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1 Voice over IP (VoIP) Essentials: Student Guide Published by ComputerPREP, Inc. Phoenix, Arizona PCL01-CNVOIP-PR-210 Version 6.0P

2 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 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, 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 Copyright by ComputerPREP, Inc. All Rights Reserved ISBN: X

3 iii Table of Contents Course Description...vii ComputerPREP Courseware...viii Course Objectives...viii Classroom Setup...x Lesson 1: Overview Pre-Assessment Questions Overview Key VoIP Applications Making an Call VoIP and the Intranet Lesson Summary Lesson 1 Review Lesson 2: s Pre-Assessment Questions Functions Voice Compression and Decompression Fax Demodulation and Remodulation Interfacing Lesson Summary Lesson 2 Review Lesson 3: Bandwidth Consumption Pre-Assessment Questions Overview Silence Suppression Trunk Duty Cycle Carrying Capacity Estimating Bandwidth Requirements Lesson Summary Lesson 3 Review Lesson 4: Quality of Service (QoS) Issues Pre-Assessment Questions Overview Network Delay and Jitter Packet Handling Silence Suppression Echo Cancellation Connection QoS Lesson Summary Lesson 4 Review Lesson 5: PC Phones Pre-Assessment Questions Using PCs as Phones PC Phone Applications Lesson Summary Lesson 5 Review

4 iv Lesson 6: Standards Pre-Assessment Questions VoIP Standards Lesson Summary Lesson 6 Review Course Assessment...Course Assessment-1 Glossary...Glossary-1 Index... Index-1 Supplemental CD-ROM Contents... Supplemental CD-ROM Contents-1 List of Figures Figure 1-1: Dial-up and dedicated access to the Figure 1-2: VoIP uses IP to save money and enhance voice and fax services Figure 1-3: Enterprise toll-bypass Figure 1-4: Tie line replacement Figure 1-5: Fax over the Figure 1-6: Voice transmission using PC phones Figure 1-7: IP-Based public phone service Figure 1-8: Call center IP telephony Figure 1-9: IP Local line doubling Figure 1-10: Premises IP telephony Figure 1-11: PSTN versus the Figure 1-12: Pulse Amplitude Modulation (PAM) output Figure 1-13: PCM coding results in an 8-bit code called DS Figure 1-14: 24 DSOs =1 DS1 frame Figure 1-15: Channelized T1 versus non-channelized T Figure 1-16: Use of gateways in Voice over IP Figure 1-17: Private intranet Figure 1-18: Managed networks have advantages for telephony Figure 2-1: VoIP gateways facilitate voice over the Figure 2-2: The gateway is responsible for the talk path Figure 2-3: Step 1: The originating gateway converts the called number Figure 2-4: Step 2: The originating gatekeeper exchanges call setup information Figure 2-5: Step 2: The originating gatekeeper negotiates options Figure 2-6: Step 2: The originating gatekeeper completes the security handshake Figure 2-7: Step 3: Digitizing function converts analog signals to digital Figure 2-8: Step 4: Voice signals compressed and converted to IP packets Figure 2-9: Step 5: Destination gateway decompresses voice signals Figure 2-10: The trunking gateway connects with the destination PSTN Figure 2-11: The gateway controller and the gateway communicate using MGCP Figure 2-12: The gateway controller sends an SIP message to the proxy server Figure 2-13: The proxy server uses DHCP to find a location server Figure 2-14: The proxy server uses SIP to INVITE a connection Figure 2-15: functions in the SIP environment Figure 2-16: The originating gateway performs compression for voice calls Figure 2-17: Voice signals are sent in talk spurts Figure 2-18: A compressed talk spurt with header information is about 7 Kbps Figure 2-19: Originating gateway demodulates fax signals Figure 2-20: Fax signals are sent as IP packets Figure 2-21: Line side signaling is used by telco switches and PBXs

5 v Figure 2-22: DID connections are on the trunk side of the telco switch Figure 2-23: PBXs are connected using tie trunks Figure 2-24: ISDN PRI is used by large offices, and ISDN BRI is used by home users Figure 2-25: Connections between telco switches use SS Figure 2-26: Toll bypass application Figure 2-27: Using the correct interface enables integrated networks Figure 2-28: Foreign exchange office (FXO) connection Figure 2-29: Foreign exchange station (FXS) connection Figure 2-30: E&M connection Figure 2-31: Integral gateway interfaces are preferable Figure 3-1: Branch offices connected to headquarters via the Figure 3-2: Network configured to support VoIP traffic Figure 3-3: Voice calls and faxes are typically half-duplex transmissions Figure 3-4: Silence suppression conserves bandwidth Figure 3-5: Increasing duty cycle reduces bandwidth consumption Figure 3-6: Number of trunks required for 95% dial tone availability Figure 3-7: Calculation of the theoretical hours of telephony service Figure 3-8: Calculation of duty cycle Figure 3-9: More trunks provide higher duty cycle Figure 3-10: Each trunk provides 2-4 Kbps Figure 3-11: WAN access link VoIP carrying capacity Figure 3-12: Residual bandwidth available for nonreal-time transmission Figure 3-13: Calculating available VoIP bandwidth Figure 3-14: Centum call seconds (CCS) based on Erlang B assumptions Figure 3-15: Calculating peak bandwidth Figure 3-16: Calculating peak bandwidth for a G.729 codec Figure 3-17: Calculating residual bandwidth Figure 4-1: Quality VoIP lines can save money Figure 4-2: The codec produces natural-sounding speech Figure 4-3: Delay and jitter affect QoS Figure 4-4: Callers speak one at a time when delay occurs Figure 4-5: If delay is too long, the last packet will be replayed Figure 4-6: Jitter buffer holds packets to control timing Figure 4-7: Jitter buffer hold time can increase overall delay Figure 4-8: 55 ms is a moderate delay time Figure 4-9: A delay time of 115 ms is perceived as poor QoS Figure 4-10: The ITU recommends a one-way delay limit of 150 ms Figure 4-11: Packet prioritization is a key factor of VoIP Figure 4-12: The effects of VoIP packet prioritization on data transmissions Figure 4-13: WAN access speed and packet size Figure 4-14: Silence suppression conserves bandwidth Figure 4-15: Advanced silence suppression prevents first-word clipping Figure 4-16: A hybrid performs conversion between two-wire and four-wire circuits Figure 4-17: Echo cancellation eliminates echo Figure 4-18: s can prevent some network problems such as trunk busy-out Figure 5-1: PC phone technology is in its early stages Figure 5-2: Telecommuter using DSL service Figure 5-3: IP phones are easy to move to any location Figure 6-1: H.323 is a set of standards Figure 6-2: H.323 includes physical components and interoperating elements Figure 6-3: H.323 architecture Figure 6-4: The G.7xx series defines audio standards Figure 6-5: Summary of the G.7xx standards

6 vi Figure 6-6: The H.26x series regulates video transmissions Figure 6-7: The T.120 series regulates data transmissions Figure 6-8: H.323 provides interoperability to basic voice conferencing Figure 6-9: SIP works with existing tools and protocols

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 Call VoIP and the s Functions Voice Compression and Decompression Fax Demodulation and Remodulation 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

8 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. Identify key applications of telephony. Identify the goals of using VoIP. Identify the seven key applications of VoIP. Differentiate between the PSTN and the for voice transmissions. Define PCM. Identify the three steps in PCM. Define gateway. Identify the steps to make an call. Define intranet.

9 ix Differentiate between the 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.

10 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 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.

11 1Lesson 1: Overview OBJECTIVES By the end of this lesson, you will be able to: Define. Identify key applications of telephony. Identify the goals of using VoIP. Identify the seven key applications of VoIP. Differentiate between the PSTN and the for voice transmissions. Define PCM. Identify the three steps in PCM. Define gateway. Identify the steps to make an call. Define intranet. Differentiate between the and an intranet for VoIP. Identify the key challenges to VoIP.

12 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. d. T1 line 2. True or false: Voice over IP on a managed network (as opposed to the ) has several advantages, including more predictable bandwidth. 3. What is telephony?

13 Lesson 1: Overview 1-3 Overview telephony, or Voice over IP (VoIP), is the use of the, or Protocol (IP), for real-time voice (and video) traffic. VoIP is unlike traditional media, which tended to be downloaded to the PC and played back. People and organizations are beginning to use telephony to handle and control the costs of voice communications. A wide area network (WAN) connecting thousands of disparate networks in industry, education, government and research. The is an abbreviation for internetwork a huge, public, and unregulated linkage of computer networks around the globe. The uses protocols to control the flow of data from one point to another. Dial-up Access PSTN Modem or ISDN TA ISP s Dedicated Access The Other s and Hosts Figure 1-1: Dial-up and dedicated access to the communications are based on the TCP/IP protocol suite. Two rival protocols have evolved for control of calls in telephony H.323 and Session Initiation Protocol (SIP). The H.323 protocol manages calls between the client and equipment at the telephony service provider (ITSP), and as such is the basis for most telephony systems. H.323, which evolved within the telephony community, is complex, relatively complete, and rigorously defined. SIP, or RFC2543, evolved within the 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/ Protocol (TCP/IP) A packet-based protocol suite used in many network architectures that provides reliable end-to-end delivery.

14 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 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 for a flat monthly access fee. But telephony can provide more than lower long-distance costs. 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 telephony systems can often be handled by the MIS department or LAN contractor. Recent events have made comparing costs and benefits of 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 service provider space has opened the possibility that 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.

15 Lesson 1: Overview 1-5 Baton Rouge Branch The VoIP Boston Headquarters with Voice 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 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 IP Data IP Data IP Voice IP Fax IP Fax IP Data 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.

16 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 telephony. / Tie Line Tie Line The Tie Line With Tie Lines Without Tie Lines Figure 1-4: Tie line replacement Fax over the Fax over the 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 The PSTN Genova, IT Fax Figure 1-5: Fax over the PC Phone to PC Phone PC phone to PC phone is similar to fax over the, except it transmits only voice. The PCs perform the gateway functions, including voice packetizing. An outside gateway is not required.

17 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. The 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 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 The Local Telco Genova, IT Figure 1-7: IP-Based public phone service

18 * 8 # 1-8 Voice over IP (VoIP) Essentials 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 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 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/ IP Enabled PBX IP Step 2 The IP Packets Analog Home Phone IP Step 3 IP Phone Figure 1-9: IP Local line doubling

19 * 8 # * 8 # Lesson 1: Overview 1-9 A call to the remote worker s number is routed through the PBX (Step 1), over the (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 PSTN Hub Hub VoIP Figure 1-10: Premises IP telephony Making an Call The is a global computer internetwork of wide area networks and many different local area networks. communications are based on TCP/IP. Because of that TCP/IP basis, the 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.

20 1-10 Voice over IP (VoIP) Essentials PSTN Continuous Voice Stream Dedicated circuit one circuit, one call IP Data IP Data IP Data IP Data Packet switching one circuit, many calls IP Data IP Data IP Data Figure 1-11: PSTN versus the As shown in the bottom diagram, the 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.

21 Lesson 1: Overview 1-11 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 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

22 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 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

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