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  • 1_1_Traditional_Telephony.pdf

    • Traditional Telephony

      • Basic Components of a Telephony Network

      • CO Switches

      • Private Switching Systems

      • Call Signaling

      • Multiplexing Techniques

  • 1_2_Packetized_Telephony_Network.pdf

    • Packetized Telephony Networks

      • Benefits of Packet Telephony Networks

      • Call Control

      • Distributed vs. Centralized Call Control

      • Packet Telephony Components

      • Best-Effort Delivery of Real-Time Traffic

  • 1_3_IP_Telephony_Applications.pdf

    • IP Telephony Applications

      • Analog Interfaces

      • Digital Interfaces

      • IP Phones

  • 1_4_Analog_Voice_Basics.pdf

    • Analog Voice Basics

      • Local-Loop Connections

      • Types of Local-Loop Signaling

      • Supervisory Signaling

      • Address Signaling

      • Informational Signaling

      • Trunk Connections

      • Types of Trunk Signaling

      • E&M Signaling Types

      • Trunk Signal Types Used by E&M

      • Line Quality

      • Management of Echo

  • 1_5_Analog_to_Digital_Voice_Encoding.pdf

    • Analog-to-Digital Voice Encoding

      • Basic Voice Encoding: Converting Analog to Digital

      • Basic Voice Encoding: Converting Digital to Analog

      • The Nyquist Theorem

      • Voice Compression and Codec Standards

      • Compression Bandwidth Requirements

      • Voice Quality Measurement

  • 1_6_Signaling_Systems.pdf

    • Signaling Systems

      • CAS Systems: T1

      • CAS Systems: E1

      • CCS Systems

      • ISDN

  • 2_1_Requirements_of_Voice.pdf

    • Requirements of Voice in an IP Internetwork

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

      • Packet Loss, Delay, and Jitter

      • Consistent Throughput

      • Reordering of Voice Packets

      • Reliability and Availability

  • 2_2_Gate_and_Their_Roles.pdf

    • Gateways and Their Roles

      • Understanding Gateways

      • Guidelines for Selecting the Correct Gateway

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

      • Determining Gateway Interconnection Requirements in a Service Provider Environment

  • 2_3_Encapsulating_Voice.pdf

    • Encapsulating Voice in IP Packets

      • Major VoIP Protocols

      • RTP and RTCP

      • Reducing Header Overhead with CRTP

      • When to Use RTP Header Compression

  • 2_4_Calculating_Bandwidth.pdf

    • Calculating Bandwidth Requirements

      • Codec Bandwidths

      • Impact of Voice Samples and Packet Size on Bandwidth

      • Data Link Overhead

      • Security and Tunneling Overhead

      • Specialized Encapsulations

      • Calculating the Total Bandwidth for a VoIP Call

      • Effects of VAD on Bandwidth

  • 3_1_Overview_of_Cisco_CME.pdf

    • Overview of Cisco CME

      • What is Cisco CallManager Express?

      • How Does Cisco CallManager Express Work?

      • Licensing

  • 3_2_Differences_between.pdf

    • Differences between Traditional Telephony and VoIP

      • Traditional Telephony

      • PCM Theory

      • Basic Voice Encoding: Converting Digital to Analog

      • PCM Theory

      • Coder-Decoder

      • Encapsulating Voice in IP Packets

  • 3_3_Challenges_and_Solutions.pdf

    • Challenges and Solutions in VoIP

      • Challenges in VoIP

      • Bandwidth Requirements in VoIP

  • 3_4_CME_Features_Functionality.pdf

    • Cisco CME Features and Functionality

      • Supported Protocols and Integration Options

      • Cisco CallManager Express Requirements

      • Cisco CallManager Express Restrictions

  • 3_5_CME_Network_Parameters.pdf

    • Cisco CME Network Parameters

      • Auxiliary VLANs

      • Configuring Auxiliary VLANs

      • DHCP Service Setup

  • 3_6_IP_Phone_Registration.pdf

    • IP Phone Registration

      • Files

      • IP Phone Information

      • Download and Registration

  • 3_7_Ephonedn_and_Ephone.pdf

    • Ephone-dn and Ephone

      • Ephone-dn

      • Ephone

      • Type of Ephone-dns

      • Number of Ephone-dns

  • 3_8_CME_Files.pdf

    • Cisco CME Files

      • Cisco CME File

      • Bundled Cisco CME File

      • Individual Cisco CME Files

      • GUI Files

      • Cisco CME - TAPI Integration

      • Additional Files

  • 3_9_Initial_Phone_Setup.pdf

    • Initial Phone Setup

      • Setting Up Phones in a CME System

      • Automated Phone Setup

      • Partially Automated Phone Setup

      • Manual Phone Setup

      • Setup Tips

      • Optional Parameters

      • Router Configuration: Two Commands

      • Optional Parameters – Locale Parameters

      • Rebooting Cisco CallManager Express Phones

      • Setup Troubleshooting

      • Verifying Cisco CallManager Express Phone Configuration

  • 4_1_Call_Establishment_Principles.pdf

    • Call Establishment Principles

      • What Are Call Legs?

      • End-to-End Calls

  • 4_10_Call_Setup.pdf

    • Call Setup and Digit Manipulation

      • End-to-End Calls

      • Matching Inbound Dial Peers

      • Matching Outbound Dial Peers

      • Digit Collection and Consumption

      • What Is Digit Manipulation?

      • PLAR

  • 4_11_Class_of_Restriction.pdf

    • Class of Restriction

      • Class of Restriction (COR)

      • Steps to Configure Class of Restriction

  • 4_2_Configuring_Dial_Peers.pdf

    • Configuring Dial Peers

      • Understanding Dial Peers

      • Configuring POTS Dial Peers

      • Configuring VoIP Dial Peers

      • Configuring Destination-Pattern Options

      • Default Dial Peer

      • Matching Inbound Dial Peers

      • Matching Outbound Dial Peers

      • Hunt-Group Commands

      • Configuring Hunt Groups

      • Digit Collection and Consumption

      • Understanding Digit Manipulation

      • Practice Item Answer Key

  • 4_3_Special_Purpose_Connections.pdf

    • Special-Purpose Connections

      • Connection Commands

      • PLAR and PLAR-OPX

      • Configuring Trunk Connections

      • Tie-Line Connections

  • 4_4_Building_Scalable_Num_Plan.pdf

    • Building a Scalable Numbering Plan

      • Scalable Numbering Plan

      • Scalable Numbering Plan Attributes

      • Hierarchical Numbering Plans

      • Internal Numbering and Public Numbering Plan Integration

      • Enhancing and Extending an Existing Plan to Accommodate VoIP

  • 4_5_Configuring_Voice_Ports.pdf

    • Configuring Voice Ports

      • Voice Port Applications

      • FXS Ports

      • FXO Ports

      • E&M Ports

      • Timers and Timing

      • Digital Voice Ports

      • ISDN

      • CCS Options

      • Monitoring and Troubleshooting

  • 4_6_Adjusting_Voice_Quality.pdf

    • Adjusting Voice Quality

      • Electrical Characteristics

      • Voice Quality Tuning

      • Echo Cancellation Commands

  • 4_7_Analog_and_Digital_Voice.pdf

    • Analog and Digital Voice Interfaces

      • Local-Loop Connections

      • Analog Voice Interfaces

      • Channel Associated Signaling Systems: T1

      • Channel Associated Signaling Systems: E1

      • Common Channel Signaling Systems

      • PRI/BRI

  • 4_8_Configuring_Analog_and_Digital.pdf

    • Configuring Analog and Digital Voice Interfaces

      • Foreign Exchange Station Ports (FXS)

      • Foreign Exchange Office Ports (FXO)

      • Ear and Mouth Ports (E&M)

      • Common Channel Signaling (CCS): ISDN BRI

      • Timers and Timing

      • Digital Voice Ports

      • Channel Associated Signaling (CAS)

      • Common Channel Signaling (CCS): ISDN Primary Rate Interface (PRI)

  • 4_9_Dial_Peers.pdf

    • Dial Peers

      • What is Dial Peer?

      • Plain Old Telephone Service Dial Peers

      • VoIP Dial Peers

      • Destination-Pattern Options

      • What is the Default Dial Peer?

  • 5_1_CME_GUI_Features.pdf

    • Cisco CME GUI Features

      • User Classes

      • Cisco CallManager Express GUI Prerequisites

      • Accessing the GUI

      • Configuring Administrative User Classes

  • 5_2_Configuring_Phone_Features.pdf

    • Configuring Phone Features

      • Call Transfer

      • Call Forwarding

      • IP Phone Display

      • Calling and Directory Features

      • Productivity Tools

      • Custom IP Phone Rings

      • Music on Hold

  • 6_1_Need_for_Signaling.pdf

    • Need for Signaling and Call Control

      • VoIP Signaling

      • Call Control Models

      • Translation Between Signaling and Call Control Models

      • Call Setup

      • Call Administration and Accounting

      • Call Status and Call Detail Records

      • Address Management

      • Admission Control

      • Centralized Call Control

      • Distributed Call Control

      • Centralized Call Control vs. Distributed Call Control

  • 6_2_Configuring_H323.pdf

    • Configuring H.323

      • H.323 and Associated Recommendations

      • Functional Components of H.323

      • H.323 Call Establishment and Maintenance

      • Call Flows Without a Gatekeeper

      • Call Flows with a Gatekeeper

      • Multipoint Conferences

      • Call Flows with Multiple Gatekeepers

      • Survivability Strategies

      • H.323 Proxy Server

      • Cisco Implementation of H.323

      • Configuring H.323 Gateways

      • Configuring H.323 Gatekeepers

      • Monitoring and Troubleshooting

  • 6_3_Configuring_MGCP.pdf

    • Configuring MGCP

      • MGCP and Its Associated Standards

      • Basic MGCP Components

      • MGCP Endpoints

      • MGCP Gateways

      • MGCP Call Agents

      • Basic MGCP Concepts

      • MGCP Calls and Connections

      • MGCP Events and Signals

      • MGCP Packages

      • MGCP Digit Maps

      • MGCP Control Commands

      • Call Flows

      • Survivability Strategies

      • Cisco Implementation of MGCP

      • Understanding Basics of Cisco CallManager

      • Configuring MGCP

      • Monitoring and Troubleshooting MGCP

  • 7_1_IP_QoS_Mechanisms.pdf

    • IP QoS Mechanisms

      • QoS Mechanisms

      • Classification

      • Marking

      • Trust Boundaries

      • Congestion Management

      • Traffic Shaping

      • Compression

      • Link Fragmentation and Interleaving

  • 7_10_Configuring_QoS_in_WAN.pdf

    • Configuring QoS in the WAN

      • Configuring AutoQoS

  • 7_11_Configuring_CAC.pdf

    • Configuring CAC

      • Need for CAC

      • CAC as Part of Call Control Services

      • RSVP

      • Understanding CAC Tools

      • H.323 CAC

  • 7_12_Voice_Bandwidth.pdf

    • Voice Bandwidth Engineering

      • Erlangs

  • 7_2_Implementing_AutoQoS.pdf

    • Implementing AutoQoS

      • AutoQoS

      • AutoQoS: Router Platforms

      • AutoQoS: Switch Platforms

      • AutoQoS Prerequisites

      • Configuring AutoQoS

      • Monitoring AutoQoS

      • Automation with Cisco AutoQoS

  • 7_3_Comparing_Voice_Quality.pdf

    • Comparing Voice Quality Measurement Standards

      • Audio Clarity

  • 7_4_VoIP_Challenges.pdf

    • VoIP Challenges

      • IP Networking Overview

      • Jitter

      • Delay

  • 7_5_QoS_and_Good_Design.pdf

    • QoS and Good Design

      • Need for QoS Mechanisms

      • Objectives of QoS

      • Applying QoS for End-to-End Improvement of Voice Quality

  • 7_6_Jitter.pdf

    • Jitter

      • Understanding Jitter

      • Overcoming Jitter

      • Adjusting Playout Delay Parameters

      • Symptoms of Jitter on a Network

      • Dynamic Jitter Buffer

      • Static Jitter Buffer

  • 7_7_Delay.pdf

    • Delay

      • Need for a Delay Budget

      • Guidelines for Acceptable Delay

      • Sources of Delay

      • Effects of Coders and Voice Sampling on Delay

      • Managing Serialization Delay

      • Managing Queuing Delay

      • Verifying End-to-End Delay

  • 7_8_Apply_QoS_in_Campus.pdf

    • Applying QoS in the Campus

      • Need for QoS in the Campus

      • Marking Control and Management Traffic

  • 7_9_QoS_Tools_in_WAN.pdf

    • QoS Tools in the WAN

      • Need for QoS on WAN Links

      • Recommendations for Generic QoS in the WAN

      • Bandwidth Provisioning

      • Optimized Queuing

      • Link Efficiency

      • Link Fragmentation and Interleaving

      • CAC

  • en_IP_Telephony_Glossary.pdf

    • IP Telephony v1.0 Course Glossary

      • Numbers

      • A

      • B

      • C

      • D

      • E

      • F

      • G

      • H

      • I

      • J

      • L

      • M

      • N

      • O

      • P

      • Q

      • R

      • S

      • T

      • U

      • V

      • W

  • en_IP_Telephony_Reference_Lists.pdf

Nội dung

Traditional Telephony Basic Components of a Telephony Network This topic introduces the components of traditional telephony networks Basic Components of a Telephony Network IP Telephony v1.0 Cisco Public © 2005 Cisco Systems, Inc All rights reserved 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 Copyright © 2005, Cisco Systems, Inc Introduction to Packet Voice Technologies > Traditional Telephony 1-3 telephones Digital telephones are typically proprietary, meaning that they work with the PBX or key system of that vendor only 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 customer 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 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 1-4 Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc CO Switches This topic describes how CO switches function and make switching decisions Central Office Switches Cisco Public © 2005 Cisco Systems, Inc All rights reserved IP Telephony v1.0 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 Copyright © 2005, Cisco Systems, Inc Introduction to Packet Voice Technologies > Traditional Telephony 1-5 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 „ 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 interexchange carrier (IXC) Although intermediate switches 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 1-6 Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc 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 What Is a PBX? IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public 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 inhouse 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 Copyright © 2005, Cisco Systems, Inc Introduction to Packet Voice Technologies > Traditional Telephony 1-7 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 company’s 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 1-8 Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc 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 Basic Call Setup IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public The figure shows the three major steps in an end-to-end call These steps include: Local signaling — originating side: The user signals the switch by going off hook and sending dialed digits through the local loop 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 Local signaling — terminating side: The terminating switch signals the call recipient by sending ringing voltage through the local loop to the recipient telephone Copyright © 2005, Cisco Systems, Inc Introduction to Packet Voice Technologies > Traditional Telephony 1-9 Supervisory Signaling IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public 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 callprogress tone and not part of supervisory signaling Note 1-10 The ringing tone in the United States is seconds of tone followed by seconds of silence Europe uses a double ring followed by seconds of silence Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc Address Signaling Tone telephone DTMF dialing IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved • Rotary telephone – Pulse dialing Cisco Public There are two types of telephones: a rotary-dial telephone and a push-button (tone) telephone These telephones use two different types of address signaling to notify the telephone company where a subscriber is calling: „ 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 highfrequency 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 Copyright © 2005, Cisco Systems, Inc Introduction to Packet Voice Technologies > Traditional Telephony 1-11 Informational Signaling IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public Tone combinations indicate call progress and are used to notify subscribers of call status Each combination of tones represents a different event in the call process These events include the following: 1-12 „ Dial tone: Indicates that the telephone company is ready to receive digits from the user telephone „ Busy: Indicates that a call cannot be completed because the telephone at the remote end is already in use „ Ringback (normal or PBX): Indicates that the telephone company is attempting to complete a call on behalf of a subscriber „ Congestion: Indicates that congestion in the long-distance telephone network is preventing a telephone call from being processed „ Reorder tone: Indicates that all the local telephone circuits are busy, thus preventing a telephone call from being processed „ Receiver off hook: Indicates that a receiver has been off hook for an extended period of time without placing a call „ No such number: Indicates that a subscriber has placed a call to a nonexistent number „ Confirmation tone: Indicates that the telephone company is attempting to complete a call Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc ... to transmit 1-1 4 Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc Frequency-Division Multiplexing IP Telephony v1.0 Cisco Public © 2005 Cisco Systems,... multiplexing techniques Time-Division Multiplexing IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public 11 Time-division multiplexing (TDM) is used extensively in telephony networks... local city 1-2 4 Cisco Networking Academy Program: IP Telephony v1.0 Copyright © 2005, Cisco Systems, Inc E&M Interface IP Telephony v1.0 © 2005 Cisco Systems, Inc All rights reserved Cisco Public

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