Voice over IP (VoIP) Essentials:

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Voice over IP (VoIP)

Essentials:

Instructor Guide

Published by ComputerPREP, Inc.

Phoenix, Arizona

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Voice over IP (VoIP) Essentials

Developers

Meagan McLaughlin and Brent Capriotti

Editors

Jill McKenna and David Oberman

Publishers

LeAnna Shank and Tina Strong

Project Manager

Karlene Copeland

TRADEMARKS

ComputerPREP£ is a registered trademark of ComputerPREP, Inc. in the United States and other countries.

Microsoft, Microsoft Internet Explorer logo, and Windows are either registered trademarks or trademarks of the Microsoft Corporation in the United States and/or other countries. All other product names and services identified throughout this book are trademarks or registered trademarks of their respective companies. They are used throughout this book in editorial fashion only. No such use, or the use of any trade name, is intended to convey endorsement or other affiliation with the book. Copyrights of any screen captures in this book are the property of the software’s manufacturer.

DISCLAIMER

ComputerPREP, Inc. makes a sincere effort to ensure the accuracy of the material described herein; however, ComputerPREP, Inc. makes no warranty, express or implied, with respect to the quality, correctness, reliability, currentness, accuracy, or freedom from error of this document or the products it describes. ComputerPREP, Inc. makes no representation or warranty with respect to the contents hereof and specifically disclaims any implied warranties of fitness for any particular purpose. ComputerPREP, Inc. disclaims all liability for any direct, indirect, incidental, consequential, special, or exemplary damages resulting from the use of the information in this document or from the use of any products described in this document. Mention of any product does not constitute an endorsement by ComputerPREP, Inc. of that product. Data used in examples and sample data files are intended to be fictional. Any resemblance to real persons or companies is entirely coincidental.

ComputerPREP makes every effort to ensure the accuracy of URLs referenced in all our materials, but we can not guarantee that all will be available throughout the life of the course. When this manual/disk was published, all URLs were checked for accuracy and completeness. However, due to the ever-changing nature of the Internet, some URLs may no longer be available or may have been re-directed.

COPYRIGHT NOTICE

This Guide is copyrighted and all rights are reserved by ComputerPREP, Inc. No part of this publication may be reproduced, transmitted, transcribed, stored in a retrieval system, or translated into any language or computer language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical, manual, or otherwise, without the prior written permission of ComputerPREP, Inc., 410 North 44th Street, Suite 600, Phoenix, Arizona 85008.

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Course Assessment ... Course Assessment-1 Glossary ...Glossary-1 Index ... Index-1 Supplemental CD-ROM Contents ... Supplemental CD-ROM Contents-1 Handouts: Activities ...Handouts: Activities-1 Handouts: Quizzes ... Handouts: Quizzes-1 Handout: Course Assessment...Handout: Course Assessment-1

List of Activities

Activity 1-1: Reviewing basic VoIP concepts...1-20 Activity 2-1: Reviewing VoIP gateways...2-26 Activity 3-1: Reviewing VoIP bandwidth consumption ...3-18 Activity 4-1: Reviewing VoIP Quality of Service (QoS) issues...4-16 Activity 5-1: Reviewing PC phones...5-10 Activity 6-1: Reviewing VoIP standards ...6-14

List of Quizzes

Lesson 1 Quiz...1-24 Lesson 2 Quiz...2-28 Lesson 3 Quiz...3-20 Lesson 4 Quiz...4-18 Lesson 5 Quiz...5-11 Lesson 6 Quiz...6-15

List of Figures

Figure 1-1: Dial-up and dedicated access to the Internet ... 1-3 Figure 1-2: VoIP uses IP to save money and enhance voice and fax services ... 1-5 Figure 1-3: Enterprise toll-bypass ... 1-5 Figure 1-4: Tie line replacement... 1-6 Figure 1-5: Fax over the Internet ... 1-6 Figure 1-6: Voice transmission using PC phones ... 1-7 Figure 1-7: IP-Based public phone service ... 1-7 Figure 1-8: Call center IP telephony... 1-8 Figure 1-9: IP Local line doubling ... 1-8

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Figure 1-10: Premises IP telephony ... 1-9 Figure 1-11: PSTN versus the Internet ...1-10 Figure 1-12: Pulse Amplitude Modulation (PAM) output...1-11 Figure 1-13: PCM coding results in an 8-bit code called DS0...1-11 Figure 1-14: 24 DSOs =1 DS1 frame ...1-12 Figure 1-15: Channelized T1 versus non-channelized T1 ...1-12 Figure 1-16: Use of gateways in Voice over IP...1-13 Figure 1-17: Private intranet ...1-14 Figure 1-18: Managed networks have advantages for Internet telephony ...1-15 Figure 2-1: VoIP gateways facilitate voice over the Internet ... 2-4 Figure 2-2: The gateway is responsible for the talk path ... 2-5 Figure 2-3: Step 1: The originating gateway converts the called number... 2-5 Figure 2-4: Step 2: The originating gatekeeper exchanges call setup information ... 2-5 Figure 2-5: Step 2: The originating gatekeeper negotiates options ... 2-6 Figure 2-6: Step 2: The originating gatekeeper completes the security handshake ... 2-6 Figure 2-7: Step 3: Digitizing function converts analog signals to digital ... 2-6 Figure 2-8: Step 4: Voice signals compressed and converted to IP packets... 2-7 Figure 2-9: Step 5: Destination gateway decompresses voice signals ... 2-7 Figure 2-10: The trunking gateway connects with the destination PSTN ... 2-8 Figure 2-11: The gateway controller and the gateway communicate using MGCP... 2-9 Figure 2-12: The gateway controller sends an SIP message to the proxy server ... 2-9 Figure 2-13: The proxy server uses DHCP to find a location server ...2-10 Figure 2-14: The proxy server uses SIP to INVITE a connection ...2-10 Figure 2-15: Gateway functions in the SIP environment...2-10 Figure 2-16: The originating gateway performs compression for voice calls ...2-12 Figure 2-17: Voice signals are sent in “talk spurts” ...2-12 Figure 2-18: A compressed talk spurt with header information is about 7 Kbps ...2-13 Figure 2-19: Originating gateway demodulates fax signals ...2-13 Figure 2-20: Fax signals are sent as IP packets ...2-14 Figure 2-21: Line side signaling is used by telco switches and PBXs...2-15 Figure 2-22: DID connections are on the trunk side of the telco switch ...2-15 Figure 2-23: PBXs are connected using tie trunks ...2-16 Figure 2-24: ISDN PRI is used by large offices, and ISDN BRI is used by home users...2-16 Figure 2-25: Connections between telco switches use SS7 ...2-17 Figure 2-26: Toll bypass application...2-18 Figure 2-27: Using the correct interface enables integrated networks ...2-19 Figure 2-28: Foreign exchange office (FXO) connection ...2-20 Figure 2-29: Foreign exchange station (FXS) connection...2-20 Figure 2-30: E&M connection ...2-20 Figure 2-31: Integral gateway interfaces are preferable...2-21 Figure 3-1: Branch offices connected to headquarters via the Internet... 3-3 Figure 3-2: Network configured to support VoIP traffic ... 3-4 Figure 3-3: Voice calls and faxes are typically half-duplex transmissions ... 3-5 Figure 3-4: Silence suppression conserves bandwidth ... 3-5 Figure 3-5: Increasing duty cycle reduces bandwidth consumption... 3-6 Figure 3-6: Number of trunks required for 95% dial tone availability ... 3-6 Figure 3-7: Calculation of the theoretical hours of telephony service... 3-6

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Figure 3-14: Centum call seconds (CCS) based on Erlang B assumptions ...3-11 Figure 3-15: Calculating peak bandwidth...3-12 Figure 3-16: Calculating peak bandwidth for a G.729 codec ...3-12 Figure 3-17: Calculating residual bandwidth...3-13 Figure 4-1: Quality VoIP lines can save money ... 4-3 Figure 4-2: The codec produces natural-sounding speech ... 4-3 Figure 4-3: Delay and jitter affect QoS... 4-4 Figure 4-4: Callers speak one at a time when delay occurs ... 4-4 Figure 4-5: If delay is too long, the last packet will be replayed ... 4-4 Figure 4-6: Jitter buffer holds packets to control timing ... 4-5 Figure 4-7: Jitter buffer hold time can increase overall delay ... 4-5 Figure 4-8: 55 ms is a moderate delay time... 4-5 Figure 4-9: A delay time of 115 ms is perceived as poor QoS ... 4-6 Figure 4-10: The ITU recommends a one-way delay limit of 150 ms... 4-6 Figure 4-11: Packet prioritization is a key factor of VoIP ... 4-7 Figure 4-12: The effects of VoIP packet prioritization on data transmissions ... 4-7 Figure 4-13: WAN access speed and packet size ... 4-8 Figure 4-14: Silence suppression conserves bandwidth... 4-8 Figure 4-15: Advanced silence suppression prevents first-word clipping ... 4-9 Figure 4-16: A hybrid performs conversion between two-wire and four-wire circuits ... 4-9 Figure 4-17: Echo cancellation eliminates echo...4-10 Figure 4-18: Gateways can prevent some network problems such as trunk busy-out ...4-11 Figure 5-1: PC phone technology is in its early stages ... 5-3 Figure 5-2: Telecommuter using DSL service ... 5-5 Figure 5-3: IP phones are easy to move to any location... 5-6 Figure 6-1: H.323 is a set of standards ... 6-3 Figure 6-2: H.323 includes physical components and interoperating elements... 6-4 Figure 6-3: H.323 architecture... 6-4 Figure 6-4: The G.7xx series defines audio standards... 6-5 Figure 6-5: Summary of the G.7xx standards... 6-6 Figure 6-6: The H.26x series regulates video transmissions ... 6-6 Figure 6-7: The T.120 series regulates data transmissions ... 6-6 Figure 6-8: H.323 provides interoperability to basic voice conferencing... 6-7 Figure 6-9: SIP works with existing Internet tools and protocols ... 6-7

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Course Description

Welcome to the Voice over IP (VoIP) Essentials course which will help prepare you for the Certified in Convergent Network Technologies (CCNT) exam, a program sponsored by the TIA

(Telecommunications Industry Association).

This course is aimed at preparation and review for the Voice over IP (VoIP) Essentials module of the CCNT exam, as well as professional development for Information Technology (IT)

professionals. The course is designed to be used in a lecture-based classroom setting.

Voice over IP (VoIP) Essentials will provide you with an understanding of Voice over IP

technology. This course has six lessons, and each lesson covers several topics. Following are the six lessons of the VoIP course, along with the topics covered in each lesson.

Topics Covered

Overview

• Overview

• Key VoIP Applications • Making an Internet Call • VoIP and the Internet

Gateways

• Gateway Functions

• Voice Compression and Decompression • Fax Demodulation and Remodulation • Gateway Interfacing

Bandwidth Consumption

• Overview

• Silence Suppression • Trunk Duty Cycle

Quality of Service (QoS) Issues

• Overview

• Network Delay and Jitter • Packet Handling • Silence Supression • Echo Cancellation • Connection QoS PC Phones • Using PCs as Phones • PC Phone Applications Standards • VoIP Standards

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ComputerPREP Courseware

This learning guide was developed for instructor-led training and will assist you during class. Along with comprehensive instructional text and objectives checklists, this learning guide also includes pre-assessment questions, tech terms, as well as lesson summaries and reviews. Each lesson in this course follows a regular structure, along with graphical cues to illustrate important terms and concepts. The structure of a typical module includes:

• Pre-Assessment Questions – Each lesson includes pre-assessment questions to test the student’s understanding of the key concepts presented in the lesson.

• Objectives – Each lesson includes a list of objectives to set the stage for the rest of the lesson.

• Tech Terms – Tech terms appear in bold in the narrative text for quick and easy access (technical terms are also included in the index and glossary).

• Lesson Summary – The Lesson Summaries at the end of each lesson include: an Application Project to extend learning, a Skills Review of key concepts and objectives presented in the lesson, and Lesson Review Questions designed to test understanding. • Glossary – The Glossary contains a list of key terms defined throughout the course which

can be used for self-study once the course has been completed.

• Table of Contents and Index – The Table of Contents appears at the beginning of the course book and the Index appears at the end. These two allow for easy access to review key areas.

Course Objectives

• Define Internet.

• Identify key applications of Internet telephony. • Identify the goals of using VoIP.

• Identify the seven key applications of VoIP.

• Differentiate between the PSTN and the Internet for voice transmissions. • Define PCM.

• Identify the three steps in PCM. • Define gateway.

• Identify the steps to make an Internet call. • Define intranet.

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• Differentiate between the Internet and an intranet for VoIP. • Identify the key challenges to VoIP.

• Distinguish between the H.323 and SIP environments. • Identify the five functions of the VoIP gateway in H.323. • Identify the components of the SIP architecture.

• Define the H.323 gatekeeper function. • Define the connection function.

• Define voice compression and decompression. • Identify the role of the codec.

• Define “talk spurt.”

• Define fax demodulation and remodulation.

• Differentiate between UDP/IP and TCP/IP in VoIP transmissions. • Identify common analog and digital interfaces.

• Describe T1 and E1 connections. • Define bandwidth.

• Define half-duplex. • Define full-duplex.

• Define silence suppression. • Describe the trunk duty cycle. • Define calculations for duty cycle.

• Identify the effect of the truck duty cycle on bandwidth consumption. • Define carrying capacity for VoIP.

• Define residual capacity.

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x

• Define packet prioritization and segmentation and identify their roles in maintaining QoS for VoIP.

• Identify high-priority real-time data applications requiring prioritization along with VoIP. • Identify the role of silence suppression in maintaining QoS for VoIP.

• Define echo cancellation and identify industry standards for echo cancellation. • Identify the role of echo cancellation in maintaining QoS for VoIP.

• Identify the role of the gateway in maintaining QoS for VoIP. • Discuss QoS in the LAN and its relationship to the WAN.

• Differentiate between using a PC as a phone, and using a VoIP gateway. • Identify advantages of using a PC as a phone.

• Identify applications for using a PC as a phone.

• Identify precautions needed when planning for PC phones or IP phones. • Define H.323.

• Define SIP.

• Identify G.7xx standards. • Define G.723.1.

• Identify H.26x standards. • Define RSVP and DiffServ.

Classroom Setup

Student computers are not required for this seminar course. However, if the instructor desires to supplement activities or quizzes electronically, computers addressing these needs will be required for each student. Otherwise, all supplemental material can be distributed as hard-copy documents and completed by students using a pen and paper.

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Classroom

Setup Guide

The Classroom Setup Guide contains information on how to teach Voice over IP (VoIP)

Essentials. Contents of the guide include the following:

• Teaching the Course • Revision History

• Using Your Instructor Guide • The Instructor CD-ROM • The Instructor Slide Show • Instructor Preparation

Teaching the Course

This course presents concepts that will provide students with the knowledge needed to work in the data and telecommunications industry, and the skills necessary to receive a CCNT

certificate from the TIA.

Concepts are presented in a seminar format so that instructors have the flexibility to customize the course to student needs and skill levels. The suggestions and reminders included here are intended as a general guide and should be adapted for the instructor’s presentation style.

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Classroom Setup Guide-2

Revision History

Released November 2002 (version 6.0P)

This release is considered a content upgrade. The main differences between this course and the earlier version of Voice over IP (VoIP) Essentials (v5.1P), released January 31, 2002, are as follows:

• Entered errata (an error in writing or printing, including but not limited to spelling, style and code errors).

• All content changed to reflect current technologies.

• Addition of course assessment tool to the Instructor Section.

Using Your Instructor Guide

Each lesson in this course follows a regular structure, along with graphical cues to illustrate important terms and concepts. The structure of a typical module includes:

• Classroom Setup Guide: The Classroom Setup Guide includes general information about hardware and software requirements for the Instructor slide show, helpful information about your CD-ROM and slide show, and a suggested checklist for preparing to teach the course.

• Pre-Assessment Questions: Each lesson includes pre-assessment questions to test students' understanding of the key concepts presented in the lesson.

• Objectives: Each lesson includes a list of objectives to set the stage for the rest of the lesson.

• Tech Terms: Technology terms appearing in bold in the narrative text and displayed for quick and easy access (terms also included in the index and glossary).

• Instructor Notes: Instructor Notes appear in the margins throughout the lesson, and provide additional support in teaching the course.

• Lesson Summary: The Lesson Summaries at the end of each lesson include an Application Project to extend learning, a Skills Review of key concepts and objectives presented in the lesson, and Lesson Review Questions designed to test understanding.

• Instructor Section: The Instructor Section at the end of each lesson provides an additional activity to be used in class, and a Lesson Quiz to check students' understanding of the key concepts presented in the lesson.

• Glossary: The Glossary contains a list of key terms defined throughout the course, and can be used for self-study once the course has been completed.

• Course Assessment: The Course Assessment checks students’ understanding of all lessons, and can be used as an in-class tool to prepare for the exam.

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• Instructor Guide Handouts: Handouts of the Activities and the Quizzes, without the answers, are for use in the classroom.

• Table of Contents and Index: The Table of Contents appears at the beginning of the book, and the Index appears at the end. These can also be used to review key areas.

The Instructor CD-ROM

The CD-ROM that is included with the Instructor Guide provides valuable tools that you can use to present this course. The CD-ROM will run automatically when placed in your drive, and you will be able to choose from a menu of the following options:

• Implementation Table: The Implementation Table provides a suggested timeline for teaching the course, by lesson and topic. It includes the slide numbers that pertain to each lesson and topic in the course.

• Activity Log: The Activity Log provides a list of all course activities. The objective, a brief description, and the course location are included in the Activity Log for each activity. • Instructor Information: This section includes the Implementation Table, Classroom Setup

Guide, the Activity Log, the Handouts Section from the Instructor Guide, and the Answer Files for the Lesson Reviews, Activities, Quizzes, and Course Assessment.

• Slide Show: The instructor CD-ROM includes a PowerPoint slide show. The PowerPoint slide show can be used to present the key concepts of the course. The slides are labeled, so you can easily navigate to the desired topic or lesson.

• Glossary: This is a comprehensive list of all terms and acronyms defined in the course, in printable format for future reference.

The Instructor Slide Show

The instructor slide show is intended to help your students visualize the concepts being

presented in the course. The slides are labeled, and you can easily navigate to the desired topic or lesson.

Workstation Requirements and Recommendations

To use the instructor slide show, the following software and hardware must be installed on your system:

Software

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Classroom Setup Guide-4

Hardware

• 486 MHz or faster processor • 16 MB RAM

• 30 MB free hard disk space • 2x CD-ROM drive

• Video card with 1MB video memory

Using the Instructor Slide Show

The slide show is intended to help students visualize concepts presented in the course. You can use the slide show by either running it from the CD-ROM or installing it on your hard drive. If you do not have Microsoft PowerPoint on your computer, you can download PowerPoint Viewer by selecting Slide Show from the menu, and then selecting Install PowerPoint Viewer. To run the show directly from the CD-ROM, select Slide Show on the menu, and then select the presentation. Used this way, the show cannot be modified.

If you want to download the slide show to your hard drive, select Slide Show from the menu, and then select Install Slide Show.

Note: It is only necessary to install PowerPoint Viewer if you do not have Microsoft PowerPoint installed on your system.

The slide show can be presented using a projection device. If you prefer to use transparencies, you can print the slides contained in the slide show onto transparency film.

Instructor Preparation

Use this checklist as you prepare for each seminar.

At least three days before you teach this seminar, we recommend that you:

• Review the Instructor Guide and the Instructor Slide Show to become familiar with the material. Determine the material you will present and create a schedule for each day of the seminar.

• Verify that you have a sufficient number of Student Guides.

• Print copies of the PowerPoint slide show on transparency film if you are displaying slides using an overhead projector.

• Practice displaying the slide show with the lecture material. Before students arrive, we recommend that you:

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• Set up the overhead projector, if applicable. • Distribute the Student Guides.

• Prepare a course roster.

• Print the course title and your name on the blackboard or whiteboard. After students arrive, we recommend that you:

• Instruct them to sign the roster. • Introduce the course.

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1 Lesson 1:

Overview

O

BJECTIVES

By the end of this lesson, you will be able to:

•

Define Internet.

•

Identify key applications of Internet telephony.

•

Identify the goals of using VoIP.

•

Identify the seven key applications

of VoIP.

•

Differentiate between the PSTN and the Internet for voice transmissions.

•

Define PCM.

•

Identify the three steps in PCM.

•

Define gateway.

•

Identify the steps to make an Internet call.

•

Define intranet.

•

Differentiate between the Internet and an intranet for VoIP.

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1-2 Voice over IP (VoIP) Essentials

Pre-Assessment Questions

1. For greater reliability and quality of service, Voice over IP (VoIP) transmissions are placed over the _______.

a. PSTN b. Intranet c. Internet d. T1 line

2. True or false: Voice over IP on a managed network (as opposed to the Internet) has several advantages, including more predictable bandwidth.

True

3. What is Internet telephony?

Use of the Internet for real-time voice and video traffic. Internet telephony makes it possible to place long-distance calls and send video information over the Internet.

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Overview

Internet telephony, or Voice over IP (VoIP), is the use of the Internet, or Internet Protocol (IP), for real-time voice (and video) traffic. VoIP is unlike traditional

Internet media, which tended to be downloaded to the PC and played back.

People and organizations are beginning to use Internet telephony to handle and control the costs of voice communications.

Internet

A wide area network (WAN) connecting thousands of disparate networks in industry, education, government and research.

The Internet is an abbreviation for internetwork—a huge, public, and

unregulated linkage of computer networks around the globe. The Internet uses

protocols to control the flow of data from one point to another.

The Internet PSTN Modem or ISDN TA _ISP’s Gateway Dial-up Access Dedicated Access Other Gateways and Hosts

Figure 1-1: Dial-up and dedicated access to the Internet

Internet communications are based on the TCP/IP protocol suite. Two rival protocols have evolved for control of calls in Internet telephony – H.323 and

Session Initiation Protocol (SIP).

The H.323 protocol manages calls between the client and equipment at the Internet telephony service provider (ITSP), and as such is the basis for most Internet telephony systems. H.323, which evolved within the telephony community, is complex, relatively complete, and rigorously defined.

SIP, or RFC2543, evolved within the Internet community. It is simpler, less rigorously defined, and gaining in popularity with both vendors and users.

protocol

A formal set of rules. In a LAN context, a protocol refers to the

standardized rules governing network functions that strongly influence

INSTRUCTOR NOTE:

To view Telephony Magazine on the Internet, see the following Web site:

www.telephonyonline. com

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H.323

A set of standards regulating VoIP transmissions.

Session Initiation Protocol (SIP)

A method of setting up sessions between endpoints for the purpose of real-time communications. Along with several related protocols, forms a set of standards regulating VoIP transmissions.

The most obvious advantage to Internet telephony is toll-free calling. Instead of paying by the minute for a long-distance call, a user chooses to run voice communications over the Internet for a flat monthly access fee. But Internet telephony can provide more than lower long-distance costs. Internet telephony can also:

• Handle voice calls, video calls, and whiteboarding sessions for true multimedia communications.

• Use PCs as phones, replacing proprietary PBX phones with conferencing software.

• Simplify wiring by merging voice and data into a single system. • Lower ownership costs, because installing and maintaining Internet

telephony systems can often be handled by the MIS department or LAN contractor.

Recent events have made comparing costs and benefits of Internet telephony with traditional telephony more difficult. Although the costs of international telephone calls remain high, particularly calls to developing countries, the costs are decreasing in the industrialized countries. Domestically, costs have dropped into the U.S. $0.5 to $0.7 per minute range, or lower. Domestic long-distance rates may have stopped decreasing, while at the same time the turmoil within the Internet service provider space has opened the possibility that Internet access and transport costs may increase.

Key VoIP Applications

The main goals of VoIP are:

• To save money on long-distance charges.

• To incorporate IP voice and fax into certain applications for enhanced services.

These goals are the primary focus of the key VoIP applications.

For additional information about VoIP applications, see the following Web sites: http://www.xchangem ag.com/ (search on VoIP) http://www.iptelephon y.org/frame/pulse.htm l

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The Internet PSTN Boston Headquarters Baton Rouge Branch Router PBX

VoIP Gateway Router with

Voice Gateway

PBX

Figure 1-2: VoIP uses IP to save money and enhance voice and fax services

Following are seven key VoIP applications: 1. Enterprise toll-bypass

2. Fax over the Internet 3. PC phone to PC phone 4. IP-based public phone service 5. Call-center IP telephony (agent-click) 6. IP local line doubling

7. Premises IP telephony

Enterprise Toll Bypass

Enterprise toll-bypass provides toll-free, company-wide voice and fax communications. This application relies on a VoIP gateway. The gateway converts real-time voice and fax signals into IP packets. It then puts them on a LAN for transmission.

Router IP Fax IP Data IP Voice IP Voice

Voice

Fax IP Voice IP Fax Voice

Gateway

IP Data IP Data

IP Voice IP Fax IP Fax IP Data

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full-duplex (FDX)

A simultaneous two-way and independent transmission. Half-duplex is one-way only.

Another version of toll bypass is tie line replacement. The larger enterprises which have telephone systems in multiple cities sometimes connect them with tie lines, which allow for uniform dialing plans and extension dialing between PBXs, while also replacing per-minute toll charges by the flat rate monthly charge for the tie lines. The tie lines can be replaced by Internet telephony.

The Internet Gateway/Router Tie Line Tie Line Tie Line

Without Tie Lines With Tie Lines

Figure 1-4: Tie line replacement

Fax over the Internet

Fax over the Internet allows for a sending toll-free or reduced-rate fax between fax machines at any two locations. This application relies on an IP gateway, but one that only packetizes fax. This gateway may have added features, such as: • Store-and-forward (to compensate for delay).

• Fax broadcast (to send one fax to many destinations).

PSTN The Internet PSTN Gateway Fax Gateway Fax

Atlanta, GA, US Genova, IT

Figure 1-5: Fax over the Internet

PC Phone to PC Phone

PC phone to PC phone is similar to fax over the Internet, except it transmits only voice. The PCs perform the gateway functions, including voice packetizing. An outside gateway is not required.

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The telephony technology is all software, as long as the PC meets minimum specifications that include:

• A sound card. • Speakers. • A microphone.

Conversely, the PC could have just a phone card and be used with a regular telephone.

The Internet

Router Router

Figure 1-6: Voice transmission using PC phones

The software required to perform this function has been bundled with Microsoft Windows XP, and therefore is rapidly becoming widespread.

IP-Based Public Phone Service

IP-based public phone service involves sending voice over the Internet or over new public IP networks. The calls might be:

• Phone to phone. • Phone to PC. • PC to phone.

Like enterprise toll-bypass, IP-based public phone service: • Requires a VoIP gateway.

• Provides FDX communications. Local Telco The Internet BellSouth PSTN ITSP Gateway Gateway

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New public carriers that provide these services have emerged. The gateways are inside the carrier’s network. The user dials the carrier’s access number, then an account number and the destination telephone number, and the call is

completed over the Internet by the carrier to a gateway at the far end.

Call Center IP Telephony

Call-center IP telephony, or agent-click, is a new application for Internet

customers. With agent-click, a customer looking at an online catalog can simply click a phone icon to talk with an agent. This application typically uses an IP telephony gateway. But if the customer and the agent are using their PCs as phones, a gateway is not necessary. Call-center IP telephony is driven primarily by e-commerce and online purchases, not toll-cost avoidance.

Figure 1-8: Call center IP telephony

IP Local Line Doubling

IP local line doubling service allows a single phone line to carry one or more calls, in addition to transmitting PC data. This application uses a VoIP gateway with FDX capability.

Line doubling is extremely useful for people working at home or on the road. A very powerful application that combines enterprise toll bypass with IP local line doubling is possible using currently available equipment. In this

application, an IP phone at the remote office or telecommuter’s home office works with an IP-enabled PBX at the headquarters location. The remote worker is assigned a telephone number at the headquarters PBX.

PSTN The Internet IP Enabled PBX Hub/Router 1 2 3 4 5 6 7 8 9 * 8 # DSL Modem DSLAM Analog Home Phone Conventional Voice IP Pack ets IP Phone IP Voice Step 1 Step 2 Step 3 IP

Figure 1-9: IP Local line doubling

One list of worldwide carriers and their Points of Presence (POPs) is located at:

http://www.iptelephon y.org/frame/popshop_l ocation.html

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A call to the remote worker’s number is routed through the PBX (Step 1), over the Internet (Step 2), to the IP phone (Step 3), which has access to the complete functionality of the PBX, including features such as caller ID, station dialing, enterprise voice mail, and the like. When used with an access service such as DSL, enterprise voice and data can be carried over the IP service on the DSL, and local telephone service carried over the analog portion of the DSL.

Premises IP Telephony

With premises IP telephony, PCs on an IP LAN could: • Make calls to telephones in the same building.

• Make outside calls by also using special VoIP equipment on the premises. Like IP local line doubling, this application uses a VoIP gateway.

The Internet PSTN 1 2 3 4 5 6 7 8 9 * 8 # Hub Router Hub VoIP Gateway 1 2 3 4 5 6 7 8 9 * 8 #

Figure 1-10: Premises IP telephony

Making an Internet Call

The Internet is a global computer internetwork of wide area networks and many different local area networks. Internet communications are based on TCP/IP. Because of that TCP/IP basis, the Internet transmits voice signals differently than does the public switched telephone network (PSTN).

public switched telephone network (PSTN)

The ordinary dial-up telephone network for switched access to local, long-distance, and international services.

As shown in the top diagram, when a call is placed between two locations on the PSTN, a circuit is dedicated to that call for as long as the call lasts. Even if the parties on the line are silent, the circuit remains in use.

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1-10 Voice over IP (VoIP) Essentials Router Router IP Data IP Data IP Data IP Data IP Data IP Data IP Data

Continuous Voice Stream PSTN Dedicated circuit

one circuit, one call

Internet Packet switching

one circuit, many calls

Figure 1-11: PSTN versus the Internet

As shown in the bottom diagram, the Internet is a packet-switched, or

"connectionless," network. The voice signal is divided into individual packets. Network routers determine the best path for each packet to travel. When the packets reach their destination, they are reassembled in the proper order. This method makes efficient use of network resources, but it also increases the chance of losing part of the original transmission.

router

A device operating at the network level that is used to connect two or more local area networks using the same protocol. The router acts in part as a packet switch to send packets from one LAN to the correct

destination on a different LAN using the best available route. The router's functions are independent of lower-layer protocols.

Most modern telecommunications systems are digital. In many business

systems, the phone itself is also digital. If the phone is analog, it is connected to a digital PBX, or digital hybrid key system, and from there to the telephone company’s digital switch.

The most common method used to translate an analog voice or fax signal into a digital signal is pulse code modulation, or PCM. PCM involves three steps. The first step is to separate the voice/fax signal, which is smooth, and sample it to come up with discrete numbers representing the changes in the signal. This activity is called pulse amplitude modulation, or PAM. The voltage of the voice signal at each point where it is sampled will determine the voltage of the new digital signal. The sampling rate is 8,000 samples per second.

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Figure 1-12: Pulse Amplitude Modulation (PAM) output

After the voice signal has been separated and sampled using PAM, the signal will be quantized and companded. Quantizing assigns a number to each sample that is related to the sample's relative voltage. Companding changes those numbers to reflect a smaller step size at low audio volumes, and larger step sizes at higher volumes, resulting in better perceptions of voice quality at the same bandwidth. The method of companding varies slightly between the United States and Europe. In the United States, an algorithm called mu-law is used, whereas in Europe, A-law is used.

The third and final step in PCM is coding. In this step, the numbers generated in the first two steps are converted to an 8-bit code, which can then be combined with the digital transmission stream and sent across the Internet or an intranet. The resulting 8-bit code is called digital signal Level 0 (DS0).

digital signal Level 0 (DS0)

Digital signal Level 0 in the North American digital hierarchy. A 64-Kbps signal which can carry data or PCM voice.

Figure 1-13: PCM coding results in an 8-bit code called DS0

For additional information about PCM, see the following Web site:

www.cs.mdx.ac.uk/sta ffpages/orhan/tel3041 /pcm.htm

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with 23 other conversations into groups of 24 DS0s to make one frame. The frames are organized into precise time sequences, and marked by framing bits at each end to ensure that the bits stay within their time sequence.

DS0

DS0 DS0 DS0 DS0 DS0 DS0 DS0 DS0 DS0 DS0 F

One 8-bit PCM sample = a DS0

24 DS0s + one framing bit = 1 DS1 frame

Figure 1-14: 24 DSOs =1 DS1 frame

The resulting signal is called a DS1. In a T1, or T carrier system, one DS1 signal is carried over two pairs of twisted-pair copper cable. DS1 signals can be

combined on fiber-optic cable also. A standard named SONET is most often used to multiplex many DS1 signals onto fiber.

digital signal Level 1 (DS1)

Digital signal Level 1 in the North American digital hierarchy. The 24 DS0s plus the framing bit, at 8000 frames per second, result in a 1.544-Mbps (T1) signal.

Modern digital PBX and key systems offer T1 interfaces as well as analog interfaces to connect to networks. Likewise, routers and gateways have T1 interfaces as well as analog interfaces.

PCM and TDM allow a voice/fax signal to be transmitted quickly and inexpensively over a T1 line as a digital signal. Twenty-four calls can be transmitted at once over existing T1 lines, and digital signals are inherently faster than analog signals.

In a non-channelized T1, the DS1 signal is carried as a contiguous bit stream through the network. In a channelized T1, individual DS0s may travel different paths inside a long-haul network, resulting in transmissions arriving out of order if the DS0s are broken apart at one end of the network and reassembled at the far end.

Transmit Receive 24 DS0s F 24 DS0s F 24 DS0s F 24 DS0s F Transmit Receive 24 Bytes F 24 Bytes F 24 Bytes F F 24 Bytes

Channelized T1 - DS0s considered separate channels

Non-channelized T1 - Continuous bit stream - 1 channel

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Now that we have looked at the conversion of the voice/fax signal to a digital signal, the next step is to discuss the options for transmitting a voice/fax call over the Internet.

A voice transmission over the Internet can take many forms. It can be from one PC to another PC, from a PC to a phone, or from one phone to another.

Gateways are required when calling from a PC to a phone, or from a phone to a

PC. ITSPs offer gateway access to subscribers. Some corporations also provide gateways to allow telephone service over their dedicated IP network.

gateway

Local area network node that interconnects networks using differing protocols. Gateways translate between protocols as necessary.

PSTN PSTN IntraNet or Internet IntraNet or Internet IntraNet or Internet Router Gateway

Figure 1-16: Use of gateways in Voice over IP

Consider the example of an Internet phone call placed from a PC and intended for a telephone. The PC, equipped with telephony software, a microphone, and speakers, places the call over the Internet to the nearest gateway server. The gateway server acts as a translator between the Internet and the PSTN.

VoIP and the Intranet

Moving voice traffic over the Internet is not without problems. The primary issues affecting Internet telephony are:

• Quality of Service (QoS). A lost or delayed packet during a voice transmission can result in a lack of clarity. A slow or crowded Internet connection can result in latency, or a gap in time between when the speaker speaks and when the listener can actually hear what was said.

• Prioritization. On the Internet, all transmissions are equal. Priority is not given to voice transmission over any other type.

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A secondary issue involves the question of “what kind of voice is it?” Consider a business location – on the wall by the copier is a phone jack with an analog phone plugged into it. The traveler from out of town visits. If that visitor picks up that analog phone and makes a call, he or she is generating one kind of voice signal. But perhaps the visitor disconnects the phone and plugs a laptop fax/modem card into the jack. From the laptop, the visitor might send a fax to his or her home office – generating a “voice” call with very different

characteristics. Yet another call might be placed using the modem to dial up a remote data network – again generating a call with different characteristics. The engineer of the local telecommunications system has relatively little control over what may be plugged into a particular jack.

Many newer gateways monitor and adjust the call signal to compensate for these challenges. They can detect the special tones generated by modems and fax machines, or fax modems, and handle the call accordingly. They can also prioritize traffic appropriately, and take other actions to improve the quality of service.

Because of the issues of quality and reliability on the Internet, some businesses restrict their Internet voice traffic to simple applications such as voice

messaging. However, businesses can extend their use of the technology by implementing it on their own intranets. Intranets are small, private, and more easily managed than the public Internet.

Private IntraNet or VPN Hub/Router Fax G/W Voice G/W 1 2 3 4 5 6 7 8 9 * 8 # 1 2 3 4 5 6 7 8 9 * 8 #

Figure 1-17: Private intranet

An intranet is a privately owned TCP/IP network running through and between locations. An intranet may be connected to the Internet, but access to an intranet by anyone outside the company is protected by security software and hardware known as a firewall.

A similar technique is to use a virtual private network (VPN) from a reliable ISP. A Service Level Agreement (SLA) is the user’s guarantee the network will be managed to the proper level of service.

virtual private network (VPN)

A connection that occurs over a switched or shared line through a process called tunneling. Traffic is routed over Internet links which are secure and tightly controlled in order to provide a network with the performance characteristics of a private network, while retaining the reduced costs associated with the use of the Internet.

For additional information about VoIP protocols and QoS, see the following Web sites: www.protocols.com/vo ip.htm www.slac.stanford.edu /grp/scs/net/talk/voip-mon/ www.iec.org/online/tut orials/voice_qual/

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Service Level Agreement (SLA)

A specification by the carrier providing the managed network, describing and guaranteeing the characteristics of the service provided. May be such things as packet delay, packet arrival jitter, and packet loss percentage. Penalties for failing to meet the standards are often specified as part of the SLA.

Using a managed network such as an intranet solves many of the problems with Internet telephony.

• Because an intranet is smaller with controlled access, it is easier to regulate VoIP transmissions and give them appropriate priority in an intranet

environment.

• Bandwidth is more predictable on an intranet than over the Internet. This factor means better support for real-time voice transmission.

• Companies using intranets can make point-to-point calls via gateway servers attached to the LAN, with no PC-based telephony software required.

Using a managed network, an organization can make point-to-point calls

through gateway servers attached to its computer network and PBX. No Internet account or PC-based telephony software is required. For example, John Smith in a New York office wants to make a point-to-point call to the Tokyo office.

Private Internet Calling Party PBX Gateway Server Router Called Party

Figure 1-18: Managed networks have advantages for Internet telephony

John Smith picks up his phone and dials an extension to connect with the gateway server. Then he dials the number of the Tokyo office. The gateway server transmits the call over the intranet to the gateway at the Tokyo end. The Tokyo gateway converts the signal back to analog or digital (T1), and delivers the call to the proper extension.

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Lesson Summary

Application project

A large international catalog company is considering implementing VoIP to facilitate order handling and communications. What are the applications from which it could benefit? What challenges would the company face, and how could it resolve them? Be sure to consider the possibility of using a private intranet.

Skills review

Following are the key points presented in this lesson:

• Internet telephony, or Voice over IP, is the use of the Internet for real-time voice (and video) traffic.

• Internet communications are based on TCP/IP. The H.323 protocol manages calls between the client and equipment at the Internet telephony service provider (ITSP), and as such is the basis for many Internet telephony

systems. The session initiation protocol (SIP) is a rival method for controlling Internet phone calls.

• Following are seven key VoIP applications: enterprise toll-bypass, fax over the Internet, PC phone to PC phone, IP-based public phone service, call-center IP telephony (agent-click), IP local line doubling, and premises IP telephony.

• When a call is placed between two locations on the PSTN, a circuit is dedicated to that call for as long as the call lasts. Even if the parties on the line are silent, the circuit remains in use.

• The Internet is a packet-switched, or "connectionless," network. The voice signal is divided into individual packets. Network routers determine the best path for each packet to travel. When the packets reach their destination, they are reassembled in the proper order. This method makes efficient use of network resources, but it also increases the chance of losing part of the original transmission.

• Pulse code modulation (PCM) is used to translate an analog voice or fax signal into a digital signal. PCM consists of three steps. The first step is pulse amplitude modulation, or PAM, which divides the voice/fax signal and samples it. After PAM, the signal will be quantized and companded.

Quantizing assigns a number to each sample that is related to the relative voltage of the sample. Companding changes those numbers to reflect the audio volume level of the signal, creating more steps where the volume is low and fewer where the volume is high. The third and final step in PCM is coding. In this step, the numbers generated in the first two steps are converted to an 8-bit code.

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• A voice transmission over the Internet can take many forms. It can be from one PC to another PC, from a PC to a phone, or from one phone to another. Gateways are required when calling from a PC to a phone, or from phone to PC.

• ITSPs offer gateway access to subscribers. Some corporations also provide gateways to allow telephone service over their dedicated IP network.

• The primary issues affecting Internet telephony are: quality of service (QoS), prioritization and bandwidth. Businesses can extend their use of the

technology by implementing it on their own intranets. Intranets are small, private, and more easily managed than the public Internet.

Now that you have completed this lesson,

you should be able to:

9Define Internet.

9Identify key applications of Internet telephony. 9Identify the goals of using VoIP.

9Identify the seven key applications of VoIP.

9Differentiate between the PSTN and the Internet for voice transmissions. 9Define PCM.

9Identify the three steps in PCM. 9Define gateway.

9Identify the steps to make an Internet call. 9Define intranet.

9Differentiate between the Internet and an intranet for VoIP. 9Identify the key challenges to VoIP.

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Lesson 1 Review

1. What VoIP application does NOT require a gateway?

PC phone to PC phone. A PC phone can perform gateway functions. The PC must also have: a sound card, speakers and a microphone; or a phone card and an ordinary telephone. Otherwise, the telephony technology is all software.

2. What translates between the PSTN and the Internet?

Gateway server. The gateway server digitizes and compresses the analog signal and transfers it to the Internet. On the receiving end, the gateway server performs these functions in reverse.

3. What is the Internet?

A packet-switched, or connectionless, network. The Internet, unlike the PSTN, divides transmissions into packets, which are sent on various paths through the network and reassembled at the receiving end.

4. For greater reliability and quality of service, Voice over IP transmissions are placed over the ______.

Intranet. Unlike Internet telephony, Voice over IP uses a company’s private, regulated intranet for voice and video traffic.

5. What are the advantages of Voice over IP on a managed network (as opposed to the Internet)?

Because the intranet is more regulated than the Internet, Voice over IP provides more predictable bandwidth, high priority for voice and video transmissions, and easy point-to-point calls to distant offices.

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Lesson 1

Instructor Section

This section is a supplement containing additional tasks for students to complete in conjunction with the lesson. The instructor may use all, some or none of these additional tools, as appropriate to the specific learning

environment. These elements are: • Activities

Pen-and-paper activities to be completed during class or as homework. • Lesson Quiz

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Activity 1-1: Reviewing basic VoIP concepts

The purpose of this activity is to reinforce what you have learned so far about Voice over IP technologies. Your instructor will tell you the correct answers. Your instructor may choose to have you answer these questions as part of a group discussion.

1. Match the following VoIP applications with the appropriate description.

IP local line doubling, premises IP telephony, call-center IP telephony, fax over the Internet, IP-based public phone service, enterprise toll-bypass, PC phone to PC phone:

a. Provides toll-free, company-wide voice and fax communications.

Enterprise toll-bypass

b. Allows for a sending toll-free or reduced-rate fax between fax machines at any two locations.

Fax over the Internet

c. Similar to fax over the Internet, except it transmits only voice.

PC phone to PC phone

d. Sending voice over the Internet or over new public IP networks.

IP-based public phone service

e. A customer looking at an online catalog can simply click a phone icon to talk with an agent, also known as agent-click.

Call-center IP telephony

f. Allows a single phone line to carry one or more calls, in addition to transmitting PC data.

IP local line doubling

g. PCs on an IP LAN can make calls to telephones in the same building or make outside calls by using special VoIP equipment on the premises.

Premises IP telephony

2. Describe an Internet telephone call that originates at a PC phone and

terminates at a standard telephone. Be sure to indicate when the call is transmitted over the Internet and when the call is transmitted over the PSTN, and describe the difference between the two. Also, indicate when a gateway is needed.

The PC, equipped with telephony software, a microphone, and speakers, places the call over the Internet to the nearest gateway server. Because the call is being sent to a standard telephone, a gateway server is needed at the receiving end to move the call from the Internet to the PSTN. The receiving gateway server acts as a translator between the Internet and the PSTN. The gateway server sends the call via the PSTN to the receiving phone, and the call is completed.

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Local Telco The Internet Gateway Router Router

Figure 1-19: VoIP via a gateway

When the call is on the Internet, it is divided into packets, and the router determines the best route for each packet to take through the Internet. The packets may arrive at their destination at slightly different times depending on the path the packet takes through the network. When the call is on the PSTN, a circuit is dedicated to that transmission for the length of the call.

3. Describe the process of pulse code modulation (PCM).

One method used to translate an analog voice or fax signal into a digital signal is pulse code modulation, or PCM. PCM consists of three steps. The first step is to break up the voice/fax signal, which is smooth, and sample it to come up with discrete numbers to represent the changes in the signal. This step is called pulse amplitude modulation, or PAM. The voltage of the voice signal at each point where it sampled will determine the voltage of the new digital signal.

Figure 1-20: Pulse Amplitude Modulation (PAM) output

After the voice signal has been divided and sampled using PAM, the signal will be quantized and companded. Quantizing assigns a number to each sample that is related to the relative voltage of the sample. Companding

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The third and final step in PCM is coding. In this step, the numbers generated in the first two steps are converted to an 8-bit code, which can then be

combined with the digital transmission stream and sent across the Internet or intranet.

Figure 1-21: PCM coding results in an 8-bit code called DS0

After the digital signal has been created by PCM, it must be multiplexed to be transmitted at high speeds. The process most commonly used to multiplex a digital signal for T1 transmission is called time-division multiplexing, or TDM. TDM takes the 8-bit “words,” or DS0 signals, generated by PCM and combines them with 23 other conversations into groups of 24 words to make one frame. The frames are organized into precise time sequences, and marked by framing bits at each end to ensure that the bits stay within their time sequence,

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Lesson 1 Quiz

1. Internet telephony is:

a. Using the Internet for real-time data traffic. b. Using the PSTN for real-time data traffic.

c. Using the Internet for real-time voice and video traffic. d. Using the PSTN for real-time voice and video traffic.

2. Which of the following VoIP applications does NOT require a gateway? a. Fax over the Internet

b. PC phone to PC phone

c. IP-based public phone service d. Premise IP telephony

3. The _____ translates between the PSTN and the Internet. a. Router b. Telephony software c. Gateway server d. Modem 4. The Internet: a. Is a dedicated circuit.

b. Is a packet-switched, or connectionless, network. c. Transmits all information over the same path. d. Is a small, managed network.

5. In a VoIP network, a router:

a. Translates between the Internet and the PSTN.

b. Decides the best path for the call to take through the network. c. Prioritizes voice calls.

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Voice over IP (VoIP) Essentials:

Voice over IP (VoIP)

Essentials:

Instructor Guide

Published by ComputerPREP, Inc.

Phoenix, Arizona

Voice over IP (VoIP) Essentials

Developers

Editors

Publishers

Project Manager

TRADEMARKS

DISCLAIMER

COPYRIGHT NOTICE

Table of Contents

List of Activities

List of Quizzes

List of Figures

Course Description

Topics Covered

ComputerPREP Courseware

Course Objectives

Classroom Setup

Classroom

Setup Guide

Teaching the Course

Revision History

Using Your Instructor Guide

The Instructor CD-ROM

The Instructor Slide Show

Workstation Requirements and Recommendations

Software

Hardware

Using the Instructor Slide Show

Instructor Preparation

1

Lesson 1:

Overview

O

BJECTIVES

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Pre-Assessment Questions

Overview

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Key VoIP Applications

Enterprise Toll Bypass

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Fax over the Internet

PC Phone to PC Phone

IP-Based Public Phone Service

Call Center IP Telephony

IP Local Line Doubling

Premises IP Telephony

Making an Internet Call

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VoIP and the Intranet

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Lesson Summary

Application project

Skills review

Now that you have completed this lesson,

you should be able to: