Larry Keefer

Traditional Voice Telephony Principles

This chapter provides an overview of the traditional voice telephony principles. The chapter begins with elaborating concepts related to analog systems. Analog refers to transmission of electronic information achieved by adding signals of varying frequency or amplitude to a carrier wave of a given frequency. Telephone systems use analog-switched lines to provide voice communications by converting sound waves, vibrations that move in the air, into electrical signals. Each telephone handset contains a transmitter covered by a diaphragm and a receiver composed of a coil attached to a speaker cone that vibrates, producing sound waves. When a person speaks into a telephone handset, acoustical energy vibrations caused by the persons voice apply varying amounts of pressure to the diaphragm. In response to the natural rise and fall of human speech, the diaphragm in turn converts this pressure into different amounts of current or electrical energy. This variation in the current is in effect an electrical representation of the human voice. This chapter discusses in depth about analog network components, voice encoding, waveform encoding, and source encoding. Basic concepts of analog signaling and digital transmission are also discussed.

VoIP Signaling and Voice Transport Protocols

This chapter provides an in-depth investigation of VoIP signaling and voice transport protocols, and explains the fundamental technologies and concepts behind them. The chapter introduces the TCP/IP protocol and provides an overview of the TCP/IP Department of Defense (DoD) protocol stack as well as the International Organization for Standardization (ISO) protocol stack. The chapter also discusses some of the fundamental concepts of IP addresses, including IP addressing, address classes, private addressing, subnetting, supernetting, fixed-length subnet masking, and variable-length subnet masking. The chapter elaborates basics of VoIP signaling and an introduction to VoIP signaling, addressing, and routing. The chapter introduces the basic concept of how an end-to-end VoIP call is completed and highlights some of the protocols that come into play in making a call happen. The chapter also explains the Session Initiation Protocol (SIP) and explains the components as well as the benefits of deploying SIP in a VoIP network. The discussion of Media Gateway Control Protocol (MGCP) introduces the specifics of the protocol and explains its various components.

Testing and Troubleshooting VoIP

This chapter presents a logical and methodological approach for testing, troubleshooting, and supporting VoIP. Troubleshooting VoIP is challenging, especially if the efforts are not structured. In order to successfully troubleshoot a VoIP, one needs thorough understanding of the steps and commands necessary to configure VoIP. This chapter discusses different tools and techniques needed to begin troubleshooting and supporting a VoIP configuration. This chapter uses the Open Systems Interconnection (OSI) model to guide VoIP troubleshooting. The OSI model provides a framework for developing and deploying network protocols, layer by layer. The OSI model itself is not a network protocol or stack such as TCP/IP or AppleTalk, but a model for developing network protocols. Its primary mission is to guide the development efforts by providing a logical structure. Each of the seven layers governs a particular aspect of networking.

An Overview of Cisco's VoIP Components

This chapter introduces various Cisco components used to implement a VoIP solution. Cisco has long been a leader in the VoIP market and is constantly developing equipment to keep up with this demand. Cisco has a broad portfolio of routers, switches, and access gateways used to complete the convergence of voice, video, and date networks. Implementing a VoIP solution can be quite a daunting task, depending on the type of environment for which it is needed. These environments can include small offices, branch offices, and widely dispersed enterprises. Numerous voice-related components are required to complete a total VoIP solution. This chapter covers the infrastructure components: voice network modules (VNMs), voice interface cards (VICs), and a few of the Cisco routers and switches used for supporting voice traffic. The chapter discusses various types of voice ports available to allow a voice call to occur. It elaborates hardware components required to support these voice ports and highlights the subsequent components required for the delivery of voice from a packet-switched (VoIP) network to a circuit-switched (PSTN) network and back. This chapter also discusses VoIP terminology to help identify some of the common terms and acronyms used in modern voice networking.

Introduction to Voice Over IP and Business Justifications

This chapter focuses on cost benefit factor associated with deploying Voice over IP (VoIP) networks, as well as the Cisco IP Telephony solution. Most companies have spent exorbitant amounts of money to install and maintain their PBXs. Packetizing voice allows for tremendous cost savings. As more standards are ratified, the cost of setting up a VoIP network continues to drop. This is quite a different model from the traditional PBX cost trends of the last few decades. This chapter also explores how to build an ROI proposal that in most cases justifies a conversion to packetized Voice. The chapter presents examples of cost justification and some return on investment (ROI) scenarios. This chapter explains the basic differences between circuit-switched and packet-switched networks. It discusses needs and cost justification for toll-bypass solutions, and explores the opportunities for replacing the traditional private branch exchange (PBX) with the Cisco IP Telephony system. The chapter reviews software integration possibilities such as TAPI integration, and elaborates the link layer VoIP technologies such as voice over Frame Relay (VoFR) and voice over asynchronous transfer mode (VoATM). This chapter also discusses advanced VoIP features such as Web integration, multimedia integration, and telephony application programming interfaces (TAPI).

Connecting PABXs with VoIP Scenarios

This chapter explains connecting private automatic branch exchanges PABXs with VoIP. Connecting PABXs with a VoIP solution is a simple method for introducing VoIP to an organization. Many corporations currently use tie lines, leased lines connecting PABXs, to connection offices that are in different geographic locations. This method of toll bypass is used to escape long distance charges that would be applied had the calls between the offices crossed the PSTN. In this chapter, a case study is developed with a sample corporation to test VoIP as a replacement for that companys current voice connections between PABXs. This progression is followed from information gathering to acceptance testing. The chapter explores some of the techniques for collecting information and ways to assemble that information into a logical presentation for company officers or key decision makers. The chapter discusses steps involved in the design process and uses the design to create configurations for a VoIP tie-line replacement. Some advanced trunking scenarios are explored in the chapter, followed by integration of voice and data networks. The chapter gives a brief discussion of basic testing and verification of the VoIP solution.

Intra- and Interoffice VoIP Scenarios

This chapter discusses various scenarios for creating intra- and interoffice dial plans. Designing and implementing an integrated voice and data system should follow the same procedures, independent of the scope of the installation. This chapter introduces the process to be followed by presenting best-practices design methodology. The chapter focuses on a simple single voice-enabled router design and builds on the deployment in successive steps. This is done by discussing how to incorporate legacy PBX equipment into design. The deployment is built on by adding multiple sites to the design. QoS tuning parameters found useful to assure voice quality within the network are presented. The chapter highlights how to design and implement a simple single-office dial plan with only a few phones and then extend it across the enterprise. The chapter demonstrates how digit manipulation can be configured for use within the dial plan.

Configuring QoS for VoIP

This chapter discusses configuring QoS (Quality of Service) for VoIP. The term QoS is used to address the unique problem of making voice and data traffic get along. QoS defines the ability of a network to provide different levels of service assurances to the various forms of traffic. QoS gives network administrators the ability to give priority to their voice traffic or alter the flow of their data traffic to prevent it from choking out voice. A host of options are available for configuring a network to enable voice and data to share the resources without sacrificing quality for either service. In order to implement QoS, it is important to take into account the types of network, number of users, and bandwidth available. It is important to focus on the quality that is acceptable and to consider the other core business applications that will be sharing the network with the voice traffic as well as their relative priorities. This chapter provides an overview of QoS and a description of several different options for implementation. The chapter also explains into detail how to configure the network for each QoS solution.

Chapter 5 – VoIP Configurations

Publisher Summary This chapter discusses various voice-supported hardware modules, cabling, tuning, and gateway/gatekeeper configurations with respect to VoIP. The chapter provides examples of some actual configurations for a VoIP network. The chapter elaborates cabling and layout of various types of voice modules in the routers. The chapter discusses in detail the Cisco port- and module-numbering schemes that are essential to be understood in order to effectively configure a router for VoIP. Port numbering on the 1700 series, 2600 series, 3600 series, MC3810 series, 7200 series, and AS5x00 series is also explained. The chapter discusses how voice interface card (VIC) modules are used in the various routers. The chapter covers various analog interface configurations, pulling them together with the remaining commands to complete a VoIP configuration such as dial peers. All this information is obtained through the process of completing the dial plan. For installations that require greater capacity, digital voice module options are covered, with an emphasis on ISDN and its configuration. The chapter includes the handling of multiple sites that have gateways and explaining how they can communicate via a gatekeeper.

Old World Technologies

This chapter provides an introduction to private branch exchange (PBX). PBX-attached phones and services are augmented with Internet Protocol-based telephony and new integrated services, including video, desktop integration, and data services. Many of these functions and offerings are available, and a clear roadmap of what can be availed in the next year is easily identified. However, there are still hurdles to overcome, including reliability and simplicity. It is important to remember that while architecture for voice, video, and integrated data can provide many features in the network and easily replace the modern PBX in many organizations, the hurdles to this convergence include capital costs, support costs, functionality, training, and reliability. A PBX consists of hardware and software designed to emulate the public telephone system within a company and provide paths into the public switched telephone network. These systems can be categorized into four primary areas, each area containing one or more functions: extension termination; trunk termination; system logic and call processing; and switching.