KnowledgeNet IP Telephony Express (IPTX) 2.0 Student Guide.pdf

8/10/2019 KnowledgeNet IP Telephony Express (IPTX) 2.0 Student Guide.pdf

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Copyright 2005, Cisco Systems, Inc. All rights reserved.

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Table of ContentsVolume 1

Course Introduction 1

Overview 1Learner Skills and Knowledge 1

Course Goal and Objectives 2Course Flow Diagram 3Additional References 4

Cisco Glossary of Terms 4

Introducing Cisco CallManager Express 1-1

Overview 1-1Module Objectives 1-1

Describing Key Features of Cisco CallManager Express and CUE 1-3

Overview 1-3Objectives 1-3

What Is Cisco CallManager Express? 1-4What Is Cisco Unity Express? 1-6

How Do Cisco CallManager Express and Cisco Unity Express Work? 1-9Licensing 1-14Summary 1-19

Explaining Differences Between Traditional Telephony and VoIP 1-21


Traditional Telephony 1-22CO Switching Systems 1-25

PCM Theory 1-29

Basic Voice Encoding: Converting Digital to Analog 1-30The Nyquist Theorem 1-31Quantization 1-32

Coder-Decoder 1-34Encapsulating Voice in IP Packets 1-39

RTP Packet Components 1-42Summary 1-44

Understanding VoIP Challenges and Solutions 1-45


Requirements of Voice in an IP Internetwork 1-46Challenges in VoIP 1-54Bandwidth Requirements in VoIP 1-56Summary 1-63

Describing the Cisco CallManager Express Voice Packet Handling Methods 1-65


IP Phone Calls 1-66

Packet Forwarding, Voice Packet Priority, and RTP Stream Information 1-72WAN Call Setup 1-74Summary 1-78

Module Summary 1-79References 1-80

Module Self-Check 1-81Module Self-Check Answer Key 1-85

Configuring Cisco CallManager Express 2-1



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ii IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Understanding Cisco CallManager Express Features and Functionality 2-3


Key Benefits and Features 2-4Supported Platforms and Telephones 2-8Supported Protocols and Integration Options 2-26Cisco CallManager Express Requirements 2-33Cisco CallManager Express Restrictions 2-34

Summary 2-36

Configuring Cisco CallManager Express Network Parameters 2-37


Voice VLANs 2-38Configuring Voice VLANs 2-41DHCP Service Setup 2-44DHCP Relay Server 2-51Network Time Protocol 2-54Transcoding 2-59Summary 2-79

Understanding the IP Phone Registration Process 2-81

Overview 2-81

Objectives 2-81Files 2-82IP Phone Information 2-88Download and Registration 2-89Summary 2-95

Defining Ephone-dn and Ephone 2-97


Ephone-dn 2-99Ephone 2-102Type of Ephone-dns 2-108Number of Ephone-dns 2-124Summary 2-126

Describing Cisco CallManager Express Files 2-127Overview 2-127

Objectives 2-127Cisco CallManager Express Files 2-128Bundled Cisco CallManager Express Files 2-129Individual Cisco CallManager Express Files 2-131

GUI Files 2-132Cisco CallManager ExpressTAPI Integration 2-134Additional Files 2-135Summary 2-136

Understanding Initial Phone Setup 2-137


Setting Up Phones in a Cisco CallManager Express System 2-138Manual Phone Setup 2-139Partially Automated Phone Setup 2-150Automated Phone Setup 2-154Optional Parameters 2-159Rebooting Cisco CallManager Express Phones 2-163Setup Troubleshooting Tips 2-166Verifying Cisco CallManager Express Phone Configuration 2-171Summary 2-172Module Summary 2-173


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Copyright 2005, Cisco Systems, Inc. IP Telephony Express (IPTX) v2.0 iii

References 2-173Module Self-Check 2-174

Module Self-Check Answer Key 2-179

Volume 2

Configuring PSTN Interfaces and Voice Dial Peers 3-1

Overview 3-1

Module Objectives 3-1

Understanding Analog and Digital Voice Interfaces 3-3


Local-Loop Connections 3-4Analog Voice Interfaces 3-5Channel Associated Signaling Systems: T1 3-8Channel Associated Signaling Systems: E1 3-10Common-Channel Signaling Systems 3-12PRI and BRI 3-13Summary 3-14

Configuring Analog and Digital Voice Interfaces 3-15


Foreign Exchange Station Port Configuration 3-17Configuration Parameters 3-18

Foreign Exchange Office Port Configuration 3-20Configuration Parameters 3-20

Ear and Mouth Port Configuration 3-22Configuration Parameters 3-22

Timers and Timing 3-24Configuration Parameters 3-24

Digital Voice Port Configuration 3-26Configuration Parameters 3-26

Channel Associated Signaling Configuration 3-29Common-Channel Signaling: BRI 3-31Common-Channel Signaling: PRI 3-38

Summary 3-43

Configuring Dial Peers 3-45


What Is a Dial Peer? 3-46Plain Old Telephone Service Dial Peers 3-49

Example 3-50VoIP Dial Peers 3-51

Example 3-52

Destination-Pattern Options 3-53Example 3-55

What Is the Default Dial Peer? 3-56Example 3-57

Summary 3-58Understanding Call Setup and Digit Manipulation 3-59


What Are Call Legs? 3-60Example 3-60

End-to-End Calls 3-61Matching Inbound Dial Peers 3-63Matching Outbound Dial Peers 3-65


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iv IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Example 3-66Digit Collection and Consumption 3-67

Example 3-68What Is Digit Manipulation? 3-70

Example 3-72PLAR 3-74Summary 3-76

Understanding Class of Restriction 3-77


Class of Restriction 3-78Example: Incoming and Outgoing COR Example 3-79

Steps to Configure Class of Restriction 3-81Example: Name the COR and Lists 3-82Example: Define the COR Lists 3-83Example: Apply the COR to the Dial Peer 3-84Example: Apply the COR to Ephone-dns 3-85Example: COR Used to Restrict Access Internally Within Cisco CallManager Express 3-86

Summary 3-90

Describing H.450.x Protocols 3-91

Overview 3-91

Objectives 3-91H.450.x Series Protocols 3-92Call Transfer Using H.450.2 3-93Call Forwarding Using H.450.3 3-100H.450.12 3-106Issues and Workarounds for H.450.x Protocols 3-109Summary 3-116Module Summary 3-117



Configuring Additional Cisco CallManager Express Features 4-1

Overview 4-1

Module Objectives 4-2Configuring Cisco CallManager Express GUI Features 4-3


User Classes 4-4Cisco CallManager Express GUI Prerequisites 4-7Accessing the GUI 4-14Configuring Administrative User Classes 4-15

Defining the Customer Administrator Credentials 4-20Summary 4-25

Configuring Phone Features 4-27


Call Transfer 4-28Call Forwarding 4-35Call Waiting 4-41Call Park 4-44IP Phone Display 4-47Softkey Customization 4-55Calling and Directory Features 4-60Conferencing 4-65Productivity Tools 4-68Custom IP Phone Rings 4-78


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Copyright 2005, Cisco Systems, Inc. IP Telephony Express (IPTX) v2.0 v

Example: Sample RingList.xml 4-80Timer Settings 4-81Music on Hold 4-82Summary 4-92

Understanding Call Center Features 4-93


Ephone Hunt Groups 4-94

Dynamic Hunt Group Login and Logout 4-104Automatic Logout of a Hunt Group 4-106B-ACD Service 4-108Summary 4-129

Defining TAPI Support for Cisco CallManager Express 4-131


Functions and Features 4-132Cisco IOS TSP Configuration on the PC 4-134Cisco IOS TSP Configuration on the Router 4-136Modifying Cisco IOS TSP Configuration on the PC 4-137Cisco CallManager Express and Microsoft CRM Integration 4-139Summary 4-142

Describing Network Management for Cisco CallManager Express 4-143Overview 4-143

Objectives 4-143Syslog Messages and MIBs 4-144

Example: Syslog Messages 4-144Billing Support 4-146

Example: Viewing the Account Code from the CLI 4-147CDR 4-150CNS 4-151Summary 4-154

Reference 4-154Module Summary 4-155

Reference 4-155Module Self-Check 4-156


Volume 3

Configuring Cisco Unity Express Automated Attendant and Voice Mail 1

Overview 1Module Objectives 2

Understanding Cisco Unity Express Features and Functionality 3

Overview 3Objectives 3

Voice Mail Features 4Auto Attendant Features 6Management Features 7System Functionality 10Summary 13

Describing Cisco Unity Express Installation and Initialization 15


Cisco Unity Express Software Download 16Hardware Installation 18IOS Router and Cisco CallManager Express Prerequisite Configuration 28Connecting to the CUE Module 33


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vi IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Restoring the Factory Defaults 35Initial Configuration 36CUE Initialization Wizard 43

Step 3: System Defaults 48Restarting the CUE Module 52Upgrading CUE Software and License 53Summary 73

Configuring Cisco Unity Express Auto Attendant 75


CUE Auto Attendant Operation 76CUE AA Editor 82

Adding Variables 84Variable Types 85Step Reference: General Steps 89Step Reference: User and Prompt Steps 93Step Reference: Contact and Call Contact Steps 95Step Reference: Media Steps 97Validate the Script 99

Holiday List 100Business Hours Schedule 104Scripts and Prompts 109

Setting Up an Automated Attendant 120Case Study 136Emergency Alternate Greeting 142Administration via TUI 144Summary 146

Configuring Cisco Unity Express Users and Groups 147


User Interface 148User Configuration 155Group Configuration 166Group Mailboxes 180Summary 188

Configuring Cisco Unity Express Voice Mail 189Overview 189

Objectives 189Voice Mail Entry Point and Port 190Message Waiting Indicator Configuration 196Broadcast Messages 202Mailbox and Message Sizes and Defaults 207Personal Mailboxes 214VPIM Networking 222Distribution Lists 243Summary 251

Troubleshooting Cisco Unity Express 253

Overview 253

Objectives 253Introduction and Tools 254

Gather Facts and Define Problem 254Continue Gathering Facts 255Consider Possibilities 255

Create and Implement the Action Plan 255Observe Results 256Repeat As Necessary 256Document the Changes 256

Software Architecture Overview 286


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Copyright 2005, Cisco Systems, Inc. IP Telephony Express (IPTX) v2.0 vii

System-Level Troubleshooting 288GUI Troubleshooting 296Voice Mail and Automated Attendant 302Summary 316Module Summary 317

Reference 317Module Self-Check 318

Module Self-Check Answer Key 322

Volume 4

Introducing IP Quality of Service 6-1


Understanding Quality of Service 6-3


Quality of Service Defined 6-4Converged Networks 6-5Converged Networks Quality Issues 6-7Lack of Bandwidth 6-9

End-to-End Delay 6-11Example: Effects of Delay 6-12

Packet Loss 6-15QoS Requirements 6-17QoS Policy 6-20QoS for Converged Networks 6-22

Example: Traffic Classification 6-23Example: Defining QoS Policies 6-24

LAN QoS Considerations 6-25Summary 6-27

Describing the Differentiated Services Model 6-29


Differentiated Services Model 6-30

DSCP Encoding 6-32Per-Hop Behaviors 6-33Backward Compatibility Using the Class Selector 6-38Mapping CoS to Network Layer QoS 6-39Summary 6-40

Understanding IP QoS Mechanisms 6-41


QoS Mechanisms 6-42

Classification 6-43Marking 6-44Trust Boundaries 6-45Congestion Management 6-48

Traffic Shaping 6-49Compression 6-50Link Fragmentation and Interleaving 6-51Summary 6-52

Introducing Modular QoS CLI 6-53


Introducing Modular QoS CLI 6-54Modular QoS CLI Components 6-55


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viii IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Example: Configuring MQC 6-55Class Maps 6-56Configuring and Monitoring Class Maps 6-58

Example: Class-Map Example 6-58Example: Using the match Command 6-60Example: Nested Traffic Class to Combine match-any and match-all Characteristics in OneTraffic Class 6-60

Policy Maps 6-62

Configuring and Monitoring Policy Maps 6-63Example: Policy Map 6-64Example: Hierarchical Policy Map 6-67

Service Policy 6-70Attaching Service Policies to Interfaces 6-71

Example: Complete MQC Configuration 6-71Summary 6-73

Implementing AutoQoS 6-75


AutoQoS 6-76AutoQoS: Router Platforms 6-80AutoQoS: Switch Platforms 6-81AutoQoS Prerequisites 6-83

Configuring AutoQoS 6-85Example: Configuring the AutoQoS VoIP Feature on a High-Speed Serial Interface 6-86Example: Configuring the AutoQoS VoIP Feature on a Low-Speed Serial Interface 6-86Example: Using the Port-Specific AutoQoS Macro 6-90

Monitoring AutoQoS 6-92Example: show auto qos command and show auto qos interface command 6-93

Automation with Cisco AutoQoS 6-98Summary 6-99

Case Study: QoS Mechanisms 6-101

Overview 6-101Relevance 6-101Objectives 6-101Learner Skills and Knowledge 6-101

Required Resources 6-102Job Aids 6-102Outline 6-102Case Study Verification 6-102

Review Customer QoS Requirements 6-103Company Background 6-103Customer Situation 6-103

Identify QoS Service Class Requirements 6-105Identify Network Locations Where QoS Mechanisms Should be Applied 6-106

Present Your Solution 6-108Case Study Answer Key 6-109Module Summary 6-113

References 6-114Module Self-Check Overview 6-116

Module Self-Check Answer Key 6-118Designing Cisco CallManager Express and Cisco Unity Express Networks 7-1


Describing Deployment Scenarios and Design Considerations 7-3


Standalone Cisco CallManager Express 7-4


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Copyright 2005, Cisco Systems, Inc. IP Telephony Express (IPTX) v2.0 ix

Cisco CallManager Express in SIP Network 7-8Cisco CallManager Express Integration with Cisco CallManager 7-10Cisco CallManager Express Migration to Cisco CallManager and SRST 7-13Cisco CallManager Express H.323 Interoperability Solutions 7-15Summary 7-26

Deploying Voice Mail with Cisco CallManager Express 7-27


SIP Integration with Cisco Unity Express 7-28Skinny Integration with Cisco Unity Server 7-29Analog DTMF Integration 7-32Router Configuration: Two Commands 7-35Summary 7-38Module Summary 7-39




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x IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.


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IPTX

Course Introduction

OverviewIP Telephony Express(IPTX) v2.0 provides an understanding of Cisco CallManager Express

and Cisco Unity Express (CUE) and of the challenges you face when configuring and

deploying the systems. The course presents Cisco Systems solutions and implementation

considerations for addressing those challenges.

Learner Skills and Knowledge

This subtopic lists the skills and knowledge that learners must possess to benefit fully from the

course.

2005 Cisco Systems, Inc. All rights reserved. IPTX v2.07

Prerequisite Learner Skills

and Knowledge

WANs

IP Switching

Basic InternetworkingSkills

PSTN Operationsand Technologies

IPTX

LANs

PBX Essentials


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2 IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Course Goal and ObjectivesUpon completing this course, you will be able to meet these objectives:

Describe the similarities and differences between a traditional PSTN, voice networks, and

IP telephony solutions

Explain the processes and standards for voice digitization, compression, and digitalsignaling as they relate to VoIP networks

Configure voice interfaces on Cisco voice-enabled equipment for connection to traditional,

nonpacketized telephony equipment

Configure the Cisco CallManager Express system from either the CLI or a GUI web

interface

Understand and configure the devices for and connections to the Cisco CallManager

Express system

Configure the call flows for POTS, VoIP, and default dial peers

Describe the fundamentals of VoIP and identify challenges and solutions regarding its

implementationInstall and configure the CUE module for voice mail services

Troubleshoot both Cisco CallManager Express and CUE

Apply QoS to the IP network with the use of the AutoQoS

Apply your knowledge of Cisco CallManager Express and CUE to deploy and design an

installation


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Copyright 2005, Cisco Systems, Inc. Course Introduction 3

Course Flow DiagramThis topic covers the suggested flow of the course materials.


Course Flow Diagram

CourseIntroduction

Lunch

Configuring PSTNInterfaces and

Voice Dial Peers

DesigningCisco

CallManagerExpress andCisco Unity

ExpressNetworks

A

M

P

M

Day 1Day 2Day 3Day 4Day 5

Configuring CiscoUnity Express

AutomatedAttendant and

Voice Mail

Introducing IPQuality ofService

IntroducingCisco

CallManagerExpress

ConfiguringCisco

CallManagerExpress

ConfiguringCisco

CallManagerExpress

Configuring PSTNInterfaces and

Voice Dial Peers

ConfiguringAdditional Cisco

CallManagerExpress Features

ConfiguringAdditional Cisco

CallManagerExpress Features

Configuring CiscoUnity Express

AutomatedAttendant and

Voice Mail

ConfiguringCisco Unity

ExpressAutomated

Attendant andVoice Mail

DesigningCisco

CallManagerExpress andCisco Unity

ExpressNetworks

The schedule reflects the recommended structure for this course. This structure allows enough

time for the instructor to present the course information and for you to work through the lab

activities. The exact timing of the subject materials and labs depends on the pace of your

specific class.


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4 IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.

Additional ReferencesThis topic presents the Cisco icons and symbols used in this course, as well as information on

where to find additional technical references.


ATM

Switch

Cisco Icons and Symbols

Voice-

Enabled

Router

PBX

(small)

Network

Cloud,

StandardColor

Network

Cloud,

White

Phone

IP Phone

Phone 2

Generic

Softswitch

Voice-Enabled

ATM Switch

PIX Firewall

(right and left)

Cisco

CallManager

Express

PC

Laptop

Voice-Enabled

Communications

Server

Workgroup

Multilayer Switch,

with Text, without Text,

and Subdued

SiSi SiSi

PBX/

Switch

Web

BrowserServer

Cisco Glossary of Terms

For additional information on Cisco terminology, refer to the Cisco Internetworking Terms and

Acronyms glossary of terms at

http://www.cisco.com/univercd/cc/td/doc/cisintwk/ita/index.htm.


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Module 1

Introducing Cisco CallManagerExpress

OverviewCisco CallManager Express is an integrated call-processing solution that is based on Cisco

midrange access routers using Cisco IOS software. Cisco CallManager Express delivers

telephony services for up to 240 users in a small- to medium-sized office. It is part of Cisco IP

Communications Solution and works in conjunction with the extended Cisco Systems product

portfolio, including routers, data switches, public switched telephone network (PSTN)

gateways, gatekeepers, Cisco Unity voice mail, and analog terminal adapters.

Cisco CallManager Express delivers a robust set of telephony features that are similar to those

commonly used by business users. Cisco CallManager Express is an optional feature of Cisco

IOS software and is available on a wide range of Cisco access routers that support as many as240 IP Phones. This allows customers to take advantage of the benefits of IP communication

without the higher costs and complexity of deploying a server-based solution. Furthermore,

because the solution is based on the Cisco access router and IOS software, it is simple to deploy

and manage, especially for customers who already use IOS software products.

Cisco Unity Express (CUE) offers local voice-mail and automated attendant capabilities for IP

Phone users in a small office or branch location who are connected to Cisco CallManager or

Cisco CallManager Express. CUE is fully integrated into the branch office router, either on a

CUE network module (NM-CUE) or on a CUE advanced integration module (AIM-CUE).

Module Objectives

Upon completing this module, you will be able to describe the similarities and differences

between traditional telephony and Voice over IP (VoIP). This includes being able to meet these

objectives:

Describe the key features and functionality of the Cisco CallManager Express system

Explain the differences between traditional voice and VoIP

Describe the challenges and solutions associated with VoIP delivery in LAN and WAN

Describe the Cisco CallManager Express voice packet handling methods


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1-2 IP Telephony Express (IPTX) v2.0 Copyright 2005, Cisco Systems, Inc.


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Lesson 1

Describing Key Features ofCisco CallManager Express

and CUE

OverviewThis lesson describes the key features and functionality of Cisco CallManager Express and

Cisco Unity Express (CUE). This includes the licensing scheme and the effect of licensing on

activation of features. Learners will be directed to the Cisco website for up-to-date information

on licensing.

ObjectivesUpon completing this lesson, you will be able to explain the differences between traditional

voice and Voice over IP (VoIP). This includes being able to meet these objectives:

Define Cisco CallManager Express

Define CUE

Describe the functionality of Cisco CallManager Express and CUE

Describe licensing requirements and the effect of licensing on feature activation


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What Is Cisco CallManager Express?This topic describes the Cisco CallManager Express system.

2005 Cisco Systems, Inc. All rights reserved. IPTX v2.01-2

What Is Cisco CallManager Express?

Cisco CallManager Express

Trunks

Call processing for small-to medium-sized

deployments VoIP integrated solution

Up to 240 IP Phones

IOS software!based solution

WAN

PSTN

Cisco CallManager Express is an integrated call-processing solution that is based on Cisco

midrange access routers that are using Cisco IOS software that delivers telephony services for

10 to 100 users in small offices. Cisco CallManager Express is part of Cisco IP

Communications Solution and works in conjunction with the extended Cisco Systems product

portfolio, including routers, data switches, public switched telephone network (PSTN)

gateways, gatekeepers, Cisco Unity voice mail, and analog telephone adaptors (ATA).

Cisco CallManager Express delivers a robust set of telephony features that are similar to those

commonly used by businesses. Cisco CallManager Express is an optional feature of Cisco IOS

software and is available on a wide range of Cisco access routers that support as many as 240

IP Phones. This allows customers to take advantage of the benefits of IP communications

without the higher cost and complexity of deploying a server-based solution. Because the

solution is based on the Cisco access router and IOS software, it is simple to deploy and

manage, especially for customers who already use IOS software products. Cisco CallManager

Express allows customers to scale IP telephony to a small or branch office site with a solution

that is easy to deploy, administer, and maintain.


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Copyright 2005, Cisco Systems, Inc. Introducing Cisco CallManager Express 1-5


What Is Cisco CallManager Express?(Cont.)

2600XM3700 1700

3800 2800

Integrated services routers

Multiservice access routers

Cisco CallManager Express enables Ciscos large portfolio of multiservice access routers and

integrated services routers to deliver features that are similar to low-end PBX and key system

features, creating a cost-effective, highly reliable, feature-rich IP communications solution for

the small office.

Cisco CallManager Express supports a new generation of intelligent IP Phones with robust

display capabilities. End users can easily customize these Phones based on their changing

needs.


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What Is Cisco Unity Express?This topic describes the CUE system.


What Is Cisco Unity Express?

Voice mail and automated attendant for small andbranch offices

Fully integrated into Cisco 3800, 2800, 2600XM,2691 and 3700 series access routers

Two form factors: NM-CUE and AIM-CUE

Two call control options: Cisco CallManagerExpress and Cisco CallManager

CUE offers local voice-mail and automated attendant capabilities for IP Phone users connected

to Cisco CallManager or Cisco CallManager Express in a small office or branch location. CUE

is fully integrated into the branch office router on either a CUE network module (NM-CUE) or

a CUE advanced integration module (AIM-CUE).


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What Is Cisco Unity Express? (Cont.)

Voice Mail

Supports up to 100 subscriber mailboxes on the NM-CUE and

NM-CUE-EC Supports up to 50 subscriber mailboxes on the AIM-CUE

Storage is configurable per subscriber

Automated Attendant

Has up to five automated attendants per system

Offers fully customizable script-driven menu structure andmenu nesting

Has time of day and day of week call treatment

Business hours can be defined

Holidays can be defined

Voice mail is essential in most enterprises. Voice mail enables messages to be left for

subscribers when they are busy or do not answer a call in a specified amount of time.

An automated attendant is a device that automatically answers calls with an interactive

recording and allows callers to route their call to the desired person or department by entering

the appropriate extension using their telephone keypad. Businesses can customize the greeting

by adding information such as hours and directions.

CUE supports a built-in automated attendant along with its voice-mail capabilities.


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How Do Cisco CallManager Express and CiscoUnity Express Work?

This topic describes how Cisco CallManager Express and CUE work.


How Do Cisco CallManager Express andCisco Unity Express Work?

Phones register with Cisco CallManager Expressand are then under its control.

Cisco CallManager Express is anIOS software!based call control agent.

Register Register

The Cisco CallManager Express system provides PBX-like features and functions for IP

Phones. These features are a result of the concept of a centralized point of control and

intelligence. The Cisco CallManager Express router provides all of the call control and

intelligence needed for IP Phones to place and receive calls. In a Cisco CallManager Expressdeployment, the IP Phones are not capable of setting up a call by themselves. In fact, the IP

Phones are completely controlled by the Cisco CallManager Express system and are instructed

how to place and receive calls.

The IP Phones boot up and register with Cisco CallManager Express. If Cisco CallManager

Express is properly configured, calls will be able to be set up and torn down to and from the IP

Phones. The IP Phones and the Cisco CallManager Express router use Skinny Client Control

Protocol (SCCP) to communicate.

Note Registration across a WAN is not supported. The IP Phones must be on the local LAN with

the Cisco CallManager Express router.


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How Do Cisco CallManager Express andCisco Unity Express Work? (Cont.)

Call Control is centralized on Cisco CallManager Express.

SCCP SCCP

RTPPhone APhone B

Phone A places callto Phone B

RTP

Cisco CallManager Express is anIOS software!based call control agent.

When a call is placed between two IP Phones that are under the control of Cisco CallManager

Express, SCCP is used to set up the call. SCCP does not go between the two IP Phones, only

between the IP Phone and the Cisco CallManager Express system. After the call is set up, Real-

Time Transport Protocol (RTP) is used to carry the audio stream. RTP is a common protocol

that is used to carry time-sensitive traffic, such as voice and real-time video. RTP is carried

inside a User Datagram Protocol (UDP) segment, which is then carried inside an IP packet.

This is the sequence of events for a phone call:

Step 1 Phone A picks up the handset and dials the number of Phone B.

Step 2 The dialed digits are sent through SCCP to Cisco CallManager Express.

Step 3 Cisco CallManager Express knows the location of Phone B (because of the

registration) and its status (busy, on hook, off hook).

Step 4 Assuming that Phone B is on hook (available), Cisco CallManager Express sends an

SCCP message to tell Phone B about the incoming call and to tell it to ring.

Step 5 Phone B is answered.

Step 6 Cisco CallManager Express informs each IP Phone about the settings of the other

Phone and instructs both Phones to construct RTP connections.

Step 7 The IP Phones construct two one-way RTP connections for the voice to travelacross, one for Phone As voice to travel to B and one for Phone B s voice to travel

to A.

Step 8 The call takes place.

Step 9 Phone B hangs up, and an SCCP message is sent to Cisco CallManager Express.

Step 10 Cisco CallManager Express sends an SCCP message to Phone A telling it that the

call has been disconnected.


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Connection(s) to PSTN

Analog Digital

PSTN

Cisco CallManager Express can act as the PSTN gateway as well as manage the IP Phones.

There are different types of connections to the PSTN, including digital and analog. The type of

connection depends on the density of connections that is needed, the technology that is

available in the region, the cost of the connections, and the interfaces that are present on the

router.


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PSTNGatewayFunction

CUE Cisco CallManager Express

1000

SIP

Step 2AnSCCP messagecauses the IPPhone to ring.

Step 1A call arrives from thePSTN that maps through DID tothe IP Phone whose extensionis 1000.

Step 3No answeroccurs within theset time value.

Step 4Cisco CallManagerExpress uses SIP to set up acall to the CUE module.

Step 5The call is set up, andvoice flows between the CUEand the PSTN gateway functionof the router.

PSTN

Cisco CallManager Express and CUE interact when Cisco CallManager Express determines

that a call needs to go either to voice mail or to the automated attendant. The slide shows a call

from the PSTN being forwarded to voice mail using the following steps:

Step 1 A call arrives from the PSTN and, based on the called number, is mapped through

the use of direct inward dialing (DID) to an internal extension of 1000.

Step 2 Cisco CallManager Express sends an SCCP message to the IP Phone and causes the

IP Phone to ring.

Step 3 The timeout value for no answer to a forwarded call is exceeded, so Cisco

CallManager Express follows the forwarding instructions and forwards the call to

the CUE voice-mail pilot number.

Step 4 A session initiation protocol (SIP) message is sent to the CUE modules IP address

to set up a voice connection using one of the virtual voice ports.

Step 5 The CUE module has a free virtual voice port and answers the call via an SIP

message that goes back to Cisco CallManager Express. Two unidirectional RTP

streams are created between the PSTN gateway function of the router and CUE.


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WAN


H.323

H.323

PSTN Gateway andIP-to-IP Gateway

Functionality

SIP

PSTN

PSTN WAN

H.323Cisco CallManager

Express Cluster

If one Cisco CallManager Express system needs to set up a call to an IP Phone that is under the

control of another Cisco CallManager Express system, then the H.323 protocol needs to be

used between the Cisco CallManager Express systems. This configuration allows for many

different deployments of Cisco CallManager Express to be integrated together through an IP-

based WAN link.

The PSTN gateway function can be performed on the Cisco CallManager Express router or on

a separate standalone gateway. If a separate PSTN gateway is used, the additional functionality

of an IP-to-IP gateway can also be run on the router. This would enable the ability to translate

between H.323 and SIP.

Note A local PSTN is needed for each site for, at the very least, 9-1-1 emergency calls.


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LicensingThis topic describes the licensing for Cisco CallManager Express and CUE.


Licensing

Licensing for Cisco CallManager Express

Capable IOS image

Feature license for number of phones

Seat license per phone up to 240

Licensing for CUE

License for 12 mailboxes is included.

Additional licenses can be purchased for up to 100mailboxes total on the NM-CUE and NM-CUE-EC.

Additional licenses can be purchased for up to 50mailboxes total on the AIM-CUE.

Both Cisco CallManager Express and CUE have licensing requirements. For Cisco

CallManager Express, first a capable IOS image must be installed on the router, then the proper

feature license must be purchased. The feature license defines how many phones will be

controlled with the Cisco CallManager Express software. The various feature licenses are as

follows:

Feature License FL-CCME-SMALL (up to 24 users)

Feature License FL-CCME-36 (up to 36 users)

Feature License FL-CCME-MEDIUM (up to 48 users)








In addition to the feature license, each analog phone controlled by an ATA and each IP Phone

requires a seat license. The Cisco CallManager Express seat license is fully transferable to a

Cisco CallManager seat license.

There are 12 licensed user mailboxes included with the CUE module when it is ordered. If

more than 12 mailboxes are needed or desired, a new license file must be installed on the CUE

module.


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ISR Bundles

Offer savings and ease of ordering whencompared with ordering each of the componentsseparately.

Have flexible base package with option to addadditional service modules to provide customerwith complete solution.

Include IOS SP Services for voice gatewayservices and features.

Can be easily upgraded.

Include DSP modules to support PSTN-to-IPconnectivity.

Allow country-specific PSTN analog or digitalmodule to meet customer needs.

Include Cisco IP Communications featureslicense.

Offer flexibility to choose appropriate CUE modulefor voice mail.

Cisco offers a broad choice of IP communications solutions for growing businesses. For

businesses with a need for secure IP data routing with full-service voice capabilities, the Cisco

CallManager Express Bundles offer an affordable entry point into Cisco IP Communications.

These turnkey communications solutions support up to 240 phones and deliver feature-rich call

processing with integrated routing and switching, as well as optional voice mail and automated

attendant.

Small businesses can expect to realize the following returns on their Cisco CallManager

Express Bundles investment:

Cost savings and productivity enhancements: The Cisco CallManager Express Bundlesare an affordable entry point into a converged IP environment that delivers cost savings and

productivity enhancements.

Investment protection:The Cisco CallManager Express Bundles are cost-effective, and

they integrate with existing legacy voice investments while allowing you to migrate to a

Cisco IP Communications system.

Ease of management:The bundle components are integrated within a single chassis,

resulting in turnkey installation and streamlined system management with a common GUI.

Growth:Designed to respond to your dynamic business needs, the Cisco CallManager

Express Bundles can be easily upgraded to support advanced voice applications and

additional users. The complete portfolio of the Cisco IP Communications Solution scales to

support up to 30,000 devices.Support:With an excellent track record in supporting mission-critical voice applications,

Cisco and its certified partners provide full life-cycle support to deliver the Cisco

CallManager Express Bundles for a maximum return on investment.


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Cisco offers a range of bundles tailored to meet the needs of your business. Each bundle

includes a Cisco IP access router for secure data routing, Cisco CallManager Express software

to support IP telephony, Cisco IOS SP Services software for voice gateway services, digital

signal processor (DSP) chips for PSTN calls, and memory. CUE may be added to the bundle in

order to have voice mail and automated attendant capabilities. The base Cisco CallManager

Express Bundles are designed to meet the diverse needs of businesses worldwide.

It is necessary to add the country-specific digital or analog trunk interfaces that are required toconnect to the PSTN or host PBX. To complete the solution, add Cisco IP Phones and Cisco

Catalyst data switches that support inline power.

The various bundles include the following SKUs:

2801-CCME/K9 !2801-V router, DSP resources for 8 calls, 24 Cisco CallManager

Express seats, and IOS SP Services







3825-CCME/K9 !3825-V/K9 router, DSP resources for 64 calls, 168 Cisco CallManager

Express seats, and IOSSP Services

3845-CCME/K9 !3845-V/K9 router, DSP resources for 64 calls, 240 Cisco CallManager



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Release Compatibility

16161616NM-CUE-EC: # of Ports

100100100100NM-CUE: Hours ofStorage

8

100

20

100

100 Mailboxes

8

100

15

50

50 Mailboxes

100100NM-CUE-EC: Hours ofStorage

105General DeliveryMailboxes

NM-CUE: # of Ports

Personal Mailboxes

Feature

2512

88

25 Mailboxes12 Mailboxes

There are four CUE license levels available on the NM-CUE and NM-CUE-EC. The hardware

associated with CUE (NM-CUE and AIM-CUE) must be purchased with an accompanying

license. Hardware and software are packaged together. Mailbox licenses are purchased

separately with the exception of the 12-mailbox license level that is included in the price of the

hardware-software bundle. Therefore, a minimum of 12 mailboxes must be ordered with each

CUE purchase.

CUE license files, such as Cisco IOS software, can be downloaded from http://cisco.comand

installed on any number of systems for which a license was purchased without change to the

file itself. When a license is purchased or when software from Cisco is used, or both, a

contractual obligation is created. The subscriber must abide by the terms spelled out in thelicense agreement, including prohibitions regarding unauthorized replication of the software

and modification to the mailbox level of the license.

The capacity limitations on ports, subscribers, and mailboxes depend on whether CUE is

running on a network module or an advanced integration module and is controlled by the

license that is installed on the CUE application.


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Release Compatibility (Cont.)

Not Supported4*44AIM-CUE, 1GB, 2600XMand 2691: # of Ports

Not Supported4*44AIM-CUE, 512 MB,2600XM and 2691: # ofPorts

Not Supported6*66AIM-CUE, 512 MB, 2800,3700, and 3800: # ofPorts

Not Supported888AIM-CUE, 512 MB: Hoursof Storage

Not Supported

Not Supported

100 Mailboxes

6

14

50 Mailboxes

1414AIM-CUE, 1GB: Hours ofStorage

66AIM-CUE, 1GB, 2800,3700 and 3800 : # ofPorts

Feature 25 Mailboxes12 Mailboxes

*Not recommended because of port blocking and mailbox size

limitations

There are three CUE license levels available with the AIM-CUE in the 512-MB model and

three license levels available with the AIM-CUE in the 1-GB model. The use of the 50-mailbox

license is discouraged when using the 512-MB model because of port and storage limitations.

The 50-port license is appropriate when using the 1-GB model installed in a 2800, 3700, or

3800 platform.

When the advanced integration module is located in the chassis of a 2600XM series or 2691

router, it is limited to a maximum of four simultaneous ports at any one time. This presents

some port blocking issues that may be manifested when the number of mailboxes approaches

the upper limit of 50.


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SummaryThis topic summarizes the key points discussed in this lesson.


Summary

Cisco CallManagerExpress is an optional featureof CiscoIOS software and is available on a widerange of Cisco access routers that supportasmany as 240 phones.

Cisco CallManagerExpress provides callprocessing for IP Phones using SCCP.

CUE provides voice mail and automated attendantfor the small office or branch office.

CUE is fully integrated into Cisco 2600XM, 2691,2800, 3700, and 3800 series access routers.


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Lesson 2

Explaining DifferencesBetween Traditional Telephony

and VoIP

OverviewThis lesson explains the differences between traditional voice and Voice over IP (VoIP). This

includes a discussion of traditional telephony, pulse code modulation (PCM) theory, and the

basics of voice digitization. It also includes a discussion of the various compression schemes

that are used to transport voice using less bandwidth, using coder-decoder attributes, and

encapsulating voice in IP packets. In addition, the use of compressed Real-Time Transport

Protocol (cRTP) headers, including when and when not to use them, is discussed.

Objectives

Upon completing this lesson, you will be able to explain the differences between traditional

voice and VoIP. This includes being able to meet these objectives:

Identify the components, processes, and features of traditional telephony networks that

provide end-to-end call functionality

Identify the steps for converting analog signals to digital signals and the steps for

converting digital signals to analog signals; state the purpose of the Nyquist theorem;

explain quantization

Explain voice compression and coder-decoder standards; name two types of voice

compression techniques; list three common voice compression standards and their

bandwidth requirements

Describe the functions of RTP and RTCP as they relate to a VoIP network; describe how IP

voice headers are compressed using cRTP and how header size is reduced in order to

efficiently carry voice across the network using VoIP protocols and cRTP


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Traditional TelephonyThis topic introduces the components of traditional telephony networks. It describes how

central office (CO) switches function and how they make switching decisions, and it explores

PBX and key telephone system functionality in environments today. The topic also discusses

the three call-signaling types: supervisory, address, and informational.


Basic Components of a Telephony Network

A number of components must be in place for an end-to-end call to succeed. These components

are shown in the figure and include the following:

Edge devices

Local loops

Private or CO switches

Trunks

Edge Devices

The two types of edge devices that are used in a telephony network include:

Analog telephones:Analog telephones are most common in home, small business, and

small office, home office (SOHO) environments. A direct connection to the public

switched telephone network (PSTN) is usually made by using analog telephones.

Proprietary analog telephones are occasionally used in conjunction with a PBX. These

phones provide additional functions, such as speakerphone, volume control, PBX message-

waiting indicator, call on hold, and personalized ringing.

Digital telephones:Digital telephones contain hardware to convert analog voice into a

digitized stream. Larger corporate environments with PBXs generally use digital

telephones. Digital telephones are typically proprietary, that is, they work with the PBX or

key system of that vendor only.


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Local Loops

A local loop is the interface to the telephone company network. Typically, it is a single pair of

wires that carry a single conversation. A home or small business may have multiple local loops.

Private or CO Switches

The CO switch terminates the local loop and handles signaling, digit collection, call routing,call setup, and call teardown.

A PBX switch is a privately owned switch located at the customers site. A PBX typically

interfaces with other components to provide additional services, such as voice mail.

Trunks

The primary function of a trunk is to provide the path between two switches. There are several

common trunk types, including:

Tie trunk:A dedicated circuit that connects PBXs directly

CO trunk:A direct connection between a local CO and a PBX

Interoffice trunk:A circuit that connects two local telephone company COs


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Central Office Switches

The figure shows a typical CO switch environment. The CO switch terminates the local loop

and makes the initial call-routing decision.

The call-routing function forwards the call to one of the following:

Another end-user telephone if it is connected to the same CO

Another CO switch

A tandem switch

The CO switch enables the telephone to work with the following components:Battery:The battery is the source of power to both the circuit and the telephone !it

determines the status of the circuit. When the handset is lifted to let current flow, the

telephone company provides the source that powers the circuit and the telephone. Because

the telephone company powers the telephone from the CO, electrical power outages should

not affect the basic telephone.

Note Some telephones, such as cordless telephones, require a supplementary power source that

the subscriber supplies. Some cordless telephones may lose function during a power

outage.

Current detector:The current detector monitors the status of a circuit by detectingwhether it is open or closed. See the table "Current Flow in a Typical Telephone.#


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Current Flow in a Typical Telephone

Handset Circuit Current Flow

On cradle On hook/open circuit No

Off cradle Off hook/closed circuit Yes

Dial tone generator:When the digit register is ready, the dial tone generator produces a

dial tone to acknowledge the request for service.

Digit register:The digit register receives the dialed digits.

Ring generator:When the switch detects a call for a specific subscriber, the ring generator

alerts the called party by sending a ring signal to that subscriber.

You must configure a PBX connection to a CO switch that matches the signaling of the CO

switch. This configuration ensures that the switch and the PBX can detect on hook, off hook,

and dialed digits coming from either direction.

CO Switching SystemsSwitching systems provide three primary functions:

Call setup, routing, and teardown

Call supervision

Customer IDs and telephone numbers

CO switches switch calls between locally terminated telephones. If a call recipient is not locally

connected, the CO switch decides where to send the call based on its call-routing table. The call

then travels over a trunk to another CO or to an intermediate switch that may belong to an

inter-exchange carrier (IXC). Although intermediate switches do not provide a dial tone, they

act as hubs to connect other switches and provide interswitch call routing.

PSTN calls are traditionally circuit-switched, which guarantees end-to-end path and resources.

Therefore, as the PSTN sends a call from one switch to another, the same resource is associated

with the call until the call is terminated.


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What Is a PBX?

A PBX is a smaller, privately owned version of the CO switches that are used by telephone

companies.

In a corporate environment, where large numbers of staff need access to each other and to the

outside, individual telephone lines are not economically viable. Most businesses have a PBX

telephone system, a key telephone system, or Centrex service. Large offices, with more than 50

telephones or handsets, choose a PBX to connect users, both in-house and to the PSTN.

PBXs come in a variety of sizes, typically from 20 to 20,000 stations. The selection of a PBX is

important to most companies because a PBX has a typical life span of seven to ten years.

All PBXs offer a standard, basic set of calling features. Optional software provides additional

capabilities.

The figure illustrates the internal components of a PBX: it connects to telephone handsets using

line cards and to the local exchange using trunk cards.

A PBX has three major components:

Terminal interface:The terminal interface provides the connection between terminals and

PBX features that reside in the control complex. Terminals can include telephone handsets,

trunks, and lines. Common PBX features include dial tone and ringing.

Switching network:The switching network provides the transmission path between

two or more terminals in a conversation, such as when two telephones within an office

communicate over the switching network.

Control complex:The control complex provides the logic, memory, and processing for

call setup, call supervision, and call disconnection.


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What Is a Key System?

Small organizations and branch offices often use a key telephone system because a PBX has

functionality and extra features that they may not require. For example, unlike the central

answering position that is required for a PBX, a key system enables small businesses to have

distributed answering from any telephone.

Today, key telephone systems are either analog or digital and are microprocessor-based. Key

systems are typically installed in offices with 30 to 40 users, but can be scaled to support more

than 100 users.

A key system has three major components:

Key service unit:A key service unit (KSU) holds the system switching components,

power, intercom, line and station cards, and system logic.

System software:System software provides the operating system and calling-feature

software.

Telephones (instruments or handsets):Telephones allow the user to choose a free line

and dial out, usually by pressing a button on the telephone.


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Basic Call Setup

Call signaling, in its most basic form, is the capacity of a user to communicate a need for

service to a network. The call-signaling process requires the ability to detect a request for

termination of service, send addressing information, and provide progress reports to the

initiating party. This functionality corresponds to the three call-signaling types: supervisory,

address, and informational.

The figure shows the three major steps in an end-to-end call. These steps include:

Step 1 Local signaling $originating side

The user signals the switch by going off hook and sending dialed digits throughthe local loop.

Step 2 Network signaling

The switch makes a routing decision and signals the next, or terminating, switch

through the use of setup messages sent across a trunk.

Step 3 Local signaling $terminating side

The terminating switch signals the call recipient by sending ringing voltage through

the local loop to the recipient telephone.


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PCM TheoryThis topic describes the process of converting analog signals to digital signals and converting

digital signals back to analog signals. The topic also describes the Nyquist theorem, which is

the basis for digital signal technology, and explains quantization and its techniques.


Digitizing Analog Signals

1. Sample the analog signal regularly.

2. Quantize the sample.

3. Encode the value into a binary expression.

4. (Optional) Compress the samples to reducebandwidth (multiplexing).

Digitizing speech was a project first undertaken by the Bell System in the 1950s. The original

purpose of digitizing speech was to deploy more voice circuits with a smaller number of wires.

This evolved into the T1 and E1 transmission methods of today.

To convert an analog signal to a digital signal, you must perform these steps:

Note The last step is optional.


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Analog to Digital Signal Conversion

Step Procedure Description

1. Sample the analog signal regularly. The sampling rate must be two times the highestfrequency in order to produce playback that appearsneither choppy nor too smooth.

2. Quantize the sample. Quantization consists of a scale made up of eightmajor divisions, or chords. Each chord is subdividedinto 16 equally spaced steps. The chords are notequally spaced, but are actually finest near theorigin. Steps are equal within the chords, butdifferent when they are compared between thechords. Finer graduations at the origin result in lessdistortion for low-level tones.

3. Encode the value into 8-bit digitalform.

PBX output is a continuous analog voice waveform.T1 digital voice is a snapshot of the wave, encodedin ones and zeros.

4. (Optional) Compress the samplesto reduce bandwidth.

Although not essential to the conversion of analogsignals to digital, signal compression is widely usedto reduce bandwidth.

Three components in the analog-to-digital conversion process include:

Sampling:Sample the analog signal at periodic intervals. The output of sampling is a pulse

amplitude modulation (PAM) signal.

Quantization:Match the PAM signal to a segmented scale. This scale measures the

amplitude (height) of the PAM signal and assigns an integer number to define that

amplitude.

Encoding:Convert the integer base-10 number to a binary number. The output of encoding

is a binary expression in which each bit is either a 1 (pulse) or a 0 (no pulse).

This three-step process is repeated 8000 times per second for telephone voice channel service.

Use the optional fourth step!compression!to save bandwidth. This optional step allows a

single channel to carry more voice calls.

Note The most commonly used method for converting analog to digital is PCM.

Basic Voice Encoding: Converting Digital to Analog

After the receiving terminal at the far end receives the digital PCM signal, it must convert the

PCM signal back into an analog signal.

The process of converting digital signals back into analog signals includes two parts, decoding

and filtering:

Decoding:The received 8-bit word is decoded to recover the number that defines the

amplitude of that sample. This information is used to rebuild a PAM signal of the original

amplitude. This process is simply the reverse of the analog-to-digital conversion.

Filtering:The PAM signal passes through a properly designed filter, which reconstructs

the original analog wave form from its digitally-coded counterpart.


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Nyquist Theorem

The Nyquist Theorem

Digital signal technology is based on the premise stated in the Nyquist theorem: when a signal

is instantaneously sampled at the transmitter in regular intervals and has a rate of at least twice

the highest channel frequency, then the samples will contain sufficient information to allow an

accurate reconstruction of the signal at the receiver.

Example

Whereas the human ear can sense sounds from 20 to 20,000 Hz and speech encompasses

sounds from about 200 to 9000 Hz, the telephone channel was designed to operate at about300 to 3400 Hz. This economical range carries enough fidelity to allow callers to identify

the party at the far end and sense their mood. Nyquist decided to extend the digitization to

4000 Hz, to capture higher-frequency sounds that the telephone channel may deliver.

Therefore, the highest frequency for voice is 4000 Hz, or 8000 samples per second, that is,

one sample every 125 microseconds.

If every sample is encoded in 8 bits, this works out to be 8000 samples a second times 8 bits per

sample. This results in a digital voice conversation requiring 64,000 bits per second. The

original digital data circuits that carried digital voice are known as DS0s and sized at 64,000

bits per second.


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Quantization

Quantization

Quantization involves dividing the range of amplitude values that are present in an analog

signal sample into a set of discrete steps that are closest in value to the original analog signal.

Each step is assigned a unique digital code word.

The figure depicts quantization. In this example, the x-axis is time and the y-axis is the voltage

value (the PAM).

The voltage range is divided into 16 segments (0 to 7 positive and 0 to 7 negative). Starting

with segment 0, each segment has fewer steps than the previous segment, which reduces thenoise-to-signal ratio and makes it uniform. This segmentation also corresponds closely to the

logarithmic behavior of the human ear. If a noise-to-signal ratio problem exists, it is resolved

by using a logarithmic scale to convert PAM to PCM.


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Quantization Techniques

Linear

! Uniform quantization

Logarithmic quantization

! Compands the signal

! Provides a more uniform signal-to-noise ratio

Two methods

! a-law (most countries)

! mu-law (Canada, United States, and Japan)

Linear sampling of analog signals causes small-amplitude signals to have a higher noise-to-

signal ratio!and therefore poorer quality!than larger-amplitude signals. The Bell System

developed the mu-lawmethod of quantization, which is widely used in North America. The

International Telecommunication Union (ITU) modified the original mu-law method and

createda-law,which is used in countries outside North America.

By allowing smaller step functions at lower amplitudes rather than higher amplitudes, mu-law

and a-law provide a method of reducing the noise-to-signal method. Both mu-law and a-law

"compand#the signal; that is, they both compress the signal for transmission, then expand the

signal back to its original form at the other end.

Using mu-law and a-law results in a more accurate value for smaller amplitudes and uniform

signal-to-noise quantization ratio across the input range.

Both mu-law and a-law are linear approximations of a logarithmic input-output relationship.

They both generate 64-kbps bit streams using 8-bit code words to segment and quantize levels

within segments.

The difference between the original analog signal and the assigned quantization level is called

quantization error, which is the source of distortion in digital transmission systems.

Quantization error is any random disturbance or signal that interferes with the quality of the

transmission or the signal itself.

Note For communication between a mu-law country and an a-law country, the mu-law country

must change its signaling to accommodate the a-law country.


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Coder-DecoderThis topic describes two types of speech-coding schemes, waveform and source coding, and

compares G.729 and G.729a compression.


Waveform algorithms

! PCM

! ADPCM

Source algorithms

! LD-CELP

! CS-ACELP

Voice-Compression Techniques

There are two voice-compression techniques.

Waveform algorithms (coders) function as follows:

!Coders take sample analog signals at the rate of 8000 times per second.

!Coders use predictive differential methods to reduce bandwidth, which reduction

strongly impacts voice quality.

!Coders do not take advantage of speech characteristics.

Source algorithms function as follows:

!Voice coders (vocoders) convert analog speech into digital speech, using a specific

compression scheme that is optimized for coding human speech.

!Vocoders take advantage of speech characteristics.

!Codebooks store specific predictive waveshapes of human speech. They match the

speech, encode the phrases, decode the waveshapes at the receiver by looking up the

codedphrase, and match the coded phrase to the stored waveshape in the receiver

codebook.


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PCM

! Waveform coding scheme

ADPCM

! Waveform coding scheme

! Adaptive: automatic companding

! Differential: changes encoded betweensamples only

ITU standards:

! G.711 rate: 64 kbps = (2 x 4 kHz) x 8 bits/sample




Example: Waveform Compression

Standard PCM is known as ITU standard G.711.

Adaptive differential PCM (ADPCM) coders, like other waveform coders, encode analog voice

signals into digital signals to predict future encodings by looking at the immediate past. The

adaptive feature of ADCPM reduces the number of bits per second that the PCM method

requires to encode voice signals.

ADPCM does this by taking 8000 samples per second of the analog voice signal and turning

them into a linear PCM sample. ADPCM then calculates the predicted value of the next sample,

based on the immediate past sample, and encodes the difference. The ADPCM process

generates 4-bit words, therefore generating 16 specific bit patterns.

The ADPCM algorithm from the ITU Telecommunication Standardization Sector (ITU-T)

(formerly the CCITT) transmits all 16 possible bit patterns. The ADPCM algorithm from the

American National Standards Institute (ANSI) uses 15 of the 16 possible bit patterns. The

ANSI ADPCM algorithm does not generate a "0000#pattern.

The ITU standards for compression are as follows:

G.711 rate:64 kbps = (2 * 4 kHz) * 8 bits per sample





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CELP

! Hybrid coding scheme

High-quality voice at low bit rates; processorintensive

G.728: LD-CELP ! 16 kbps

G.729: CS-ACELP! 8 kbps

! G.729A variant! 8 kbps, less processor-intensive, allows more voice channels encodedper digital signal processor

! Annex-B variant ! VAD and CNG

Example: Source Compression

Code-excited linear prediction (CELP) compression transforms analog voice signals as follows:

The input to the coder is converted from an 8-bit PCM to a 16-bit linear PCM sample.

A codebook uses feedback to continuously learn and predict the voice waveform.

A white noise generator excites the coder.

The mathematical result (recipe) is sent to the far-end decoder for synthesis and generation

of the voice waveform.

Low-delay CELP (LD-CELP) is similar to Conjugate Structure Algebraic Code Excited Linear

Prediction (CS-ACELP) (see next paragraph) except:

LD-CELP uses a smaller codebook and operates at 16 kbps to minimize look-ahead delay,

keeping it to 2 to 5 ms.

The 10-bit codeword is produced from every five speech samples from the 8-kHz input.

Four of these 10-bit codewords are called a subframe, which takes approximately 2.5 ms to

encode.

Two of these subframes are combined into a 5-ms block for transmission. CS-ACELP is a

variation of CELP that performs these functions:

Codes on 80-byte frames, which take approximately 10 ms to buffer and process.

Adds a look-ahead of 5 ms. A look-ahead is a coding mechanism that continuouslyanalyzes, learns, and predicts the next waveshape.

Adds noise reduction and pitch-synthesis filtering to processing requirements.

Example

The Annex B variant adds voice activity detection (VAD) in strict compliance with G.729b

standards. When this coder-decoder (codec) variant is used, VAD is not tunable for music

threshold. However, when Cisco VAD is configured, music threshold is tunable.


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G.729 and G.729A Comparison

Both are ITU standards.

Both are 8-kbps CS-ACELP.

G.729 is more complex and processor intensive.

G.729 is slightly higher quality than G.729A.

Compression delay is the same (10 to 20 ms).

Annex-B variant can be applied to either.

G.729, G.729 Annex A (G.729a), G.729 Annex B (G.729b), and G.729a Annex B (G.729ab)

are variations of CS-ACELP.

There is little difference between the ITU recommendations for G.729 and G.729a. All of the

platforms that support G.729 also support G.729a.

G.729 is the compression algorithm that Cisco uses for high-quality 8-kbps voice. When G.729

is properly implemented, it sounds as good as the 32-kbps ADPCM. G.729 is a high-

complexity, processor-intensive compression algorithm that monopolizes processing resources.

Although G.729a is also an 8-kbps compression, it is not as processor-intensive as G.729. It is amedium-complexity variant of G.729 with slightly lower voice quality. The quality of G.729a

is not as high as G.729 and is more susceptible to network irregularities such as delay,

variation, and "tandeming.#Tandeming causes distortion that occurs when speech is coded,

decoded, then coded and decoded again, much like the distortion that occurs when a videotape

is repeatedly copied.


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Example

On Cisco IOS gateways, you must use the variant (G.729 or G.729a) that is related to the codec

complexity configuration on the voice card. This variant does not show up explicitly in the

Cisco IOS command-line interface (CLI) codec choice. For example, the CLI does not display

g729r8(alpha code) as a codec option. However, if the voice card is defined as medium-

complexity, then the g729r8option is the G.729a codec.

G.729b is a high-complexity algorithm, and G.729ab is a medium-complexity variant of

G.729b with slightly lower voice quality. The difference between the G.729 and G.729b codecs

is that the G.729b codec provides built-in Internet Engineering Task Force (IETF) VAD and

comfort noise generation (CNG).

The following G.729 codec combinations interoperate:

G.729 and G.729a

G.729 and G.729

G.729a and G.729a

G.729b and G.729ab

G.729b and G.729b

G.729ab and G.729ab


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Encapsulating Voice in IP PacketsThis topic describes the functions of RTP and RTP Control Protocol (RTCP) as they relate to

the VoIP network. The topic also describes how IP voice headers are compressed using cRTP,

and it describes when to use cRTP.


Real-Time Transport Protocol

Provides end-to-end network functions anddelivery services for delay-sensitive, real-timedata, such as voice and video

Works with queuing to prioritize voice traffic overother traffic

Services include:

!

Payload type identification! Sequence numbering

! Time-stamping

! Delivery monitoring

RTP provides end-to-end network transport functions intended for applications that are

transmitting real-time data, such as audio and video. The functions include payload type

identification, sequence numbering, time-stamping, and delivery monitoring.

RTP typically runs on top of User Datagram Protocol (UDP) to utilize the multiplexing and

checksum services of that protocol. Although RTP is often used for unicast sessions, it is

primarily designed for multicast sessions. In addition to defining the roles of sender and

receiver, RTP also defines the roles of translator and mixer to support the multicast

requirements.

Example

RTP is a critical component of VoIP because it enables the destination device to reorder and

retime the voice packets before they are played out to the user. An RTP header contains a time

stamp and a sequence number, which allows the receiving device to buffer and remove jitter

and latency by synchronizing the packets to play back a continuous stream of sound. RTP

uses sequence numbers to order the packets only. RTP does not request retransmission if apacket is lost.

For more information on RTP, refer to RFC 1889.


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Monitors the quality of the data distribution andprovides control information

Provides feedback on current network conditions

Allows hosts that are involved in an RTP sessionto exchange information about monitoring andcontrolling the session

Provides a separate flow from RTP for UDPtransport use

Real-Time Transport Control Protocol

RTCP monitors the quality of the data distribution and provides control information. RTCP

provides the following feedback on current network conditions:

RTCP provides a mechanism for hosts involved in an RTP session to exchange information

about monitoring and controlling the session. RTCP monitors the quality of such elements

as packet count, packet loss, delay, and inter-arrival jitter. RTCP transmits packets as a

percentage of session bandwidth, but at a specific rate of at least every 5 seconds.

The RTP standard states that the Network Time Protocol (NTP) time stamp is based on

synchronized clocks. The corresponding RTP time stamp is randomly generated and based

on data-packet sampling. Both NTP and RTP are included in RTCP packets by the sender

of the data.

RTCP provides a separate flow from RTP for transport use by UDP. When a voice stream

is assigned UDP port numbers, RTP is typically assigned an even-numbered port and

RTCP is assigned the next odd-numbered port. Each voice call has four ports assigned:

RTP plus RTCP in the transmit direction and RTP plus RTCP in the receive direction.

Example

Throughout the duration of each RTP call, the RTCP report packets are generated at least every

5 seconds. In the event of poor network conditions, a call may be disconnected because of high

packet loss. When using a packet analyzer to view packets, a network administrator can check

information in the RTCP header that includes packet count, octet count, number of packets lost,

and jitter. The RTCP header information helps in determining why calls are disconnected.


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RTP Packet Components

When speech samples are framed every 20 ms in a packet voice environment that is using

G.729, a payload of 20 bytes is generated. Without cRTP, the total packet size includes the

following components:

IP header (20 bytes)

UDP header (8 bytes)

RTP header (12 bytes)

Payload (20 bytes)

The header is twice the size of the payload: IP/UDP/RTP (20 + 8 + 12 = 40 bytes) versus the

payload (20 bytes). When generating packets every 20 ms on a slow link, the header consumes

a large portion of bandwidth.

As shown in the previous figure, RTP header compression reduces the header to 2 bytes. Now,

instead of the header being twice the size of the payload, the payload is ten times the size of the

compressed header.


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Congested WAN links

Slow links (less than 2 Mbps)

Bandwidth on a WAN interface that needs to be conserved

When to Use RTP Header Compression

You must configure cRTP on a specific serial interface or subinterface if you have any of these

conditions:

Congested WAN links

Slow links (less than 2 Mbps)

Bandwidth on a WAN interface that needs to be conserved

Compression works on a link-by-link basis and must be enabled for each link that has any of

those conditions. You must enable compression on both sides of the link for proper results.

Enabling compression on both ends of a low-bandwidth serial link can greatly reduce thenetwork overhead if there is a significant volume of RTP traffic on that slow link.

Note Compression adds to processing overhead. You must check resource availability on each

device prior to turning on RTP header compression.

Example

If you want the router to compress RTP packets, use the ip rtp header-compression command.

The ip rtp header-compression command defaults to active mode when it is configured.

However,this command provides a passive mode setting in instances where you want the

router to compress RTP packets onlyif it has received compressed RTP on that interface. When

applying to a Frame Relay interface, use the frame-relay ip rtp header-compressio

Documents

KnowledgeNet IP Telephony Express (IPTX) 2.0 Student Guide.pdf