Voice over IP (VoIP) Essentials:

Voice over IP (VoIP) Essentials: Instructor Guide Published by ComputerPREP, Inc. Phoenix, Arizona PCN01-CNVOIP-PR-210 Version 6.0P

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. Copyright 2001-2002 by ComputerPREP, Inc. All Rights Reserved ISBN: 0-7423-1938-5

iii Table of Contents Course Description...vii ComputerPREP Courseware...viii Course Objectives...viii Classroom Setup...x Classroom Setup Guide... Classroom Setup Guide-1 Lesson 1: Overview... 1-1 Pre-Assessment Questions... 1-2 Overview... 1-3 Key VoIP Applications... 1-4 Making an Internet Call... 1-9 VoIP and the Intranet...1-13 Lesson Summary...1-16 Lesson 1 Review...1-18 Lesson 1 Instructor Section...1-19 Lesson 2: Gateways... 2-1 Pre-Assessment Questions... 2-2 Gateway Functions... 2-3 Voice Compression and Decompression...2-11 Fax Demodulation and Remodulation...2-13 Gateway Interfacing...2-14 Lesson Summary...2-22 Lesson 2 Review...2-24 Lesson 2 Instructor Section...2-25 Lesson 3: Bandwidth Consumption... 3-1 Pre-Assessment Questions... 3-2 Overview... 3-3 Silence Suppression... 3-4 Trunk Duty Cycle... 3-5 Carrying Capacity... 3-8 Estimating Bandwidth Requirements...3-10 Lesson Summary...3-14 Lesson 3 Review...3-16 Lesson 3 Instructor Section...3-17 Lesson 4: Quality of Service (QoS) Issues... 4-1 Pre-Assessment Questions... 4-2 Overview... 4-3 Network Delay and Jitter... 4-4 Packet Handling... 4-6 Silence Suppression... 4-8 Echo Cancellation... 4-9 Connection QoS...4-10 Lesson Summary...4-12 Lesson 4 Review...4-14 Lesson 4 Instructor Section...4-15

iv Lesson 5: PC Phones... 5-1 Pre-Assessment Questions... 5-2 Using PCs as Phones... 5-3 PC Phone Applications... 5-4 Lesson Summary... 5-7 Lesson 5 Review... 5-8 Lesson 5 Instructor Section... 5-9 Lesson 6: Standards... 6-1 Pre-Assessment Questions... 6-2 VoIP Standards... 6-3 Lesson Summary... 6-9 Lesson 6 Review...6-11 Lesson 6 Instructor Section...6-13 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

v 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 Figure 3-8: Calculation of duty cycle... 3-7 Figure 3-9: More trunks provide higher duty cycle... 3-7 Figure 3-10: Each trunk provides 2-4 Kbps... 3-8 Figure 3-11: WAN access link VoIP carrying capacity... 3-9 Figure 3-12: Residual bandwidth available for nonreal-time transmission... 3-9 Figure 3-13: Calculating available VoIP bandwidth...3-10

vi 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

vii 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 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 Bandwidth Consumption Overview Standards VoIP Standards Silence Suppression Trunk Duty Cycle Carrying Capacity Estimating Bandwidth Requirements

viii 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.

ix 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. Identify usage data which may be available from voice system or provider. Calculate peak bandwidth required and average in use for voice traffic. Explain the significance of QoS to VoIP. Define network delay and jitter and identify solutions.

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 hardcopy documents and completed by students using a pen and paper.

Classroom Setup Guide-1 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.

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.

Classroom Setup Guide-3 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 Microsoft PowerPoint or a PowerPoint Viewer Note: PowerPoint Viewer is included on the CD-ROM and can be downloaded, if necessary. Windows 95/98/2000 or Windows NT 4.0 Workstation

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: Install the slide show on your hard drive.

Classroom Setup Guide-5 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.

Classroom Setup Guide-6

1Lesson 1: Overview OBJECTIVES 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. Identify the key challenges to VoIP.

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.

Lesson 1: Overview 1-3 INSTRUCTOR NOTE: To view Telephony Magazine on the Internet, see the following Web site: www.telephonyonline. com 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. Dial-up Access PSTN Modem or ISDN TA ISP s Gateway Dedicated Access The Internet 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 the design of network components. Transmission Control Protocol/Internet Protocol (TCP/IP) A packet-based protocol suite used in many network architectures that provides reliable end-to-end delivery.

1-4 Voice over IP (VoIP) Essentials 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. INSTRUCTOR NOTE: 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 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.

Lesson 1: Overview 1-5 Baton Rouge Branch Router The Internet VoIP Gateway Boston Headquarters Router with Voice Gateway PSTN PBX 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. Fax IP Voice IP Fax Voice Voice Gateway IP Data IP Data IP Voice IP Fax IP Fax IP Data Router IP Fax IP Data IP Voice IP Voice Figure 1-3: Enterprise toll-bypass At the same time, the gateway takes packets off the LAN and converts them back into voice and fax signals. The ability to send and receive transmissions simultaneously is called full-duplex (FDX) communication.

1-6 Voice over IP (VoIP) Essentials 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. Gateway/Router Tie Line Tie Line The Internet Tie Line With Tie Lines Without 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). Fax PSTN Atlanta, GA, US Gateway The Internet Gateway PSTN Genova, IT Fax 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.

Lesson 1: Overview 1-7 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. Router The Internet 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. BellSouth PSTN Atlanta, GA, US ITSP Gateway The Internet Gateway Local Telco Genova, IT Figure 1-7: IP-Based public phone service

1 2 3 4 5 6 7 8 9 * 8 # 1-8 Voice over IP (VoIP) Essentials INSTRUCTOR NOTE: One list of worldwide carriers and their Points of Presence (POPs) is located at: http://www.iptelephon y.org/frame/popshop_l ocation.html 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. Conventional Voice Step 1 PSTN Voice DSLAM DSL Modem Hub/Router IP Enabled PBX IP Step 2 The Internet IP Packets Analog Home Phone IP Step 3 IP Phone Figure 1-9: IP Local line doubling

1 2 3 4 5 6 7 8 9 * 8 # 1 2 3 4 5 6 7 8 9 * 8 # Lesson 1: Overview 1-9 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. Router The Internet PSTN Hub Hub VoIP Gateway 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, longdistance, 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.

1-10 Voice over IP (VoIP) Essentials PSTN Continuous Voice Stream Dedicated circuit one circuit, one call IP Data Router IP Data IP Data Internet IP Data Packet switching one circuit, many calls IP Data IP Data Router IP Data 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.

Lesson 1: Overview 1-11 Figure 1-12: Pulse Amplitude Modulation (PAM) output INSTRUCTOR NOTE: For additional information about PCM, see the following Web site: www.cs.mdx.ac.uk/sta ffpages/orhan/tel3041 /pcm.htm 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 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 transmission over either copper or fiber optic systems is called time-division multiplexing, or TDM. In the United States, the first step in multiplexing takes the 8-bit DS0 signals generated by PCM and combines them

1-12 Voice over IP (VoIP) Essentials 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 One 8-bit PCM sample = a DS0 DS0 DS 0 DS0 DS0 DS0 DS0 DS0 DS0 DS0 DS0 F 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. Channelized T1 - DS0s considered separate channels Transmit 24 DS0s F F 24 DS0s 24 DS0s F F 24 DS0s Receive Non-channelized T1 - Continuous bit stream - 1 channel Transmit 24 Byt es F 24 Bytes F F 24 Bytes F 24 Bytes Receive Figure 1-15: Channelized T1 versus non-channelized T1

Lesson 1: Overview 1-13 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. Router IntraNet or Internet IntraNet or Internet IntraNet or Internet Gateway PSTN PSTN 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. Bandwidth. Bandwidth is limited on dial-up connections to the Internet.

1 2 3 4 5 6 7 8 9 * 8 # 1 2 3 4 5 6 7 8 9 * 8 # 1-14 Voice over IP (VoIP) Essentials INSTRUCTOR NOTE: 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/voipmon/ www.iec.org/online/tut orials/voice_qual/ 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. Voice G/W Hub/Router Fax G/W Private IntraNet or VPN 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.

Lesson 1: Overview 1-15 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 Router Gateway Server PBX Calling Party 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. Internet telephony is changing very quickly. In the future, calls and video transmitted on the Internet will have nearly the same quality and reliability as those sent on an intranet. This technology will enable users to access Voice over IP services as easily and reliably over the Internet as on a managed network.

1-16 Voice over IP (VoIP) Essentials 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, callcenter 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.