Module 1: Introduction to Packet Voice Technologies

 

The purpose of this module is to describe the similarities and differences between traditional PSTN voice networks and IP telephony solutions.

 

Lesson: Understanding Traditional Telephony

 

This lesson defines how to identify the components, processes, and features of traditional telephony networks that provide end-to-end call functionality. The lesson includes these topics:

· Overview

· Basic Components of a Telephony Network

· CO Switches

· Private Switching Systems

· Call Signaling

· Multiplexing Techniques

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Describe the components and functionality of traditional telephony networks

· Explain how CO switches process telephone calls

· Identify types of private switching systems used in traditional telephony networks and list the main features of each

· Describe the three types of signaling in traditional telephony networks and identify how each is used

· Describe two methods used to multiplex voice in traditional telephony networks

 

Lesson: Understanding Packetized Telephony Networks

 

This lesson defines how to describe two methods of call control used on voice and data networks and provide one protocol example for each. The lesson includes these topics:

· Overview

· Benefits of Packet Telephony Networks

· Call Control

· Distributed vs. Centralized Call Control

· Packet Telephony Components

· Best-Effort Delivery of Real-Time Traffic

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· List five benefits of Packet Telephony Networks compared to circuit-switched networks

· Briefly describe three mechanisms of call control that are used in packet telephony

· Compare the gateway functions and signaling processes in centralized and distributed call control models

· List seven basic components of a packet voice network

· Identify a solution for transmitting real-time traffic, such as VoIP, on a best-effort delivery network such as IP

 

Lesson: Understanding IP Telephony Applications

 

This lesson defines how to list five components or capabilities that are required to provide integrated voice and data services in campus LAN, enterprise, and service provider environments. The lesson includes these topics:

 

Overview

 

· Analog Interfaces

· Digital Interfaces

· IP Phones

· Campus LAN Environment

· Enterprise Environment

· Service Provider Environment

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Describe the three basic analog interfaces and the telephony devices that they connect

· Describe the three basic digital interfaces and the type of signaling that each interface supports

· Describe the three physical connectivity options for IP Phones and explain the functioning of the Cisco IP Phone

· List five basic components of an integrated voice and data campus LAN network

· Explain the difference between distributed and centralized enterprise environments and list five main components of each enterprise network type

· List four capabilities that allow service provider networks to offer more efficient, less expensive alternatives to the PSTN

 

Module 2: Analog and Digital Voice Connections

 

The purpose of this module is to explain the processes and standards for voice digitization, compression, digital signaling, and fax transport as they relate to VoIP networks.

 

Lesson: Understanding Analog Voice Basics

 

This lesson defines how to select the appropriate analog voice connection to a Cisco device. The lesson includes these topics:

· Overview

· Local-Loop Connections

· Types of Local-Loop Signaling

· Supervisory Signaling

· Address Signaling

· Informational Signaling

· Trunk Connections

· Types of Trunk Signaling

· E&M Signaling Types

· Trunk Signal Types Used by E&M

· Line Quality

· Management of Echo

· Summary

· Quiz

· Lab Exercise 2-1

 

After completing this lesson, the learner will be able to:

· Identify the components of local-loop connections

· Identify the signaling used on local loops

· Identify on-hook, off-hook, and ringing supervisory signaling

· Identify pulse dialing and dual tone multifrequency

· List the call-progress indicators and their functions

· State the purpose and types of trunk connections

· Identify the signaling used on trunks

· Identify the five types of wiring schemes used in E&M signaling

· Describe Wink-Start, immediate-start, and delay-start signaling

· Describe impairments that interfere with line quality

· Identify two methods that are used in the industry to reduce the problem of echo

 

Lesson: Understanding Analog-to-Digital Voice Encoding

 

This lesson defines how to choose a voice compression scheme that best suits your needs. The lesson includes these topics:

· Overview

· Basic Voice Encoding: Converting Analog to Digital

· Basic Voice Encoding: Converting Digital to Analog

· The Nyquist Theorem

· Quantization

· Voice Compression and Codec Standards

· G.729 and G.729A Compared

· Compression Bandwidth Requirements

· Voice Quality Measurement

· Summary

· Quiz

· Lab Exercise 2-1

 

After completing this lesson, the learner will be able to:

· Identify the steps for converting analog signals to digital signals

· Identify the steps for converting digital signals to analog signals

· State the purpose of the Nyquist Theorem

· Explain quantization

· Name two types of voice compression techniques

· Describe the similarities and differences of G.729 and G.729A (Annex A) compression

· List three common voice compression standards and their bandwidth requirements

· State the purpose of voice quality measurement and the method of calculation for each type of measurement

 

Lab Exercise: Lab Familiarity

 

After completing this exercise, learners will be able to:

· List the data interfaces and voice ports on your routers

· Refer to the other routers in your pod, and elsewhere, by aliases

· Access the client/servers in all other pods in the classroom

· Access the classroom servers

 

Lesson: Understanding Signaling Systems

 

This lesson defines how to describe the appropriate signaling method to deploy in a telephony system. The lesson includes these topics:

· Overview

· CAS Systems: T1

· CAS Systems: E1

· CCS Systems

· ISDN

· QSIG

· DPNSS

· SIGTRAN

· SS7

· Signaling Systems Interworking

· Summary

· Quiz

· Lab Exercise 2-1

 

After completing this lesson, the learner will be able to:

· State the uses and types of CAS systems used for T1

· State the uses and types of CAS systems used for E1

· State the uses and types of CCS systems

· Name the benefits of using ISDN to convey voice

· Name the benefits of using QSIG to convey voice

· Name the functions of DPNSS signaling and its benefits

· Name the functions of SIGTRAN signaling and its benefits

· Name the functions of SS7 and its benefits

· Describe how separate signaling systems can be interfaced

Lesson: Understanding Fax and Modem over VoIP

 

This lesson defines how to implement an effective method of transporting fax and modem traffic over a VoIP network. The lesson includes these topics:

· Overview

· Cisco Fax Relay

· T.38 Fax Relay

· T.37 Fax Store and Forward

· Fax Pass-Through

· Modem Pass-Through

· Modem Relay

· Summary

· Quiz

· Lab Exercise 2-1

 

After completing this lesson, the learner will be able to:

· State the method used for conveying fax using Cisco fax relay

· State the method used for conveying fax using T.38 fax relay

· State the applications and the method used to convey fax using T.37 store-and-forward fax

· Describe how fax pass-through operates in a VoIP network

· Describe how modem pass-through operates in a VoIP network

· Describe how modem relay operates in a VoIP network

 

Module 3: Configuring Voice Interfaces

 

The purpose of this module is to configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, nonpacketized telephony equipment.

Lesson: Configuring Voice Ports

 

This lesson defines how to configure analog and digital voice interfaces as new devices are introduced into the voice path. The lesson includes these topics:

· Overview

· Voice Port Applications

· FXS Ports

· FXO Ports

· E&M Ports

· Timers and Timing

· Digital Voice Ports

· ISDN

· CCS Options

· Monitoring and Troubleshooting

· Summary

· Quiz

· Lab Exercise 3-1

 

After completing this lesson, the learner will be able to:

· Provide examples of seven types of voice port applications

· Set the configuration parameters for FXS voice ports

· Set the configuration parameters for FXO voice ports

· Set the configuration parameters for E&M voice ports

· Set timers and timing requirements on ports to adjust the time allowed for specific functions

· Set the configuration parameters for digital voice ports

· Set the configuration parameters for ISDN voice ports

· Configure T-CCS in a VoIP environment

· Use the show, debug, and test commands to monitor and troubleshoot voice ports

 

Lesson: Adjusting Voice Quality

 

This lesson defines how to configure analog and digital voice ports for optimal voice quality. The lesson includes these topics:

· Overview

· Electrical Characteristics

· Voice Quality Tuning

· Echo Cancellation Commands

· Summary

· Quiz

· Lab Exercise 3-1

 

After completing this lesson, the learner will be able to:

· Describe the electrical characteristics of analog voice and the factors affecting voice quality

· Configure voice port parameters to fine-tune voice quality

· Configure echo cancellation on the voice ports to improve voice quality

 

Lab Exercise: Voice Port Configuration

 

After completing this exercise, the learner will be able to:

· Identify default voice port settings

· Customize and verify analog port operations

· Create, customize, and verify digital port operations

 

Module 4: Voice Dial Plans

 

The purpose of this module is to configure the call flows for POTS, VoIP, and default dial peers.

 

Lesson: Understanding Call Establishment Principles

 

This lesson defines how to describe how call legs relate to inbound and outbound dial peers by following all the steps in the call setup process. The lesson includes these topics:

· Overview

· What Are Call Legs?

· End-to-End Calls

· Summary

· Quiz

· Lab Exercise 4-1

 

After completing this lesson, the learner will be able to:

· Describe call legs and their relationships to other components

· Describe how call legs are interpreted by routers to establish end-to-end calls

 

Lesson: Configuring Dial Peers

 

This lesson defines how to describe the proper use of digit manipulation and configuration of dial peers to implement a successful VoIP network. The lesson includes these topics:

· Overview

· Understanding Dial Peers

· Configuring POTS Dial Peers

· Configuring VoIP Dial Peers

· Configuring Destination-Pattern Options

· Default Dial Peer

· Matching Inbound Dial Peers

· Matching Outbound Dial Peers

· Hunt-Group Commands

· Configuring Hunt Groups

· Digit Collection and Consumption

· Understanding Digit Manipulation

· Summary

· Quiz

· Lab Exercise 4-1

 

After completing this lesson, the learner will be able to:

· Describe dial peers and their application

· Configure POTS dial peers

· Configure VoIP dial peers

· Describe destination-pattern options and the applicable shortcuts

· Describe the default dial peer

· Describe how the router matches inbound dial peers

· Describe how the router matches outbound dial peers

· List hunt-group commands

· Configure hunt groups

· Describe how the router and the attached telephony equipment collect and consume digits, and apply them to the dial peer

· Describe digit manipulation and the commands that are used to connect to a specified destination

 

Lab Exercise: POTS Dial Peers

 

After completing this exercise, the learner will be able to:

· Configure dial peers for locally terminated calls, PBX calls, and PSTN calls

· Determine appropriate method of digit forwarding and manipulation

· Create hunt groups and determine hunting behavior

 

Lesson: Understanding Special-Purpose Connections

 

This lesson defines how to configure voice ports for connection types necessary to integrate VoIP technologies with legacy PBXs and PSTN correctly. The lesson includes these topics:

· Overview

· Connection Commands

· PLAR and PLAR-OPX

· Configuring Trunk Connections

· Tie-Line Connections

· Summary

· Quiz

· Lab Exercise 4-2

 

After completing this lesson, the learner will be able to:

· Identify different special-purpose connection commands

· Describe how the network establishes PLAR and PLAR-OPX connections

· Configure trunk connections

· Describe how the network establishes tie-line connections

 

Lab Exercise: Special-Purpose Connections

 

After completing this exercise, the learner will be able to:

· Simulate auto attendant functions through use of PLAR and PLAR-OPX

· Create a tie-line connection for calls between two PBXs

· Use appropriate show and debug commands to monitor and troubleshoot the connections

 

Lesson: Building a Scalable Numbering Plan

 

This lesson defines how to assess the need for and implement a scalable numbering plan in a VoIP network. The lesson includes these topics:

· Overview

· Scalable Numbering Plan

· Scalable Numbering Plan Attributes

· Hierarchical Numbering Plans

· Internal Numbering and Public Numbering Plan Integration

· Enhancing and Extending an Existing Plan to Accommodate VoIP

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· List four customer components that must be considered when implementing a scalable numbering plan and explain why they are important

· Describe the required attributes of a scalable numbering plan and list the benefits provided

· Describe the five attributes and advantages of a hierarchical numbering plan and list the benefits provided

· Describe the challenges associated with the integration of internal numbering with the public numbering plan

· Describe two methods that are used to integrate existing dial plans into a VoIP network

 

Module 5: Introduction to VoIP

 

The purpose of this module is to describe the fundamentals of VoIP and identify challenges and solutions regarding its implementation.

 

Lesson: Understanding the Requirements of Voice in an IP Internetwork

 

This lesson defines how to determine the best method for improving delivery of voice packets with minimal loss, delay, or jitter. The lesson includes these topics:

· Overview

· Real-Time Voice in a Best-Effort IP Internetwork

· Packet Loss, Delay, and Jitter

· Consistent Throughput

· Reordering of Voice Packets

· Reliability and Availability

· Summary

· Quiz

· Lab Exercise 5-1

 

After completing this lesson, the learner will be able to:

· Explain which characteristics of IP networks cause problems for real-time traffic

· Describe packet loss, delay, and jitter problems in IP networks and identify a solution for each problem

· Describe five techniques that are used by Cisco IOS software to ensure consistent delivery and throughput of voice packets in an IP network

· Illustrate the steps used by the RTP to reorder packets on IP networks

· Describe four methods for improving reliability and availability of the IP internetwork for the delivery of voice packets

 

Lesson: Understanding Gateways and Their Roles

 

This lesson defines how to select the correct gateway for an enterprise and service provider network. The lesson includes these topics:

· Overview

· Understanding Gateways

· Guidelines for Selecting the Correct Gateway

· Determining Gateway Interconnection Requirements in an Enterprise Environment, Central and Remote Site

· Determining Gateway Interconnection Requirements in a Service Provider Environment

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Describe the role of gateways and their application when connecting VoIP to traditional PSTN and telephony equipment

· Select the correct gateway to connect VoIP to traditional PSTN and telephony equipment

· Determine gateway interconnection requirements in an enterprise environment

· Describe three gateway interconnection requirements in service provider environments

 

Lesson: Encapsulating Voice in IP Packets

 

This lesson defines how to reduce header size to efficiently carry voice across the network, using VoIP protocols and CRTP. The lesson includes these topics:

· Overview

· Major VoIP Protocols

· RTP and RTCP

· Reducing Header Overhead with CRTP

· When to Use RTP Header Compression

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Define the major VoIP protocols and how they map to the seven layers of the OSI model

· Describe the functions of RTP and RTCP as they relate to a VoIP network

· Describe how IP voice headers are compressed using CRTP

· Identify three conditions that necessitate the compression of the RTP header

Lesson: Calculating Bandwidth Requirements

 

This lesson defines how to list the bandwidth requirements for various codecs and data links, and the methods to reduce bandwidth consumption. The lesson includes these topics:

· Overview

· Codec Bandwidths

· Impact of Voice Samples and Packet Size on Bandwidth

· Data Link Overhead

· Security and Tunneling Overhead

· Specialized Encapsulations

· Calculating the Total Bandwidth for a VoIP Call

· Effects of VAD on Bandwidth

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· List five types of codecs and their associated bandwidth requirements

· Describe how the number of voice samples that are encapsulated impacts bandwidth requirements

· List the overhead for various Layer 2 protocols

· Describe how IPSec and GRE/L2TP affect bandwidth overhead

· Describe how MPLS, MLP, and other technologies affect bandwidth overhead

· Use a formula to calculate the total bandwidth that is required for a VoIP call

· Describe the operation of, and bandwidth savings associated with, the use of VAD

 

Lesson: Understanding Security Implications

 

This lesson defines how to describe the implications of implementing security measures in IP networks that will transport voice. The lesson includes these topics:

· Overview

· Security Policies for VoIP Networks

· Cisco SAFE Blueprint for VoIP

· Communicating Through a Firewall

· Delivery of VoIP Through a VPN

· Bandwidth Overhead Associated with VPN

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Describe three elements of a security policy that are necessary for a VoIP network

· Describe how the Cisco SAFE Blueprint details best practices for secure VoIP

· Describe the dynamic access control process used by firewalls to allow voice packets to pass

· Outline the steps needed to reduce overhead and delay for VoIP in a VPN

· Describe how to calculate the bandwidth required for a VPN packet

 

Lab Exercise: Basic VoIP

 

After completing this exercise, the learner will be able to:

· Configure VoIP connections

· Describe how dial-peer matching occurs

· Describe and configure proper use of dial-peer codec parameters

· Verify basic call setup through debug commands

· Use appropriate show and debug commands to monitor and troubleshoot the connections

 

Module 6: VoIP Signaling and Call Control

 

The purpose of this module is to compare centralized and decentralized call control and signaling protocols.

Lesson: Understanding the Need for Signaling and Call Control

 

This lesson defines how to identify the appropriate call control model for your network. The lesson includes these topics:

· Overview

· VoIP Signaling

· Call Control Models

· Translation Between Signaling and Call Control Models

· Call Setup

· Call Administration and Accounting

· Call Status and Call Detail Records

· Address Management

· Admission Control

· Centralized Call Control

· Distributed Call Control

· Centralized Call Control vs. Distributed Call Control

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Describe the endpoints and the common control components that are used in VoIP signaling

· List five protocols used for call control in VoIP

· Explain why the call control gateway must translate signals from different call control models to support end-to-end calls

· List five call parameters that must be negotiated before placing a call

· Name the functions that are performed by call accounting and administration as common call control components

· Name the functions that are performed by call status and call detail records as common call control components

· Describe the address registration and address resolution functions that are performed by the address management common control component

· Explain how admission control can protect network resources when it is used as a common control component

· Illustrate the setup of a centralized call control model

· Illustrate the setup of a distributed call control model

· Identify the advantages and disadvantages of centralized and distributed call control

 

Lesson: Configuring H.323

 

This lesson defines how to configure, monitor, and troubleshoot H.323 gateways and gatekeepers. The lesson includes these topics:

· Overview

· H.323 and Associated Recommendations

· Functional Components of H.323

· H.323 Call Establishment and Maintenance

· Call Flows Without a Gatekeeper

· Call Flows with a Gatekeeper

· Multipoint Conferences

· Call Flows with Multiple Gatekeepers

· Survivability Strategies

· H.323 Proxy Server

· Cisco Implementation of H.323

· Configuring H.323 Gateways

· Configuring H.323 Gatekeepers

· Monitoring and Troubleshooting

· Summary

· Quiz

· Lab Exercise 6-1

 

After completing this lesson, the learner will be able to:

· Describe the recommendations that are associated with H.323 and the control and signaling functions they perform

· List the functions that are performed by the components of an H.323 environment

· Identify three types of end-to-end connections that are established with H.323 and list eight types of registration, admission, and status protocol messages that are used to establish these connections

· Provide two scenarios of call flow without a gatekeeper

· Provide a scenario of call flow with a gatekeeper and explain the function of gatekeeper-routed call signaling in the call setup

· Describe three types of multipoint conferences supported by H.323

· Provide a scenario of call flow with multiple gatekeepers and explain how this allows scalability

· Describe four strategies that are used by H.323 to provide fault-tolerant networks

· List the steps involved in using an H.323 proxy server to set up end-to-end connections

· List the components that are supported by the Cisco Systems implementation of H.323

· Use the commands that are required to configure gateways in a two-zone, two-gatekeeper scenario

· Use the commands that are required to configure gatekeepers in a two-zone, two-gatekeeper scenario

· List the show and debug commands that are used to monitor and troubleshoot Cisco H.323 gateways and gatekeepers

 

Lab Exercise: VoIP with H.323

 

After completing this exercise, the learner will be able to:

· Configure single-zone and multizone H.323 gatekeeper environments for VoIP scalability

· Use debug and show commands to monitor the status and progress of call setup procedures in an H.323 environment

 

Lesson: Configuring SIP

 

This lesson defines how to configure, monitor, and troubleshoot SIP on a Cisco router. The lesson includes these topics:

· Overview

· SIP and Its Associated Standards

· Components of SIP

· SIP Messages

· SIP Addressing

· Call Setup Models

· Survivability Strategies

· Cisco Implementation of SIP

· Configuring SIP on a Cisco Router

· Monitoring and Troubleshooting

· Summary

· Quiz

· Lab Exercise 6-2

 

After completing this lesson, the learner will be able to:

· Describe three IETF standards that help SIP in the establishment, maintenance, and termination of multimedia sessions

· List the types of user agents and servers that are used by SIP and describe their functions

· List six examples of SIP request and response messages

· Identify three types of SIP addresses and the servers that are involved in address registration and resolution

· Describe three SIP call setup procedures and list their advantages and disadvantages

· Illustrate two strategies that are used by SIP to provide fault tolerance

· List SIP gateway and network server devices that are supported by Cisco Systems

· Use the sip-ua command with subcommands to configure SIP on a Cisco router

· Use show and debug commands to monitor and troubleshoot SIP

 

Lab Exercise: VoIP with SIP

 

After completing this exercise, the learner will be able to:

· Configure dial peers to use SIP call control procedures to set up VoIP calls

 

Lesson: Configuring MGCP

 

This lesson defines how to configure, monitor, and troubleshoot MGCP on a Cisco router. The lesson includes these topics:

· Overview

· MGCP and Its Associated Standards

· Basic MGCP Components

· MGCP Endpoints

· MGCP Gateways

· MGCP Call Agents

· Basic MGCP Concepts

· MGCP Calls and Connections

· MGCP Events and Signals

· MGCP Packages

· MGCP Digit Maps

· MGCP Control Commands

· Call Flows

· Survivability Strategies

· Cisco Implementation of MGCP

· Understanding Basics of Cisco CallManager

· Configuring MGCP

· Monitoring and Troubleshooting MGCP

· Summary

· Quiz

· Lab Exercise 6-3

 

After completing this lesson, the learner will be able to:

· Define MGCP and its functions

· List the basic components of MGCP

· Name eight types of MGCP endpoints defined in RFC 2705 and identify their functions

· Name seven types of MGCP gateways defined in RFC 2705 and identify their functions

· Describe the function of a call agent in an MGCP environment

· List the basic concepts of MGCP

· List the steps involved in the process of MGCP call establishment

· Describe the function of MGCP events and signals and give five examples of each

· List eight types of MGCP gateways and the packages that are associated with each gateway

· Explain how digit maps reduce the load on the network during call setup

· List nine MGCP control messages that are used to control and manage endpoints and their connections

· Describe MGCP call setup and control procedures

· Identify two strategies implemented by Cisco Systems to provide high availability in an MGCP environment

· Identify the Cisco devices that implement MGCP

· List the basic steps necessary to implement Cisco CallManager as an MGCP call agent

· Use the mgcp command and subcommands to configure an MGCP residential and trunk gateway on a Cisco router

· Use show and debug commands to monitor and troubleshoot MGCP

 

Lab Exercise: VoIP with MGCP

 

After completing this exercise, the learner will be able to:

· Configure your routers as MGCP residential gateways and have the routers use an MGCP call agent to establish voice calls between them

· Use debug commands to analyze the interactions between MGCP gateways and a call agent

· Use show commands to view the status of MGCP endpoints, connections, and calls

 

Lesson: Comparing Call Control Models

 

This lesson defines how to determine the best call control model for your network. The lesson includes these topics:

· Overview

· Feature Comparison Charts

· Strengths of H.323, SIP, and MGCP

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Compare the features and benefits of H.323, SIP, and MGCP

· Describe the environments best suited to H.323, SIP, and MGCP

 

Module 7: Improving and Maintaining Voice Quality

 

The purpose of this module is to describe specific voice quality issues and the QoS solutions used to solve them.

Lesson: Comparing Voice Quality Measurement Standards

 

This lesson defines how to provide voice quality on a network. The lesson includes these topics:

· Overview

· Audio Clarity

· Comfort Factors

· MOS and PSQM

· PESQ

· Comparison of Codec Quality Scores

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· List the attributes that affect audio clarity

· Describe the psychological comfort factors that affect voice quality

· State the purpose of MOS and PSQM and the methods used to calculate them

· State the purpose of PESQ and the method used to calculate it

· Describe how the codec voice quality scores differ and how they are measured

 

Lesson: Understanding VoIP Challenges

 

This lesson defines how to implement a converged voice and data IP network. The lesson includes these topics:

· Overview

· IP Networking Overview

· Jitter

· Delay

· Effect of Packet Loss on Quality

· Competition Between Voice, Data, and Video for Bandwidth

· Packet Sequencing

· Reliability and Availability

· Summary

· Quiz

 

After completing this lesson, the learner will be able to:

· Name some of the inherent problems that occur when delivering voice in IP networks

· Describe the cause of jitter and the solution used to eliminate it from an IP network

· List the two types of packet delay and the solution that is used to eliminate packet delay from an IP network

· State the cause of packet loss and its effect on voice quality

· Describe the contention issues associated with transmitting voice, data, and video on the same outbound interface

· Describe the method that is used to sequence voice packets

· Describe the effect of circuit reliability and availability on voice quality