS-to-POTS call routing.
Chapter 12, “Configuring Class of Restrictions”—This chapter explains what COR is and how it operates. It covers using COR with CallManager Express (CME) implementations, and on a gateway with Survivable Remote Site Telephony (SRST), restricting both inbound and outbound calls.
Chapter 13, “SRST and MGCP Gateway Fallback”—This chapter gives an overview of SRST with ways to adjust the failover and fallback times, and shows how to configure SRST on both the gateway and CallManager. It describes MGCP fallback and how to configure it. SRST dial plan considerations are discussed, along with configuring SRST call features, configuring SIP SRST, preserving calls, and troubleshooting SRST and MGCP fallback.
Chapter 14, “DSP Resources”—This chapter examines codec considerations and configuration, digital signal processor (DSP) considerations, determination of DSP resources needed, the DSP modes, and configuration of conferencing and transcoding resources on both the gateway and CallManager.
Chapter 15, “Using TCL Scripts and VoiceXML”—This chapter explains how to support interactive voice response (IVR) and advanced call-handling applications using Toolkit Command Language (Tcl) and VoiceXML. It gives examples, tells how to download Tcl scripts, and shows how to configure gateways to use them. This chapter also examines the auto attendant (AA) script and creation of audio files in more detail and discusses some caveats and restrictions.
Chapter 16, “Deploying Gatekeepers”—This chapter provides the conceptual information that you need to integrate gatekeepers into your VoIP network. It describes gatekeeper functionality in networks with and without a CallManager, gatekeeper CAC and address resolution, redundancy and load balancing, gatekeeper security, and use of hierarchical gatekeepers. A discussion of planning a gatekeeper implementation includes placement, bandwidth, and dial plan issues. Scalabilty and multizone enhancements are also addressed.
Chapter 17, “Gatekeeper Configuration”—This chapter is a practical guide to gatekeeper configuration. It covers router configuration for basic and advanced gatekeeper functions such as directory gatekeepers, gatekeeper redundancy, Resource Availability Indication (RAI), and security. CallManager configuration for gatekeeper use and gatekeeper redundancy is also discussed, along with ways to troubleshoot your gatekeeper implementation.
Chapter 18, “Cisco Multiservice IP-to-IP Gateway”—This chapter explains the functions of an IP-to-IP gateway; its use in service provider and enterprise environments and in CallManager networks; its use of the Open Settlement Protocol; and the way it functions with RSVP. This chapter also covers using these gateways with both H.323 and SIP, configuring IP-to-IP gateways, configuring a via-zone gatekeeper, and troubleshooting the configuration.