Voice over IP (VoIP) Essentials:

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Voice over IP (VoIP)

Essentials:

Student Guide

Published by ComputerPREP, Inc.

Phoenix, Arizona

PCL01-CNVOIP-PR-210 Version 6.0P

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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.

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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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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 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

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Lesson 6: Standards ... 6-1

Pre-Assessment Questions... 6-2 VoIP Standards ... 6-3 Lesson Summary ... 6-9 Lesson 6 Review ...6-11

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 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 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

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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 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

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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 • Carrying Capacity

• Estimating Bandwidth Requirements

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.

• 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.

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• 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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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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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.

_________

3. What is Internet telephony?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________

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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 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.

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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.

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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

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

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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

Atlanta, GA, US Genova, IT

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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

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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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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

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

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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

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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.

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.

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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?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ 2. What translates between the PSTN and the Internet?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ 3. What is the Internet?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ 4. For greater reliability and quality of service, Voice over IP transmissions

are placed over the ______.

__________________________________________________________________________ __________________________________________________________________________ 5. What are the advantages of Voice over IP on a managed network (as

opposed to the Internet)?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________

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2 Lesson 2:

Gateways

O

BJECTIVES

This lesson defines gateways and their various functions. By the end of this

lesson, you will be able to:

•

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.

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Pre-Assessment Questions

1. Which gateway function is performed ONLY for fax transmissions? a. Connection function

b. Demodulation function c. Gatekeeper function d. Compression function

2. True or false: The gatekeeper is responsible for providing the talk path for the voice call.

_________

3. Which interface connects the gateway to telephones, fax machines and KTS lines?

__________________________________________________________________________ __________________________________________________________________________ __________________________________________________________________________

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Gateway Functions

In general, a gateway is an interface between different protocols, which usually operates at all levels of the protocol stack. In the context of VoIP, there are two generic types of gateways – the signaling gateway and the media gateway. This lesson introduces media gateway functions, discusses signaling and call control functions, and then covers conventional telephony signaling in more detail.

signaling gateway

The signaling gateway translates call control and administrative signals present on the circuit switched PSTN into one of two call control

methodologies found in VoIP implementations – SIP or H.323.

media gateway

The media gateway takes the conventional telephony information – either digital or analog – and packetizes it for transmission across the Internet or an intranet.

Media Gateway Functions

The information to be carried across the Internet is known as media, and might be voice, fax, video or data. The functionality of the media gateway can include: • Conversion between analog and PCM digital information, as described in

Lesson 01. The gateway may support a variety of codecs and methods for compressing the information.

• Transcoding – changing from one coding format to another. For example, the input could be PCM using A-law while the output is PCM using mu-law. • Fax/Modem detection – deciding what “kind” of voice is present on the

analog or digital channel based on the tones present, including selecting and controlling the correct codec functions.

• Playing tones or other call progress indicators or announcements. • Playing IVR voice prompts and collecting end user responses.

• Monitoring the physical state of the telephony interface and formulating messages to the signaling gateway or call control agent indicating changes in state (such as originating a call by going off-hook, or terminating a call by going on-hook).

Gateway Functions in the H.323 Environment

The H.323 suite of protocols originated within the international telephony community, and is related to protocols used to control audio and video

conferencing using a number of different transport networks. H.323 is relatively mature and complex, and envisions intelligent end points communicating with each other in a peer-to-peer relationship for call setup.

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A VoIP gateway in H.323 performs the following five basic steps: • Gatekeeper function (optional—one per zone)

• Connection function

• Digitizing function, if required

• Voice packetization and optionally compression or fax demodulation function

• Decompression or remodulation function

gatekeeper

Device in a VoIP transmission responsible for voice call setup and monitoring, as well as additional features such as call transfer,

conference calling, and call breakdown. One gatekeeper is required per zone.

packetization

Division of signals into packets for transmission.

compression

Method to reduce bandwidth used for digital transmission.

demodulation

Conversion of a fax into its original digital format for transmission over the Internet.

decompression

Conversion of a digital voice signal to the appropriate telephony interface to complete transmission.

remodulation

Re-conversion of a fax signal into an analog signal for transmission to the destination fax machine.

The route illustrated in Figure 2-1 makes it possible to transmit voice and fax over an intranet or the Internet.

The Internet Gateway Server Gatekeeper Router Router Gateway Server

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The gateway and the gatekeeper have different functions, and are often two separate servers. The gateway is responsible for providing the talk path for the voice call. The gatekeeper is responsible for call setup and monitoring, as well as additional features such as call transfer, conference calling, and call breakdown. There is only one gatekeeper for every zone.

The Internet Gateway Server Gatekeeper Router Router Gateway Server 902-591-1234

Figure 2-2: The gateway is responsible for the talk path

Step 1 - The caller’s IP gateway (originating gateway) receives the phone number being called. The gateway requests the gatekeeper to convert that number into the IP address of the far-end gateway (destination gateway).

The Internet Gateway Server Gatekeeper Router Router Gateway Server IP 24.7.29.16

Figure 2-3: Step 1: The originating gateway converts the called number

Step 2 - Using the IP address, the originating gatekeeper establishes a connection between the originating and destination gateways. This includes: • Exchanging call setup and compatibility information.

The Internet Gateway Server Gatekeeper Router Router Gateway Server Call coming. OK?

Can you do H.450?

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• Performing any option negotiation. The Internet Gateway Server Gatekeeper Router Router Gateway Server H.450 is ok .

Figure 2-5: Step 2: The originating gatekeeper negotiates options

• Completing the security handshake.

Call completed - signal down to phone

Figure 2-6: Step 2: The originating gatekeeper completes the security handshake

Step 3 - The digitizing function of the gateway converts analog signals into a digital format, usually 64-Kbps PCM. Digital signals, such as ISDN and T1/E1, are already in this format. Therefore, the gateway bypasses this step for those signals. The Internet Gateway Server Gatekeeper Router Router Gateway Server 1011010010 Analog to Digital

Figure 2-7: Step 3: Digitizing function converts analog signals to digital

Step 4 varies, depending on the original signal:

• Voice: compression of the signal from 64 Kbps to 10-15 Kbps.

• Fax: demodulation of the signal to its original 2.4-14.4 Kbps digital format. As part of both functions, the gateway puts the digital information into IP packets for transmission to the destination.

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The Internet Gateway Server Gatekeeper Router Router Gateway Server 11011010101101010110110 IP Voice IP Voice IP Voice From voice bitstream To IP packets

Figure 2-8: Step 4: Voice signals compressed and converted to IP packets

The last step actually occurs at the same time as the first four steps. It is a function of the destination gateway. As the destination gateway receives

transmissions, it decompresses voice signals and remodulates fax signals back to their appropriate formats.

The Internet Gateway Server Gatekeeper Router Router Gateway Server IP Voice IP Voice IP Voice

Figure 2-9: Step 5: Destination gateway decompresses voice signals

The gateway then decompresses voice signals to the proper telephony interface via one of the following modes:

• Analog • ISDN • T1/E1

Integrated Services Digital Network (ISDN)

ISDN. A collection of standards that defines interfaces for the operation of digital switching equipment. Instead of using one analog telephone line, ISDN uses three digital channels. Each channel carries voice, video, data, images, or combinations of these. ISDN has two basic formats (for the United States): Basic Rate Interface (BRI) and Primary Rate Interface (PRI).

T1/E1

A digital transmission link consisting of two twisted pairs that can carry at least 24 voice conversations. T1 lines are used for local and long distance transmissions. In Europe, T1 is called E1, and carries 30 DS0 channels for information plus two for signaling.

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Note that the originating interface does not have to match the destination

interface. The gateway-to-gateway connection can usually perform the necessary conversions.

Gateway Functions in the SIP Environment

The SIP and related protocols evolved from the Internet community. SIP is less rigorously defined in terms of required functionality and the physical placement of the functionality. SIP envisions intelligent call agents which use SIP to

perform call setup and administration, and a Media Gateway Control Protocol – currently MGCP – to control media gateways. Work has been done by the Internet and telephony communities to develop a replacement for MGCP called

MeGaCo.

Media Gateway Control Protocol (MGCP)

A master-slave protocol that regulates the functions of the media gateway controller in an SS7 network. Used between a media gateway controller and a media gateway, and with SIP for VoIP implementations. MGCP is described in RFC2705.

Media Gateway Control (MeGaCo)

A protocol in joint development between the SIP and H.323 communities. When implemented, MeGaCo would allow a media gateway to be

controlled by either an SIP controller or an H.323 controller.

The operation of SIP will be described using a toll bypass scenario in which there is a residential gateway, serving a residential analog telephone, which makes a call to a distant trunking gateway. The trunking gateway interconnects with the PSTN in a distant city.

Internet PSTN Location Server Trunking Gateway Trunking Gateway Controller Proxy Server Residential Gateway Controller Residential Gateway T1 Trunks Signaling Links jersey.com

domain destingate.comdomain MGCP MGCP

850-333-1234

Figure 2-10: The trunking gateway connects with the destination PSTN

The residential gateway controller, located in the Internet domain jersey.com, communicates with the residential gateway using MGCP. At initialization, the controller would send a command to the gateway to watch for off-hook, return dial tone and collect digits.

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At the far end of the network, in the destingate.com domain, is a location server, a trunking gateway with its controller, and connections from the trunking gateway to the PSTN.

When the residential phone goes off hook and a call is made, the residential gateway, using MGCP, passes that information to the call agent, which in our scenario is located in the residential gateway controller.

Figure 2-11: The gateway controller and the gateway communicate using MGCP

The call agent in the controller formulates an SIP message to a proxy server. The proxy server contains information such as dialing plans ("what do I do when someone dials 1+850?") and ways of reaching different destinations. The SIP message contains the dialed number and domain, a return address for SIP messages coming back, and a description of the services requested and options using Session Description Protocol (SDP).

Session Description Protocol (SDP)

A protocol describing how information relating to a session (such as type of codec preferred and types of alternate codecs available) is formatted for SIP. Defined by the IETF as RFC2327.

Figure 2-12 illustrates a gateway controller sending an SIP message to a proxy server. Internet jersey.com domain Residential Gateway Controller Residential Gateway Proxy Server SIP:+18503331234@destingate.com; user=phone (Optional parameters)

Figure 2-12: The gateway controller sends an SIP message to the proxy server

The proxy server uses DHCP and DNS procedures to locate a server in destingate.com that provides location services. The proxy server queries the location server for the IP address of the “owner” of 850-333-1234 using SIP, and then receives the address of the proper trunking gateway controller.

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Internet

Location Server jersey.com

domain destingate.comdomain

Residential Gateway Controller Proxy Server Where is 18503331234? IP xxx.yyy.zzz.101

Figure 2-13: The proxy server uses DHCP to find a location server

The proxy server, using SIP, INVITEs the trunking gateway controller to set up a connection. It passes along the optional details. The trunking gateway controller will reply, and modify or negotiate, options as required.

Internet

jersey.com

domain destingate.comdomain

Residential Gateway Controller Proxy Server INVITE 18503331234 SDP parameters passed Return address passed

Trunking Gateway Controller

Figure 2-14: The proxy server uses SIP to INVITE a connection

Once the method of communication between the trunking gateway controller and the proxy has been agreed upon, the trunking gateway controller will command the gateway to formulate a SS7 Initial Address Message into the PSTN, containing call setup information and the trunk selected to carry the call (see Step 1 in Figure 2-20).

Internet PSTN Trunking Gateway Trunking Gateway Controller Proxy Server Residential Gateway Controller Residential Gateway T1 Trunks Signaling Links jersey.com

domain destingate.comdomain MGCP MGCP 850-333-1234 1 2 3 4 5 Media Packets

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Presuming that 1-850-333-1234 is not busy, the PSTN will send ringing to that phone. An SS7 message will come back across the signaling link indicating that ringing has begun. At the same time, audible ring tone from the far end switch will be present on the trunk, and should be passed to the originating phone. Using MGCP, the trunking gateway notifies its controller (Step 2 in Figure 2-20) that ringing has begun. Using SIP, the trunking gateway controller notifies the proxy (Step 3 in Figure 2-20), and the proxy notifies the residential gateway controller (Step 4 in Figure 2-20). The residential gateway controller, using MGCP, then notifies the controller that media packets will begin arriving from the trunking gateway. A one way media session begins from the trunking gateway to the residential gateway, carrying the tone from the PSTN trunk. When 850-333-1234 answers, an SS7 answer message is passed from the PSTN to the trunking gateway. Following essentially the same five steps, both

gateways establish a two-way media session allowing conversation to take place. A similar process is followed when one party disconnects.

Note that the proxy server is still involved in the call. This fact may be desirable if the proxy server will be involved in any call changes, such as may occur if call waiting is invoked or if the proxy server is used to collect billing information. If a proxy server is not required except at call setup, another mode, called redirect, may be used.

In redirect mode, a redirect server takes the place of the proxy server shown. As soon as the redirect server has determined the location of the trunking gateway controller, it passes the information to the residential gateway controller, which then completes call setup and administration directly with the trunking

controller.

Another server, which is not shown in our illustration, is also involved in the architecture. It is called a registration server. When an end point changes locations, thus changing its address, the user registers with the registration server, which in turn notifies the location server of the new location of that end point.

This architecture also allows for application servers, which could provide advanced services such as unified messaging or conferencing.

Voice Compression and

Decompression

For voice calls, the originating media gateway (or gateway server in the H.323 environment) may perform the compression function. This option compresses the voice signal from 64 Kbps to 10-15 Kbps. It then puts this information into IP packets for transmission to the destination gateway, which performs the decompression function.

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Figure 2-16: The originating gateway performs compression for voice calls

The compression and decompression functions are performed by the gateway’s

codec (CODer/DECoder). The codec compresses or decompresses the voice in

real time.

codec

Coder DECoder. A device that converts analog signals to digital signals for transmission on digital lines. At the receiving end, the codec

reconverts the signal from digital to analog.

To avoid conversation delays, the codec divides the voice signal into short fragments of 20 to 40 bytes, called “talk spurts.” Talk spurts are based on models of the human vocal tract. Forty bytes represent 40 milliseconds of speech. This is the longest practical period to use that will not contribute too much to the overall delay.

talk spurt

Voice signal divided into short fragments of 20 to 40 bytes, based on models of the human vocal tract; prevents overall delay of the voice transmission.

Figure 2-17: Voice signals are sent in “talk spurts”

A gateway’s voice packet is constructed as a UDP/IP packet, rather than a TCP/IP packet. This is because TCP/IP tries to correct corrupted or delayed packets by retransmitting them, which causes too much delay for voice transmissions. The UDP/IP, on the other hand, simply discards corrupted or delayed packets, and then replays the previous good packet.

UDP/IP

A part of the TCP/IP suite that is used for unreliable delivery end to end. Commonly used for VoIP environments in which timing is more important than lost packets.

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UDP/IP also uses last-packet replay:

• If a voice packet arrives too late to be of use. • If a packet is lost and never arrives.

The listener rarely notices packet replay if the replay frequency is kept to a low percentage of packets (<5 percent). If the degree of packet corruption is too great, IP telephony relies on the time-tested human correction technique: the listener asks the speaker to repeat what was said.

A compressed talk spurt can be up to 40 bytes and can be sent in one second. In addition to the voice signal, header information must be added. This

information takes up about 7 Kbps.

RTP Header UDP Header IP Header Voice

Header Voice Data

Figure 2-18: A compressed talk spurt with header information is about 7 Kbps

A voice packet has headers from the supporting protocols attached to the voice data from the codec. A 20 byte packet will have roughly 40 bytes of header as overhead.

Fax Demodulation and

Remodulation

For faxes, the originating media gateway (or gateway server in H.323) performs the demodulation function, and the destination gateway performs the

remodulation function. This process delivers the fax in real time.

Figure 2-19: Originating gateway demodulates fax signals

A fax starts as a digital signal. As it leaves the fax machine, it is converted to an analog format. The originating gateway’s demodulation function converts this analog signal back into the fax’s original 2.4-14.4 Kbps digital format. The digital information is put into IP packets and transmitted to the destination gateway.

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A fax is formed by scanning and digital processing, followed by modulation RTP/UDP/IP Protocol Stack IP Fax VoIP Fax Demodulating and Coding VoIP Fax Decoding and Remodulating Transmit Process Fax Signal Receive Process Fax Signal RTP/UDP/IP Protocol Stack IP Fax

Figure 2-20: Fax signals are sent as IP packets

The destination gateway’s remodulation function converts the digital information it receives into an analog signal. It then delivers the analog signal to the remote fax machine. The fax machine is unaware of the demodulation/remodulation process, and thus behaves as if the incoming analog signals came directly from the sending fax machine.

A gateway constructs the fax packet as a UDP/IP packet, rather than a TCP/IP packet. This is because TCP/IP tries to correct transmission errors by

retransmitting the packet, which terminates the fax transmission. The UDP/IP, on the other hand, does not retransmit corrupted packets. The corrupted fax packet might only affect one line of a fax, which is preferable to fax termination. Corrupted and lost fax packets are rare in enterprise IP networks because these networks are very reliable.

Gateway Interfacing

Many different signaling interfaces can be found in today’s networks. The implementers of the telephone system in an enterprise or home selected the interface based on technical, or operational, requirements at the time, along with the economic cost of the service from the telephone company.

Identifying the Telephony Signaling Interface

The VoIP engineer will place gateways in different places in the network and thus will encounter different interfaces. This section reviews in some detail the characteristics of many common interfaces.

While the terms “line” and “trunk” are used loosely within the industry, it is helpful to review the classic definitions as a starting point.

line

A connection between a telephone set and a switch which can handle one conversation.

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trunk

A direct communication line between two switching systems, which can handle one conversation. A central office trunk is the line connecting a PBX and the central office. A tie trunk is the line connecting one PBX to another PBX.

The most common interface is a two-wire, loop start line, used by ordinary analog telephones, fax machines and modems. The two-wire loop start line interface is provided from what is called the “line side” of the telecom switch, and uses what is called “line side” signaling. PBXs with their analog station cards, which can be used for modems and fax machines, also use this type of signaling. Telco Switch Modem Line Side Connection Trunk Side

Figure 2-21: Line side signaling is used by telco switches and PBXs

Many PBXs today have only one group of trunks which connect to the telco switch. These trunks handle both incoming and outgoing calls. They use line-side signaling, but are ground start rather than loop start because ground start lines are more reliable than loop start in two-way operations.

Many PBXs use a feature called Direct Inward Dialing (DID), in which the telco switch sends the last three to five digits of the dialed number to the PBX, which in turn completes a call directly to a station without the need for an attendant or auto attendant. These connections are on the “trunks side” of the telco switch.

PBX _{Telco Switch} DID Trunks Dial 9 Trunks Trunk Side Station Side Line Side Trunk Side

Voice over IP (VoIP) Essentials: