Upon completing this course, you will be able to meet these objectives: Describe the similarities and differences between traditional PSTN voice networks and IP telephony solutions Expla
Trang 1Copyright © 2004, Cisco Systems, Inc Student Guide i
CVOICE
Cisco Voice Over IP
Version 4.2
Student Guide
Trang 2ii Enterprise Voice Over Data Design (EVODD) v3.3 Copyright © 2004, Cisco Systems, Inc
Copyright 2004, Cisco Systems, Inc All rights reserved
Cisco Systems has more than 200 offices in the following countries and regions Addresses, phone numbers, and fax numbers are listed on the Cisco Web site at www.cisco.com/go/offices
Argentina Australia Austria Belgium Brazil Bulgaria Canada Chile China PRC Colombia Costa Rica Croatia Czech Republic Denmark Dubai, UAE Finland France Germany Greece Hong Kong SAR Hungary India Indonesia Ireland Israel Italy Japan Korea Luxembourg Malaysia Mexico The Netherlands New Zealand Norway Peru Philippines Poland Portugal Puerto Rico Romania Russia Saudi Arabia Scotland Singapore Slovakia Slovenia South Africa Spain Sweden Switzerland Taiwan Thailand Turkey Ukraine United
Kingdom United States Venezuela Vietnam Zimbabwe Copyright 2004, Cisco Systems, Inc All rights reserved CCIP, the Cisco Powered Network mark, the Cisco Systems Verified logo, Cisco Unity, Follow Me Browsing, FormShare, Internet Quotient, iQ Breakthrough, iQ Expertise, iQ FastTrack, the iQ logo, iQ Net Readiness Scorecard, Networking Academy,
ScriptShare, SMARTnet, TransPath, and Voice LAN are trademarks of Cisco Systems, Inc.; Changing the Way
We Work, Live, Play, and Learn, Discover All Thats Possible, The Fastest Way to Increase Your Internet
Quotient, and iQuick Study are service marks of Cisco Systems, Inc.; and Aironet, ASIST, BPX, Catalyst,
CCDA, CCDP, CCIE, CCNA, CCNP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, the Cisco IOS logo, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Empowering the Internet Generation, Enterprise/Solver, EtherChannel, EtherSwitch, Fast Step, GigaStack, IOS, IP/TV, LightStream,
MGX, MICA, the Networkers logo, Network Registrar, Packet, PIX, Post-Routing, Pre-Routing, RateMUX, Registrar, SlideCast, StrataView Plus, Stratm, SwitchProbe, TeleRouter, and VCO are registered trademarks of Cisco Systems, Inc and/or its affiliates in the U.S and certain other countries
All other trademarks mentioned in this document or Web site are the property of their respective owners The use of the word partner does not imply a partnership relationship between Cisco and any other company
(0203R)
Trang 3Understanding Packetized Telephony Networks 1-25
Understanding IP Telephony Applications 1-41
Trang 4ii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Analog and Digital Voice Connections 2-1
Types of Trunk Signaling 2-17
E & M Signaling Types 2-21 Trunk Signal Types and Used by E & M 2-26
Undetstanding Analog-to-Digital Voice Encoding 2-43
Trang 5iii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Compresson Bandwidth Requiremens 2-58 Voice Quality Measurement 2-59
Understanding Signaling Systems 2-67
Understanding Fax and Modem over VoIP 2-93
Trang 6iv Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Voice Port Applications 3-5
Echo Cancellation Commands 3-49
Trang 7v Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Configuring Destination-Pattern Options 4-25
Matching Inbound Dial Peers 4-30 Matching Outbound Dial Peers 4-33
Understanding Special-Purpose Connections 4-61
Configuring Trunk Connections 4-67
Building a Scalable Numbering Plan 4-75
Trang 8vi Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Packet Loss, Delay, and Jitter 5-7
Understanding Gateways and Their Roles 5-21
Central and Remote Site 5-26 Determining Gateway Interconnection Requirements in a Service Provider
Encapsulating Voice in IP Packets 5-35
Calculating Bandwidth Requirements 5-51
Impact of Voice Samples and Packet Size on Bandwidth 5-55
Security and Tunneling Overhead 5-57 Specialized Encapsulations 5-58 Calculating the Total Bandwidth for a VoIP Call 5-59 Effects of VAD on Bandwidth 5-60
Trang 9vii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Understanding Security Implications 5-67
Translation Between Signaling and Call Control Models 6-8
Trang 10viii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
H.323 Call Establishment and Maintenance 6-42 Call Flows Without a Gatekeeper 6-45 Call Flows with a Gatekeeper 6-48 Multipoint Conferences 6-50 Call Flows with Multiple Gatekeepers 6-52 Survivability Strategies 6-54
Cisco Implementation of H.323 6-58 Configuring H.323 Gateways 6-59 Configuring H.323 Gatekeepers 6-62 Monitoring and Troubleshooting 6-64
Trang 11ix Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Survivability Strategies 6-129 Cisco Implementation of MGCP 6-131 Understanding Basics of Cisco CallManager 6-132
Comparing Call Control Models 6-151
Improving and Maintaining Voice Quality 7-1
Understanding VoIP Challenges 7-17
Trang 12x Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Understanding QoS and Good Design 7-37
Trang 13xi Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Verifying End-to-End Delay 7-84
Applying QoS in the Campus 7-91
Understanding QoS Tools in the WAN 7-111
Configuring QoS in the WAN 7-133
CiscoWorks QPM for QoS 7-152
Trang 14xii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Effects of Oversubscribing Bandwidth on Overall Voice Quality 7-164 CAC as Part of Call Control Services 7-165
Understanding Voice Bandwidth Engineering 7-195
Trang 15xiii Cisco Voice Over IP Copyright © 2004, Cisco Systems, Inc
Computer Telephony Integration 14 Collaborative Computing 16 Voice-Enabled Web Applications 17
Trang 16Course Goal and Objectives
This section describes the course goal and objectives
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.23
Course Goal
To provide an understanding
of converged voice and data networks as well as the
challenges faced by their various technologies
Cisco Voice over IP (CVOICE) v4.2
Trang 172 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Upon completing this course, you will be able to meet these objectives:
Describe the similarities and differences between traditional PSTN voice networks and IP telephony solutions
Explain the processes and standards for voice digitization, compression, digital signaling, and fax transport as they relate to VoIP networks
Configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, nonpacketized telephony equipment
Configure the call flows for POTS, VoIP, and default dial peers Describe the fundamentals of VoIP and identify challenges and solutions regarding its implementation
Compare centralized and decentralized call control and signaling protocols Describe specific voice quality issues and the QoS solutions used to solve them
Course Outline
The outline lists the modules included in this course
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.24
Course Outline
Introduction to Packet Voice Technologies
Analog and Digital Voice Connections
Configuring Voice Interfaces
Voice Dial Plans
Introduction to VoIP
VoIP Signaling and Call Control
Improving and Maintaining Voice Quality
Trang 18Copyright © 2004, Cisco Systems, Inc Course Introduction 3
Cisco Certifications
This topic lists the certification requirements of this course
Cisco provides three levels of general career certifications for IT professionals with several different tracks to meet individual needs Cisco also provides a variety of Cisco Qualified Specialist (CQS) certifications, which enable learners to demonstrate knowledge in specific technologies, solutions, or job roles In contrast to general certifications, each CQS certification
is focused on a designated area such as cable communications, voice, or security All CQS certifications are customized to meet current market needs They may also have special focused prerequisite requirements
There are many paths to Cisco certification, but only one requirementpassing one or more exams demonstrating knowledge and skill For details, go to
http://www.cisco.com/go/certifications
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.25
Course Certifications
Trang 194 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Learner Skills and Knowledge
This topic lists the course prerequisites
To benefit fully from this course, you must have these prerequisite skills and knowledge:
A working knowledge of LANs, WANs, and IP switching and routing Basic internetworking skills taught in the Interconnecting Cisco Network Devices (ICND)course, or its equivalent
Knowledge of traditional public switched telephone network (PSTN) operations and voice fundamentals
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.26
Prerequisite Learner Skills and Knowledge
WANS Basic Telephony Fundamentals
IP Switching Basic Internetworking Skills PSTN Operations and Technologies
CVOICE LANS
PBX Essentials
Trang 20Copyright © 2004, Cisco Systems, Inc Course Introduction 5
Learner Responsibilities
This topic discusses the responsibilities of the learners
To take full advantage of the information that is presented in this course, you must have
completed the prerequisite requirements
In class, you are expected to participate in all lesson exercises and assessments
In addition, you are encouraged to ask any questions that are relevant to the course materials
If you have pertinent information or questions concerning future Cisco product releases and product features, please discuss these topics during breaks or after class The instructor will answer your questions or direct you to an appropriate information source
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.27
Learner Responsibilities
Completeprerequisites
Introduceyourself
Ask questions
Trang 216 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
General Administration
This topic lists the administrative issues for the course
The instructor will discuss these administrative issues:
Sign-in process Starting and anticipated ending times of each class day Class breaks and lunch facilities
Appropriate attire during class Materials that you can expect to receive during class What to do in the event of an emergency
Location of the rest rooms How to send and receive telephone and fax messages
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.28
locations
Trang 22Copyright © 2004, Cisco Systems, Inc Course Introduction 7
Course Flow Diagram
This topic covers the suggested flow of the course materials
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.29
Course Flow Diagram
Course Introduction
Lunch
Configuring Voice Interfaces
Improving and Maintaining Voice Quality
A
M
PM
Introduction
to VoIP (Cont.)
Voice Dial Plans
Analog and Digital Voice Connections
Configuring Voice Interfaces (Cont.) Introductionto VoIP
VoIP Signaling and Call Control (Cont.)
Intro to Packet Voice Technologies
VoIP Signaling and Call Control
Improving and Maintaining Voice Quality (Cont.)
Analog and Digital Voice Connections (Cont.) Voice Dial Plans
(Cont.)
VoIP Signaling and Call Control (Cont.)
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 laboratory exercises The exact timing of the subject materials and labs depends on the pace of your specific class
Trang 238 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Icons and Symbols
This topic shows the Cisco icons and symbols used in this course
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.211
Cisco Icons and Symbols (Cont.)
Voice-Enabled ATM Switch
Voice-Enabled Communications Server
Multilayer Switch, With and Without Text and Subdued
Si Si
SC2200 Signaling Controller
Web Cluster
Camera PC/Video
PBX/
Switch
Data Center Switch, Reversed
Web Browser
Server
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.210
Cisco Icons and Symbols
Enabled Router
Voice-PBX (Small)
Network Cloud, Standard Color
Network Cloud, White
Phone
IP Phone
Phone 2
Generic Softswitch
Cisco PIX Firewall Right and Left
Trang 24Copyright © 2004, Cisco Systems, Inc Course Introduction 9
Learner Introductions
This is the point in the course where you introduce yourself
Prepare to share the following information:
Your name Your company
If you have most or all of the prerequisite skills
A profile of your experience What you would like to learn from this course
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.212
Learner Introductions
Your name
Your company
Skills and knowledge
Brief history
Objective
Trang 2510 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Trang 26Migration is a process that involves gradually phasing out old components and replacing them with new ones Many terms have been used to describe the technologies and applications for transporting voice in a converged packet network environment When designing a converged network, it is necessary to clearly define all requirements and understand the various options that are available
An important first step in designing a converged network is to understand the traditional telephony network and how it interfaces with voice components You must know, from the start, how legacy voice equipment is connected and its possible migration paths
The next step toward a good design is being knowledgeable about the components available for Packet Telephony Networks You should be aware of the difference between voice and data flow within the network and the tools for controlling voice calls Network requirements vary according to the location size Knowing the difference between campus, enterprise, and service provider environments is crucial in choosing the right components and technologies
This module provides an overview of the basic telephony functions and devices, including PBXs, switching functions, call signaling, and multiplexing techniques It also reviews the basic components of the Packet Telephony Network and identifies the different requirements in campus, enterprise, and service provider environments Together, these concepts and
techniques provide a solid introduction to the VoIP arena
Trang 271-2 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Identify the components, processes, and features
of traditional telephony networks that provide end-to-end call functionality
Describe two methods of call control used on voice and data networks and provide one protocol example for each
List five components or capabilities that are required to provide integrated voice and data services in campus LAN, enterprise, and service provider environments
Module Outline
The outline lists the components of this module
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-3
Module Outline
Understanding Traditional Telephony
Understanding Packetized Telephony Networks
Understanding IP Telephony Applications
Trang 28be supported and the processes that are necessary to ensure end-to-end call functionality
Objectives
Upon completing this lesson, you will be able to identify the components, processes, and features of traditional telephony networks that provide end-to-end call functionality This includes being able to meet these objectives:
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
Learner Skills and Knowledge
To benefit fully from this lesson, you must have these prerequisite skills and knowledge: General knowledge of telephony technology, including customer premises equipment (CPE) and the PSTN
Trang 291-4 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Outline
The outline lists the topics included in this lesson
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-2
Trang 30Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-5
Basic Components of a Telephony Network
This topic introduces the components of traditional telephony networks
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-3
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 central office (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 office/home office (SOHO), and small business environments Direct connection to the PSTN is usually made by using analog telephones Proprietary analog telephones are occasionally used in conjunction with a PBX These telephones 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, meaning that they work with the PBX or key system of that vendor only
Trang 311-6 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
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
Example: Telephony Components
The telephone installed in your home is considered an edge device because it terminates the service provided by your local telephone company PBXs or key systems installed in a business would also be considered edge devices The local loop is the pair of wires that come to your house to provide residential telephone service Trunks are the interconnections between
telephone switches They can be between private switches or telephone company switches
Trang 32Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-7
CO Switches
This topic describes how CO switches function and make switching decisions
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-4
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 makes the telephone 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 on the market offer additional features that require a supplementary power
source that the subscriber supplies; for example, cordless telephones Some cordless telephones may lose function during a power outage
Current detector: The current detector monitors the status of a circuit by detecting whether it is open or closed The table here describes current flow in a typical telephone
Trang 331-8 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
Current Flow in a Typical Telephone
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
Dial 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 Systems
Switching systems provide three primary functions:
Call setup, routing, and teardown Call supervision
Customer ID 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 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
Example: CO Switches
CO switches provide local service to your residential telephone The CO switch provides dial tone, indicating that the switch is ready to receive digits When you dial your phone, the CO switch receives the digits, then routes your call The call routing may involve more than one switch as the call progresses through the network
Trang 34Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-9
Private Switching Systems
In a corporate environment, where large numbers of staff need access to each other and the outside, individual telephone lines are not economically viable This topic explores PBX and key telephone system functionality in environments today
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-5
What Is a PBX?
A PBX is a smaller, privately owned version of the CO switches used by telephone companies 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, 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
Trang 351-10 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
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 For example, 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
Example: PBX Installations
PBX switches are installed in large business campuses to relieve the public telephone company switches from having to switch local calls When you call a coworker locally in your office campus, the PBX switches the call locally instead of having to rely on the public CO switch The existence of PBX switches also limits the number of trunks needed to connect to the telephone companys CO switch With a PBX installed, every office desktop telephone does not need its own trunk to the CO switch Rather, the trunks are shared among all users
Trang 36Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-11
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-6
What Is a Key System?
Small organizations and branch offices often use a key telephone system, because a PBX offers functionality and extra features that they may not require A key system offers small businesses distributed answering from any telephone, unlike the central answering position required for a PBX
Today, key telephone systems are either analog or digital and are microprocessor-based Key systems are typically used in offices with 30 to 40 users, but can be scaled to support over
100 users
A key system has three major components:
Key service unit: A key service unit (KSU) holds the system switching components, power supply, intercom, line and station cards, and the 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
Trang 371-12 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-7
Comparing Key Systems with PBXs
Larger companies use proprietary telephone networks with PBXs In a key telephone system, each telephone has multiple lines that allow users to access outside lines to their CO When a call comes into the company, a line or a key lights up on the telephone and indicates that a particular line is in use Users can call another extension or let another person know where to pick up a call by using an intercom function, such as an overhead paging system or
The main difference between a key telephone system and a hybrid telephone system is whether
a single-line telephone can access a single CO local loop or trunk (key telephone system) only,
or whether the single-line telephone can access a pool of CO local loops or trunks (hybrid telephone system)
Trang 38Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-13
Call Signaling
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 service and termination of service, send addressing information, and provide progress reports to the initiating party This functionality corresponds to the three call-signaling types discussed in this topic: supervisory, address, and informational signaling
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-8
Basic Call Setup
The figure shows the three major steps in an end-to-end call These steps include:
1 Local signaling originating side: The user signals the switch by going off hook and sending dialed digits through the local loop
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
3 Local signaling terminating side: The terminating switch signals the call recipient by sending ringing voltage through the local loop to the recipient telephone
Trang 391-14 Cisco Voice over IP (CVOICE) v4.2 Copyright © 2004, Cisco Systems, Inc
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-9
Supervisory Signaling
A subscriber and telephone company notify each other of call status with audible tones and an exchange of electrical current This exchange of information is called supervisory signaling There are three different types of supervisory signaling:
On hook: When the handset rests on the cradle, the circuit is on hook The switch prevents current from flowing through the telephone Regardless of the signaling type, a circuit goes
on hook when the handset is placed on the telephone cradle and the switch hook is toggled
to an open state This prevents the current from flowing through the telephone Only the ringer is active when the telephone is in this position
Off hook: When the handset is removed from the telephone cradle, the circuit is off hook The switch hook toggles to a closed state, causing circuit current to flow through the electrical loop The current notifies the telephone company equipment that someone is requesting to place a telephone call When the telephone network senses the off-hook connection by the flow of current, it provides a signal in the form of a dial tone to indicate that it is ready
Ringing: When a subscriber makes a call, the telephone sends voltage to the ringer to notify the other subscriber of an inbound call The telephone company also sends a ringback tone to the caller alerting the caller that it is sending ringing voltage to the recipient telephone Although the ringback tone sounds similar to ringing, it is a call-progress tone and not part of supervisory signaling
Note The ringing tone in the United States is 2 seconds of tone followed by 4 seconds of silence
Europe uses a double ring followed by 2 seconds of silence
Trang 40Copyright © 2004, Cisco Systems, Inc Introduction to Packet Voice Technologies 1-15
© 2004 Cisco Systems, Inc All rights reserved CVOICE v4.21-10
Dual tone multifrequency: Each button on the keypad of a touch-tone pad or push-button telephone is associated with a set of high and low frequencies On the keypad, each row of keys is identified by a low-frequency tone and each column is associated with a high-frequency tone The combination of both tones notifies the telephone company of the number being called, thus the term dual tone multifrequency (DTMF)
Pulse: The large numeric dial-wheel on a rotary-dial telephone spins to send digits to place
a call These digits must be produced at a specific rate and within a certain level of tolerance Each pulse consists of a break and a make, which are achieved by opening and closing the local loop circuit The break segment is the time during which the circuit is open The make segment is the time during which the circuit is closed The break-and-make cycle must correspond to a ratio of 60 percent break to 40 percent make
A governor inside the dial controls the rate at which the digits are pulsed; for example, when a subscriber calls someone by dialing a digit on the rotary dial, a spring winds When the dial is released, the spring rotates the dial back to its original position While the spring rotates the dial back to its original position, a cam-driven switch opens and closes the connection to the telephone company The number of consecutive opens and closes, or breaks and makes, represents the dialed digit