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CCNP ONT Official
Exam Certification Guide

Amir S. Ranjbar, CCIE No. 8669

Cisco Press
800 East 96th Street
Indianapolis, IN 46240 USA
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CCNP ONT Official Exam Certification Guide
Amir S. Ranjbar, CCIE No. 8669
Copyright© 2007 Cisco Systems, Inc.
Published by:
Cisco Press
800 East 96th Street
Indianapolis, IN 46240 USA
All rights reserved. No part of this book may be reproduced or transmitted in any form or by any means, electronic or mechanical,
including photocopying, recording, or by any information storage and retrieval system, without written permission from the pub-
lisher, except for the inclusion of brief quotations in a review.
Printed in the United States of America 1 2 3 4 5 6 7 8 9 0
First Printing: May 2007
Library of Congress Cataloging-in-Publication data is on file.
ISBN-10: 1-58720-176-3
ISBN-13: 978-1-58720-176-9

Warning and Disclaimer
This book is designed to provide information about the topics covered on the Optimizing Converged Cisco Networks (642-845
ONT) CCNP exam. Every effort has been made to make this book as complete and as accurate as possible, but no warranty or fitness
is implied.
The information is provided on an “as is” basis. The author, Cisco Press, and Cisco Systems, Inc. shall have neither liability nor
responsibility to any person or entity with respect to any loss or damages arising from the information contained in this book or from
the use of the discs or programs that may accompany it.
The opinions expressed in this book belong to the author and are not necessarily those of Cisco Systems, Inc.

Trademark Acknowledgments
All terms mentioned in this book that are known to be trademarks or service marks have been appropriately capitalized. Cisco Press
or Cisco Systems, Inc. cannot attest to the accuracy of this information. Use of a term in this book should not be regarded as affecting
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Feedback Information
At Cisco Press, our goal is to create in-depth technical books of the highest quality and value. Each book is crafted with care and pre-
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Publisher: Paul Boger Cisco Representative: Anthony Wolfenden
Associate Publisher: David Dusthimer Cisco Press Program Manager: Jeff Brady
Executive Editor: Mary Beth Ray Technical Editors: Dave Minutella, Mike Valentine
Managing Editor: Patrick Kanouse Book and Cover Designer: Louisa Adair
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Copy Editor: Karen A. Gill
Publishing Coordinator: Vanessa Evans
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About the Author
Amir S. Ranjbar, CCIE No. 8669, is an internetworking trainer and consultant. Born in Tehran,
Iran, he moved to Canada in 1983. He received his bachelor’s degree in computer science (1989)
and master of science degree in knowledge-based systems (1991) from the University of Guelph
in Guelph, Ontario, Canada. After graduation, Amir worked as a programmer/analyst for Statistics
Canada until 1995 when he was hired by Digital Equipment Corporation as a certified Microsoft
trainer. After performing training on Microsoft Backoffice products such as Windows NT,
Exchange Server, and Systems Management Server for three years, he shifted his focus to Cisco
Systems. In 1998, he joined GEOTRAIN Corporation, which was later acquired by Global
Knowledge Network, and worked for them as a full-time Certified Cisco Systems Instructor until
2005. In October 2005, Amir started his own business (AMIRACAN Inc.) in the field of
internetwork consulting, but his major activity is still conducting training for Global Knowledge
Network on a contractual basis. His areas of specialty are MPLS, BGP, QoS, VoIP, and advanced
routing and switching. Amir’s e-mail address is aranjbar@rogers.com

About the Contributing Author
Troy Houston, CCNP, CCDP, and CCIE-written, independently provides contracted business and
knowledge solutions to enterprise customers in the Mid-Atlantic area. The first half of his career
was in the Aerospace industry where he gained extensive RF knowledge making him the WLAN
SME today. Over the past 10 years, Troy has planned, designed, implemented, operated, and
troubleshot LANs, WANs, MANs, and WLANs. He attained his bachelor of science degree in
management of information systems from Eastern University. Additionally, he is an inventor and
holds a patent for one of his many ideas. Formerly in the military, Troy returned to the military
on a reserve basis after 9/11. He provides the Air Force Reserves his skills and knowledge as a
Computers-Communications Systems Specialist (3C0). He can be contacted at
troy@houstonshome.com.
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About the Technical Reviewers
Dave Minutella (CCNP, CCDP, CCSP, INFOSEC, CISSP, MCSA, MCDST, CTP, Security+,
Network +, A+) has been working in the IT and telecom industry for more than 12 years. He
currently serves as vice president of educational services for TechTrain/The Training Camp. Prior
to that, he was the lead Cisco instructor, primarily teaching CCNA, CCDA, and CCNP courses.
Dave is also the technical author of CSVPN Exam Cram 2 and coauthor of CCNA Exam Prep 2
from Que Publishing, and he is the present Cisco certifications expert for SearchNetworking.com’s
Ask the Networking Expert panel.

Mike Valentine has 12 years of experience in the IT field, specializing in network design and
installation. His projects include the installation of network services and infrastructure at the
largest private aircraft maintenance facility in Canada, Cisco Unified CallManager implementations
for small business clients in southwest Florida, and implementation of network mergers and
development for Prospera Credit Union in British Columbia. He now heads up his own network
consulting company near Vancouver, BC, providing contract Cisco certification instruction and
network infrastructure consulting services to clients throughout North America.

Mike is the senior Cisco instructor for The Training Camp. His diverse background and exceptional
instructional skills make him a consistent favorite with students. In addition to providing training
and developing courseware for The Training Camp, he is the senior network engineer for The
Client Server, Inc. in Bonita Springs, Florida, responsible for network infrastructure, security,
and VoIP projects. Mike holds a Bachelor of Arts in anthropology, in addition to the following
certifications: MCP+i, MCSA, MCSE (Security, Sec+, Net+), CCDA, CCNP, IPTX, C|EH, and
CTP.

Mike coauthored the popular CCNA Exam Cram 2, published in December 2005.
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Dedications
This book is dedicated to my wife, Elke Haugen-Ranjbar, whose love, hard work, understanding,
and support have made my home a dream come true. Should my children Thalia, Ariana, and
Armando choose a life partner when they grow up, I wish they will make as good of a choice as I
did.

—Amir Ranjbar
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Acknowledgments
I would like to thank the technical editors, Dave and Mike, for their valuable comments and
feedback.

Special thanks to Mary Beth Ray for her patience and understanding, and to Andrew Cupp for a
well-done job.

This book is the product of the hard work of a team and not just a few individuals. Managers,
editors, coordinators, and designers: All of you, please accept my most sincere appreciation for
your efforts and professional input.
viii

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Contents at a Glance
Foreword xvii
Introduction xviii
Part I Voice over IP 3
Chapter 1 Cisco VoIP Implementations 5
Part II Quality of Service 55
Chapter 2 IP Quality of Service 57
Chapter 3 Classification, Marking, and NBAR 93
Chapter 4 Congestion Management and Queuing 123
Chapter 5 Congestion Avoidance, Policing, Shaping, and Link
Efficiency Mechanisms 149
Chapter 6 Implementing QoS Pre-Classify and Deploying End-to-End QoS 177
Chapter 7 Implementing AutoQoS 201
Part III Wireless LAN 229
Chapter 8 Wireless LAN QoS Implementation 231
Chapter 9 Introducing 802.1x and Configuring Encryption and Authentication
on Lightweight Access Points 255
Chapter 10 WLAN Management 287
Part IV Appendix 319
Appendix A Answers to the “Do I Know This Already?” Quizzes and Q&A Sections 321
Index 354
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Contents
Foreword xvii
Introduction xviii
Part I Voice over IP 3
Chapter 1 Cisco VoIP Implementations 5
“Do I Know This Already?” Quiz 5
Foundation Topics 10
Introduction to VoIP Networks 10
Benefits of Packet Telephony Networks 10
Packet Telephony Components 11
Analog Interfaces 13
Digital Interfaces 14
Stages of a Phone Call 15
Distributed Versus Centralized Call Control 16
Digitizing and Packetizing Voice 19
Basic Voice Encoding: Converting Analog to Digital 19
Basic Voice Encoding: Converting Digital to Analog 20
The Nyquist Theorem 21
Quantization 22
Compression Bandwidth Requirements and Their Comparative Qualities 24
Digital Signal Processors 25
Encapsulating Voice Packets 27
End-to-End Delivery of Voice 27
Protocols Used in Voice Encapsulation 30
Reducing Header Overhead 32
Bandwidth Calculation 34
Impact of Voice Samples and Packet Size on Bandwidth 34
Data Link Overhead 37
Security and Tunneling Overhead 37
Calculating the Total Bandwidth for a VoIP Call 39
Effects of VAD on Bandwidth 41
Implementing VoIP Support in an Enterprise Network 42
Enterprise Voice Implementations 42
Voice Gateway Functions on a Cisco Router 44
Cisco Unified CallManager Functions 45
Enterprise IP Telephony Deployment Models 46
Single-Site Model 46
Multisite with Centralized Call Processing Model 46
Multisite with Distributed Call Processing Model 47
Clustering over WAN Model 48
Identifying Voice Commands in IOS Configurations 48
Call Admission Control (CAC) 49
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Foundation Summary 50
Q&A 52
Part II Quality of Service 55
Chapter 2 IP Quality of Service 57
“Do I Know This Already?” Quiz 57
Foundation Topics 62
Introduction to QoS 62
Converged Network Issues Related to QoS 62
Available Bandwidth 63
End-to-End Delay 64
Delay Variation 65
Packet Loss 66
Definition of QoS and the Three Steps to Implementing It 68
Implementing QoS 69
Identifying and Comparing QoS Models 72
Best-Effort Model 72
Integrated Services Model 73
Differentiated Services Model 74
QoS Implementation Methods 76
Legacy Command-Line Interface (CLI) 76
Modular QoS Command-Line Interface (MQC) 76
AutoQoS 79
Router and Security Device Manager (SDM) QoS Wizard 81
Foundation Summary 89
Q&A 91
Chapter 3 Classification, Marking, and NBAR 93
“Do I Know This Already?” Quiz 93
Foundation Topics 97
Classification and Marking 97
Layer 2 QoS: CoS on 802.1Q/P Ethernet Frame 98
Layer 2 QoS: DE and CLP on Frame Relay and ATM (Cells) 99
Layer 2 1/2 QoS: MPLS EXP Field 100
The DiffServ Model, Differentiated Services Code Point (DSCP), and Per-Hop Behavior
(PHB) 100
IP Precedence and DSCP 102
QoS Service Class 106
Trust Boundaries 108
Network Based Application Recognition (NBAR) 110
Cisco IOS Commands to Configure NBAR 112
Foundation Summary 118
Q&A 120
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Chapter 4 Congestion Management and Queuing 123
“Do I Know This Already?” Quiz 123
Foundation Topics 127
Introduction to Congestion Management and Queuing 127
First-In-First-Out, Priority Queuing, Round-Robin, and Weighted Round-Robin Queuing 130
Weighted Fair Queuing 132
WFQ Classification and Scheduling 133
WFQ Insertion and Drop Policy 135
Benefits and Drawbacks of WFQ 135
Configuring and Monitoring WFQ 135
Class-Based Weighted Fair Queuing 138
Classification, Scheduling, and Bandwidth Guarantee 139
Benefits and Drawbacks of CBWFQ 140
Configuring and Monitoring CBWFQ 141
Low-Latency Queuing 142
Benefits of LLQ 144
Configuring and Monitoring LLQ 144
Foundation Summary 146
Q&A 147
Chapter 5 Congestion Avoidance, Policing, Shaping, and Link
Efficiency Mechanisms 149
“Do I Know This Already?” Quiz 149
Foundation Topics 153
Congestion Avoidance 153
Tail Drop and Its Limitations 153
Random Early Detection 154
Weighted Random Early Detection 156
Class-Based Weighted Random Early Detection 158
Configuring CBWRED 158
Traffic Shaping and Policing 163
Measuring Traffic Rates 165
Cisco IOS Policing and Shaping Mechanisms 167
Link Efficiency Mechanisms 167
Layer 2 Payload Compression 168
Header Compression 169
Link Fragmentation and Interleaving 171
Applying Link Efficiency Mechanisms 171
Foundation Summary 172
Q&A 175
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Chapter 6 Implementing QoS Pre-Classify and Deploying End-to-End QoS 177
“Do I Know This Already?” Quiz 177
Foundation Topics 180
Implementing QoS Pre-Classify 180
Virtual Private Networks (VPN) 180
QoS Pre-Classify Applications 181
QoS Pre-Classification Deployment Options 183
Deploying End-to-End QoS 185
QoS Service Level Agreements (SLAs) 186
Enterprise Campus QoS Implementations 188
WAN Edge QoS Implementations 190
Control Plane Policing (CoPP) 192
Foundation Summary 194
Q&A 198
Chapter 7 Implementing AutoQoS 201
“Do I Know This Already?” Quiz 201
Foundation Topics 205
Introducing AutoQoS 205
Implementing and Verifying AutoQoS 207
Two-Step Deployment of AutoQoS Enterprise on Routers 209
Deploying AutoQoS VoIP on IOS-Based Catalyst Switches 210
Verifying AutoQoS on Cisco Routers and IOS-Based Catalyst Switches 212
AutoQoS Shortcomings and Remedies 215
Automation with Cisco AutoQoS 215
Common AutoQoS Problems 218
Interpreting and Modifying AutoQoS Configurations 219
Foundation Summary 222
Q&A 227
Part III Wireless LAN 229
Chapter 8 Wireless LAN QoS Implementation 231
“Do I Know This Already?” Quiz 231
Foundation Topics 235
The Need for Wireless LAN QoS 235
WLAN QoS Description 237
Split MAC Architecture and Light Weight Access Point 238
Current Wireless LAN QoS Implementation 239
Configuring Wireless LAN QoS 243
Foundation Summary 247
Q&A 252
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Chapter 9 Introducing 802.1x and Configuring Encryption and Authentication
on Lightweight Access Points 255
“Do I Know This Already?” Quiz 255
Foundation Topics 258
Overview of WLAN Security 258
WLAN Security Issues 258
Evolution of WLAN Security Solutions 259
802.1x and EAP Authentication Protocols 260
EAP Authentication Protocols 262
Cisco LEAP 262
EAP-FAST 264
EAP-TLS 266
PEAP 267
WPA, 802.11i, and WPA2 269
Configuring Encryption and Authentication on Lightweight Access Points 272
Open Authentication 272
Static WEP Authentication 273
WPA Preshared Key 274
Web Authentication 276
802.1x Authentication 278
Foundation Summary 281
Q&A 285
Chapter 10 WLAN Management 287
“Do I Know This Already?” Quiz 287
Foundation Topics 291
The Need for WLAN Management 291
Cisco Unified Wireless Networks 291
Cisco WLAN Implementation 292
WLAN Components 294
CiscoWorks Wireless LAN Solution Engine 295
WLSE Software Features 295
WLSE Key Benefits 296
CiscoWorks WLSE and WLSE Express 296
Simplified WLSE Express Setup 297
WLSE Configuration Templates 298
WLSE IDS Features 298
WLSE Summary 298
Cisco Wireless Control System 299
WCS Location Tracking Options 300
WCS Base Software Features 300
WCS Location Software Features 301
WCS Location + 2700 Series Wireless Location Appliance Features 301
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WCS System Features 301
Cisco WCS User Interface 302
Cisco WCS System Requirements 302
WCS Summary Pages 303
Wireless Location Appliance 304
Wireless Location Appliance Architecture 305
Wireless Location Appliance Applications 305
WCS Configuration Examples 306
WCS Login Steps 306
Changing the Root Password 306
Adding a Wireless LAN Controller 307
Configuring Access Points 308
WCS Map 309
Adding a Campus Map 309
Adding a New Building 310
Rogue Access Point Detection 312
Rogue Access Point Alarms 312
Rogue Access Point Location 313
Foundation Summary 314
Q&A 317
Part IV Appendix 319
Appendix A Answers to the “Do I Know This Already?” Quizzes and Q&A Sections 321
Index 354
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Icons Used in This Book

Terminal PC Laptop File Router
Server

Si
V
Multilayer Core ATM Switch Access Voice-Enabled
Switch Switch Switch Router

IP PBX

CallManager IP Phones PBX Phones Access Point PBX

Network Cloud Line: Ethernet Line: Serial Line: Switched Serial

Command Syntax Conventions
The conventions used to present command syntax in this book are the same ones used in the IOS
Command Reference. The Command Reference describes these conventions as follows:

■ Boldface indicates commands and keywords that are entered literally as shown. In actual
configuration examples and output (not general command syntax), boldface indicates
commands that are manually input by the user (such as a show command).
■ Italics indicate arguments for which you supply actual values.
■ Vertical bars (|) separate alternative, mutually exclusive elements.
■ Square brackets [ ] indicate optional elements.
■ Braces { } indicate a required choice.
■ Braces within brackets [{ }] indicate a required choice within an optional element.
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Foreword
CCNP ONT Official Exam Certification Guide is an excellent self-study resource for the 642-845
ONT exam. Passing the exam certifies that the successful candidate has important knowledge and
skills in optimizing and providing effective QoS techniques for converged networks. Passing the
exam is one of the requirements for the Cisco Certified Network Professional (CCNP)
certification.

Gaining certification in Cisco technology is key to the continuing educational development of
today’s networking professional. Through certification programs, Cisco validates the skills and
expertise required to effectively manage the modern enterprise network.

Cisco Press exam certification guides and preparation materials offer exceptional—and flexible—
access to the knowledge and information required to stay current in your field of expertise, or to
gain new skills. Whether used as a supplement to more traditional training or as a primary source
of learning, these materials offer users the information and knowledge validation required to gain
new understanding and proficiencies.

Developed in conjunction with the Cisco certifications and training team, Cisco Press books are
the only self-study books authorized by Cisco and offer students a series of exam practice tools
and resource materials to help ensure that learners fully grasp the concepts and information
presented.

Additional authorized Cisco instructor-led courses, e-learning, labs, and simulations are available
exclusively from Cisco Learning Solutions Partners worldwide. To learn more, visit
http://www.cisco.com/go/training.

I hope that you find these materials to be an enriching and useful part of your exam preparation.

Erik Ullanderson
Manager, Global Certifications
Learning@Cisco
March 2007
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Introduction
Professional certifications have been an important part of the computing industry for many years
and will continue to become more important. There are many reasons for these certifications, but
the most popularly cited reason is that of credibility. All other considerations held equal, the
certified employee/consultant/job candidate is considered more valuable than one who is not.

Goals and Methods
The most important and somewhat obvious goal of this book is to help you pass the Optimizing
Converged Cisco Networks (ONT) exam 642-845. In fact, if the primary objective of this book
were different, the book title would be misleading; however, the methods used in this book to help
you pass the ONT exam are also designed to make you much more knowledgeable about how to
do your job. Although this book and the accompanying CD-ROM together have more than enough
questions to help you prepare for the actual exam, the method in which they are used is not to
simply make you memorize as many questions and answers as you possibly can.

One key methodology used in this book and on the CD-ROM is to help you discover the exam
topics that you need to review in more depth, to help you fully understand and remember those
details, and to help you prove to yourself that you have retained your knowledge of those topics.
Therefore, this book does not try to help you pass by memorization; it helps you truly learn and
understand the topics. The ONT exam is just one of the foundation topics in the CCNP
certification, and the knowledge contained within is vitally important to considering yourself a
truly skilled routing/switching engineer or specialist. This book would do you a disservice if it did
not attempt to help you learn the material. To that end, this book will help you pass the ONT exam
by using the following methods:

■ Helping you discover which test topics you have not mastered
■ Providing explanations and information to fill in your knowledge gaps
■ Supplying exercises and scenarios that enhance your ability to recall and deduce the
answers to test questions
■ Providing practice exercises on the topics and the testing process via test questions on
the CD-ROM

Who Should Read This Book?
This book is not designed to be a general networking topics book, although you can use it for that
purpose. This book is intended to tremendously increase your chances of passing the CCNP ONT
exam. Although you can achieve other objectives from using this book, the book was written with
one goal in mind: to help you pass the exam.
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Why should you want to pass the CCNP ONT exam? Because it is one of the milestones toward
getting the CCNP certification—no small feat in itself. What would achieving CCNP mean to you?
A raise, a promotion, or recognition? How about to enhance your résumé? Maybe it is to
demonstrate that you are serious about continuing the learning process and not content to rest on
your laurels. Or perhaps it is to please your reseller-employer, who needs more certified employees
for a higher discount from Cisco. Or it could be for one of many other reasons.

Strategies for Exam Preparation
The strategy that you use for CCNP ONT might be slightly different from strategies that other
readers use, mainly based on the skills, knowledge, and experience you already have obtained. For
instance, if you have attended the ONT course, you might take a different approach than someone
who learned VoIP or QoS via on-the-job training. Regardless of the strategy you use or the
background you have, this book is designed to help you get to the point where you can pass the
exam with the least amount of time required. For instance, it is unnecessary for you to read a
chapter if you fully understand it already. However, many people like to make sure that they truly
know a topic and thus read over material that they already know. Several book features, such as
the “Do I Know This Already?” quizzes, will help you gain the confidence you need to be
convinced that you know some material already and to help you know what topics you need to
study more.

The following are some additional suggestions for using this book and preparing for the exam:

■ Familiarize yourself with the exam objectives in Table I-1 and thoroughly read the chapters on
topics that you are not familiar with. Use the assessment tools provided in this book to identify
areas where you need additional study. The assessment tools include the “Do I Know This
Already?” quizzes, the “Q&A” questions, and the sample exam questions on the CD-ROM.
■ Take all quizzes in this book and review the answers and the answer explanations. It is not
enough to know the correct answer; you also need to understand why it is correct and why the
others are incorrect. Retake the chapter quizzes until you pass with 100 percent.
■ Take the CD-ROM test in this book and review the answers. Use your results to identify areas
where you need additional preparation.
■ Review other documents, RFCs, and the Cisco website for additional information. If this book
references an outside source, it’s a good idea to spend some time looking at it.
■ Review the chapter questions and CD-ROM questions the day before your scheduled test.
Review each chapter’s “Foundation Summary” when you make your final preparations.
■ On the test date, arrive at least 20 minutes before your test time. This plan gives you time to
register and glance through your notes before the test without feeling rushed or anxious.
■ If you are not sure of an answer to a question, attempt to eliminate incorrect answers.
■ You might need to spend more time on some questions than others. Remember, you have an
average of 1 minute to answer each question.
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How This Book Is Organized
Although you can read this book cover to cover if you want to, it is designed to be flexible and
allow you to easily move between chapters and sections of chapters to cover just the material that
you need more work with. Chapter 1 of this book matches the “Cisco VoIP Implementations”
module of the Cisco ONT official training curriculum. Chapter 2 of this book matches the
“Introduction to IP QoS” module of the Cisco ONT official training curriculum. Chapters 3, 4, 5,
and 6 of this book match the “Implement the DiffServ QoS Model” module of the Cisco ONT
official training curriculum. Chapter 7 of this book matches the “Implementing AutoQoS” module
of the Cisco ONT official training curriculum. Finally, Chapters 8, 9, and 10 of this book match
the “Implement Wireless Scalability” module of the Cisco ONT official training curriculum.

Following is a short description of the topics covered in this book:

■ Chapter 1, “Cisco VoIP Implementations”—This chapter describes the benefits of, and the
basic components of, VoIP networks. Conversion of analog voice signal to digital voice signal
and vice versa, plus encapsulation of voice for transport across an IP network, and calculating
bandwidth requirements for VoIP are also discussed in this chapter. The final section of this
chapter identifies the components necessary for VoIP support in an enterprise, describes the
main IP telephony deployment models, and defines call admission control.
■ Chapter 2, “IP Quality of Service”—This chapter provides the essential background,
definitions, and concepts for learning IP Quality of Service. First, QoS is defined, the main
issues that must be addressed in a converged network are presented, and the key steps in
implementing a QoS policy in a network are described. The three main QoS models and the
key features, merits, and drawbacks of each model are discussed next. The last part of this
chapter explains the legacy Command Line Interface (CLI), Modular Quality of Service
Command Line Interface (MQC), Cisco AutoQoS, and Cisco Router and Security Device
Manager (SDM) QoS Wizard. The advantages and disadvantages of each of these QoS
implementation methods are compared.
■ Chapter 3, “Classification, Marking, and NBAR”—This chapter defines classification and
marking, and presents the markings that are available at data link and network layers. QoS
service classes and how they can be used to create a service policy throughout a network are
described next, followed by a discussion on Network trust boundaries. Network Based
Application Recognition (NBAR), as well as Packet Description Language Modules
(PDLM), are described next. The chapter concludes by presenting the IOS commands
required to configure NBAR.
■ Chapter 4, “Congestion Management and Queuing”—This chapter starts by defining what
congestion is and why congestion happens. Next, the need for queuing or congestion manage-
ment is explained and the router queuing components are listed and described. The rest if this
chapter is dedicated to explaining and providing configuration and monitoring commands for
queuing methods, namely FIFO, PQ, RR, WRR, WFQ, Class-Based WFQ, and LLQ.
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■ Chapter 5, “Congestion Avoidance, Policing, Shaping, and Link Efficiency
Mechanisms”—This chapter provides an overview of three main QoS concepts: congestion
avoidance, traffic shaping and policing, and link efficiency mechanisms. WRED and class-
based WRED are the main mechanisms covered. Traffic shaping and policing concepts are
explained in the next section; you will learn the purpose of these mechanisms and where it is
appropriate to use them. Different compression techniques, plus the concept of link fragmentation
and interleaving are the topics of discussion in the third and final section of this chapter.
■ Chapter 6, “Implementing QoS Pre-Classify and Deploying End-to-End QoS”—This
chapter describes the concept of QoS pre-classify, and how it is used to ensure that IOS QoS
features work in conjunction with tunneling and encryption. The second part of this chapter
deals with the topics related to deploying end-to-end QoS. The final part of this chapter
discusses the concept of control plane policing.
■ Chapter 7, “Implementing AutoQoS”—This chapter explains AutoQoS, including
discussions on AutoQoS VoIP and AutoQoS Enterprise. It also presents the key elements of
QoS deployment, protocol discovery with NBAR, and AutoQoS deployment restrictions.
Configuring and verifying AutoQoS on routers and switches is another major topic of this
chapter. A discussion on common AutoQoS problems and suggestions on mitigating those
problems by modifying the active AutoQoS configuration completes this chapter.
■ Chapter 8, “WLAN QoS Implementation”—This chapter starts by explaining the need for
QoS in wireless LANs and describing WLAN QoS, which is work in progress. WLAN QoS
implementation between client and wireless access point, between access point and
controller, and between controller and Ethernet switch are described next. Configuring
WLAN QoS through defining QoS profiles and WLAN IDs on wireless controllers is the last
topic of this chapter.
■ Chapter 9, “Introducing 802.1x and Configuring Encryption and Authentication on
Lightweight Access Points”—The focus of this chapter is wireless security. It starts by
explaining the need for wireless security and describing WLAN security. Next, 802.11x,
LEAP, EAP (FAST and TLS), and PEAP are briefly introduced, and the concept of WiFi
protected access (WPA) is explained. The final section of this chapter discusses how
encryption and authentication on lightweight access points is configured.
■ Chapter 10, “WLAN Management”—This chapter begins by describing the Cisco unified
wireless networks: the business drivers, the elements and, of course, the Cisco implementation
model and its components. The second part of this chapter describes Cisco Wireless LAN
Solution Engine (WLSE) and WLSE Express and their features and benefits; it also presents
a quick lesson on WLSE Express setup. The final parts of this chapter discuss Cisco Wireless
Control Systems (WCS base and location software and system features), Cisco Wireless
Location Appliance (architecture and applications), and rogue access point detection.
■ Appendix A, “Answers to the ”Do I Know This Already?” Quizzes and Q&A Sections”—
This appendix provides the answers and explanations to all of the questions in the book.
1763fm.book Page xxii Monday, April 23, 2007 8:58 AM

xxii

Features of This Book
This book features the following:

■ “Do I Know This Already?” Quizzes—Each chapter begins with a quiz that helps you
determine the amount of time you need to spend studying that chapter. If you follow the
directions at the beginning of the chapter, the “Do I Know This Already?” quiz directs you to
study all or particular parts of the chapter.
■ Foundation Topics—These are the core sections of each chapter. They explain the protocols,
concepts, and configuration for the topics in that chapter. If you need to learn about the topics
in a chapter, read the “Foundation Topics” section.
■ Foundation Summaries—Near the end of each chapter, a summary collects the most
important information from the chapter. The “Foundation Summary” section is designed to
help you review the key concepts in the chapter if you scored well on the “Do I Know This
Already?” quiz. This section is an excellent tool for last-minute review.
■ Q&A—Each chapter ends with a “Q&A” section that forces you to exercise your recall of the
facts and processes described inside that chapter. The questions are generally harder than the
actual exam. These questions are a great way to increase the accuracy of your recollection of
the facts.
■ CD-ROM Test Questions—Using the test engine on the CD-ROM, you can take simulated
exams. You can also choose to be presented with several questions on an objective that you
need more work on. This testing tool gives you practice to make you more comfortable when
you actually take the CCNP exam.

ONT Exam Topics
Cisco lists the topics of the ONT exam on its website at www.cisco.com/web/learning/le3/
current_exams/642-845.html. The list provides key information about what the test covers. Table
I-1 lists the ONT exam topics and the corresponding parts in this book that cover those topics.
Each part begins with a list of the topics covered. Use these references as a road map to find the
exact materials you need to study to master the ONT exam topics. Note, however, that because all
exam information is managed by Cisco Systems and is therefore subject to change, candidates
should continually monitor the Cisco Systems site for course and exam updates at
www.cisco.com.
1763fm.book Page xxiii Monday, April 23, 2007 8:58 AM

xxiii

Table I-1 ONT Topics and the Parts of the book Where They Are Covered

Topic Part
Describe Cisco VoIP implementations.
Describe the functions and operations of a VoIP network (e.g., packetization, bandwidth I
considerations, CAC, etc.).
Describe and identify basic voice components in an enterprise network (e.g. Gatekeepers, I
Gateways, etc.).
Describe QoS considerations.
Explain the necessity of QoS in converged networks (e.g., bandwidth, delay, loss, etc.). II
Describe strategies for QoS implementations (e.g. QoS Policy, QoS Models, etc.). II
Describe DiffServ QoS implementations.
Describe classification and marking (e.g., CoS, ToS, IP Precedence, DSCP, etc.). II
Describe and configure NBAR for classification. II
Explain congestion management and avoidance mechanisms (e.g., FIFO, PQ, WRR, II
WRED, etc.).
Describe traffic policing and traffic shaping (i.e., traffic conditioners). II
Describe Control Plane Policing. II
Describe WAN link efficiency mechanisms (e.g., Payload/Header Compression, MLP with II
interleaving, etc.).
Describe and configure QoS Pre-Classify. II
Implement AutoQoS.
Explain the functions and operations of AutoQoS. II
Describe the SDM QoS Wizard. II
Configure, verify, and troubleshoot AutoQoS implementations (i.e., MQC). II
Implement WLAN security and management.
Describe and configure wireless security on Cisco Clients and APs (e.g., SSID, WEP, III
LEAP, etc.).
Describe basic wireless management (e.g., WLSE and WCS). Configure and verify basic III
WCS configuration (i.e., login, add/review controller/AP status, security, and import/review
maps).
Describe and configure WLAN QoS. III
1763fm.book Page 2 Monday, April 23, 2007 8:58 AM

This part covers the following ONT exam topics. (To view the ONT exam
overview, visit http://www.cisco.com/web/learning/le3/current_exams/642-
845.html.)

■ Describe the functions and operations of a VoIP network (e.g., packetization,
bandwidth considerations, CAC, etc.).
■ Describe and identify basic voice components in an enterprise network (e.g.,
Gatekeepers, Gateways, etc.).
1763fm.book Page 3 Monday, April 23, 2007 8:58 AM

Part I: Voice over IP

Chapter 1 Cisco VoIP Implementations
1763fm.book Page 4 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:

■ Introduction to VoIP Networks

■ Digitizing and Packetizing Voice

■ Encapsulating Voice Packets

■ Bandwidth Calculation

■ Implementing VoIP Support in an Enterprise
Network
1763fm.book Page 5 Monday, April 23, 2007 8:58 AM

CHAPTER 1
Cisco VoIP Implementations

This chapter describes Cisco Voice over IP (VoIP) implementations. Expect to see several exam
questions based on the material in this chapter.

This chapter has five major topics. The first topic helps you understand the basic components
of VoIP networks and the benefits of VoIP networks. The second topic is about converting
an analog voice signal to a digital voice signal and the concepts of sampling, quantization,
compression, and digital signal processors (DSP). The third section discusses encapsulating
voice for transport across an IP network using Real-Time Transport Protocol. The fourth focuses
on calculating bandwidth requirements for VoIP, considering different data link layer possibilities.
The fifth section identifies the components necessary for VoIP support in an enterprise,
describes the main IP Telephony deployment models, and briefly defines call admission control.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read this entire chapter. The 20-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 1-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 1-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“Introduction to VoIP Networks” 1–5
“Digitizing and Packetizing Voice” 6–10
“Encapsulating Voice Packets” 11–12
“Bandwidth Calculation” 13–17
“Implementing VoIP Support in an Enterprise Network” 18–20
Total Score (20 possible)
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6 Chapter 1: Cisco VoIP Implementations

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this chapter.
If you do not know the answer to a question or are only partially sure of the answer, mark this
question wrong for purposes of the self-assessment. Giving yourself credit for an answer you
correctly guess skews your self-assessment results and might provide you with a false sense of
security.

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 15 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 16–17 overall score—Begin with the “Foundation Summary” section and then follow up
with the “Q&A” section at the end of the chapter.

■ 18 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which one of the following is not a benefit of VoIP compared to traditional circuit-switched
telephony?
a. Consolidated network expenses
b. Improved employee productivity
c. Access to new communication devices
d. Higher voice quality
2. Which one of the following is not considered a packet telephony device?
a. IP phone
b. Call agent
c. PBX
d. Gateway
3. Which one of the following is not an analog interface?
a. FXO
b. BRI
c. FXS
d. E&M
1763fm.book Page 7 Monday, April 23, 2007 8:58 AM

“Do I Know This Already?” Quiz 7

4. Which one of the following digital interface descriptions is incorrect?
a. T1 CAS with 30 voice channels
b. T1 CCS with 23 voice channels
c. BRI with 2 voice channels
d. E1 with 30 voice channels
5. Which one of the following is not one of the three stages of a phone call?
a. Call setup
b. Call maintenance
c. Call teardown
d. Call processing
6. Which one of the following is not a step in analog-to-digital signal conversion?
a. Sampling
b. Quantization
c. Encoding
d. Decompression
7. Based on the Nyquist theorem, what is the appropriate sampling rate for an analog voice
signal with a maximum frequency of 4000 Hz?
a. 8800
b. 8000
c. 4000
d. 4400
8. Which of the following accurately describes the 8-bit encoding?
a. 1 polarity bit, 3 segment bits, 4 step bits
b. 1 polarity bit, 4 segment bits, 3 step bits
c. 4 polarity bits, 3 segment bits, 1 step bit
d. 3 polarity bits, 4 segment bits, 1 step bit
9. Which of the following codec descriptions is incorrect?
a. G.711 PCM 64 Kbps
b. G.726 ADPCM 8 Kbps
c. G.728 LD-CELP 16 Kbps
d. G.729 CS-ACELP 8 Kbps
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8 Chapter 1: Cisco VoIP Implementations

10. Which of the following is not a telephony application that requires usage of a DSP?
a. Voice termination
b. Conferencing
c. Packetization
d. Transcoding
11. Which of the following is a false statement?
a. Voice needs the reliability that TCP provides.
b. Voice needs the reordering that RTP provides.
c. Voice needs the time-stamping that RTP provides.
d. Voice needs the multiplexing that UDP provides.
12. Which of the following correctly specifies the header sizes for RTP, UDP, and IP?
a. 8 bytes of RTP, 12 bytes of UDP, and 20 bytes of IP
b. 20 bytes of RTP, 12 bytes of UDP, and 8 bytes of IP
c. 8 bytes of RTP, 20 bytes of UDP, and 12 bytes of IP
d. 12 bytes of RTP, 8 bytes of UDP, and 20 bytes of IP
13. Which of the following is not a factor influencing VoIP media bandwidth?
a. Packet rate
b. Packetization size
c. TCP overhead
d. Tunneling or security overhead
14. If 30 ms of voice is packetized, what will the packet rate be?
a. 50 packets per second
b. 60 packets per second
c. 30 packets per second
d. 33.33 packets per second
15. With G.711 and a 20-ms packetization period, what will be the bandwidth requirement over
Ethernet (basic Ethernet with no 802.1Q or any tunneling)?
a. 87.2 kbps
b. 80 kbps
c. 64 Kbps
d. 128 Kbps
1763fm.book Page 9 Monday, April 23, 2007 8:58 AM

“Do I Know This Already?” Quiz 9

16. With G.729 and 20 ms packetization period, what will be the bandwidth requirement over
PPP if cRTP is used with no checksum?
a. 8 Kbps
b. 26.4 Kbps
c. 11.2 Kbps
d. 12 Kbps
17. Which of the following is not a factor in determining the amount of bandwidth that can be
saved with VAD?
a. Type of audio (one-way or two-way)
b. Codec used
c. Level of background noise
d. Language and character of the speaker
18. Which of the following is not a voice gateway function on a Cisco router (ISR)?
a. Connect traditional telephony devices
b. Survivable Remote Site Telephony (SRST)
c. CallManager Express
d. Complete phone feature administration
19. Which of the following is not a Cisco Unified CallManager function?
a. Converting analog signal to digital format
b. Dial plan administration
c. Signaling and device control
d. Phone feature administration
20. Which of the following is not an enterprise IP Telephony deployment model?
a. Single site
b. Single site with clustering over WAN
c. Multisite with either centralized or distributed call processing
d. Clustering over WAN
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10 Chapter 1: Cisco VoIP Implementations

Foundation Topics

Introduction to VoIP Networks
Upon completion of this section, you will know the primary advantages and benefits of packet
telephony networks, the main components of packet telephony networks, the definition of analog
and digital interfaces, and the stages of a phone call. The final part of this section helps you
understand the meaning of distributed and centralized call control and the differences between
these two types of call control.

Benefits of Packet Telephony Networks
Many believe that the biggest benefit of packet telephony is toll bypass, or simply long-distance
cost savings. However, because the cost of a long-distance call to most parts of the world has
decreased substantially, this is not even one of the top three reasons for migrating to packet
telephony networks in the North American market.

The main benefits of packet telephony networks are as follows:

■ More efficient use of bandwidth and equipment, and lower transmission costs—Packet
telephony networks do not use a dedicated 64-kbps channel (DS0) for each VoIP phone call.
VoIP calls share the network bandwidth with other applications, and each voice call can use
less bandwidth than 64 kbps. Packet telephony networks do not use expensive circuit-
switching equipment such as T1 multiplexers, which helps to reduce equipment and operation
costs.

■ Consolidated network expenses—In a converged network, the data applications, voice,
video, and conferencing applications do not have separate and distinct hardware, software,
and supporting personnel. They all operate over a common infrastructure and use a single
group of employees for configuration and support. This introduces a significant cost saving.

■ Improved employee productivity—Cisco IP phones are more than just simple phones. With
IP phones, you can access user directories. Furthermore, you can access databases through
extensible markup language (XML). Therefore, you can utilize the Cisco IP phone as a
sophisticated communication device that allows users to run applications from their IP
phones. In short, Cisco IP Phones enhance the user experience by bringing informational
resources to the end user.

■ Access to new communications devices—Unlike the traditional analog and PBX phones, IP
phones can communicate with a number of devices such as computers (computer telephony
applications), networking devices, personal digital assistants, and so on, through IP
connectivity.
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Introduction to VoIP Networks 11

Despite the stated benefits of packet telephony networks, when an organization decides to migrate
to packet telephony, it will have to make an initial investment, which will probably not have an
attractive short-term return on investment (ROI). Also, if the existing telephony equipment is not
fully depreciated, there will be more reluctance to migrating to packet telephony at this time.
Finally, it is not easy to consolidate and train the different groups of personnel who used to
separately support the data and telephone equipment and networks.

Packet Telephony Components
A packet telephony network must perform several mandatory functions, and it can perform many
optional ones. This requires existence and proper operation of various components. Some devices
can perform multiple functions simultaneously; for example, for a small deployment a gateway
can also act as a gatekeeper. The following is a list of the major components of a packet telephony
network, but not all of the components are always present and utilized:

■ Phones—There might be analog phones, PBX phones, IP phones, Cisco IP Communicator,
and so on. Please note that non-IP phones require the existence of IP gateway(s).

■ Gateways—Gateways interconnect and allow communication among devices that are not all
necessarily accessible from within the IP network. For instance, a call from inside an IP
network to a friend or relative’s residential analog phone line must go through at least one
gateway. If a call from an analog phone, on a router’s FXS port for example, must go through
a Wide Area Network (WAN) connection such as a Frame-Relay virtual circuit to get to a
remote office, it will also have to go through a gateway. Connectivity of IP networks to Private
Branch Exchange (PBX) systems is also accomplished through gateways.

■ Multipoint control units (MCU)—An MCU is a conference hardware component. MCU is
comprised of a Multipoint Controller and an optional Multipoint Processor that combines the
received streams from conference participants and returns the result to all the conference
participants.

■ Application and database servers—These servers are available for each of the required and
optional applications within the IP/packet telephony network. For instance, TFTP servers
save and serve IP phone operating systems and configuration files, and certain application
servers provide XML-based services to IP phones.

■ Gatekeepers—You can obtain two distinct and independent services from gatekeepers:
1. Call routing, which is essentially resolving a name or phone number to an IP address, and
2. CAC, which grants permission for a call setup attempt.

■ Call agents—In a centralized call control model, call routing, address translation, call setup,
and so on are handled by call agents (CA) rather than the end devices or gateways. For
example, Media Gateway Control Protocol (MGCP) is a centralized model that requires the
existence of CAs. Outside the context of MGCP, the Call Agents are often referred to as
Common Components.
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12 Chapter 1: Cisco VoIP Implementations

■ Video end points—To make video calls or conferences, you must have video end points.
Naturally, for video conferencing, the MCU must also have video capabilities.

■ DSP—Devices that convert analog signals to digital signals and vice versa use DSPs.
Through utilization of different coding and decoding (codec) algorithms such as G.729, DSPs
also allow you to compress voice signals and perhaps perform transcoding (converting one
type of signal to another, such as G.711 to G.729). IP Phones, Gateways, and conference
equipment such as MCUs use DSPs.

At this point, it is important to clarify the difference between two concepts: digital signal and VoIP.
Today, in almost all cases, one of the early tasks performed in voice communication is digitizing
analog voice. This is true regardless of whether the call stays within the PBX system, goes through
the PSTN, or traverses through an IP network. Figure 1-1 shows a company that has two branches.
The local (main) branch has IP phones, but the remote branch has only PBX phones. Even though
all voice calls need digitization, calls that remain within the remote branch are not VoIP calls and
need not be encapsulated in IP packets.

Figure 1-1 Packet Telephony Components
Local Branch

IP Phones Application Servers

IP

Call Agent LAN Switch Gateway

PSTN
V

Remote Branch
MCU Gatekeeper

PBX
V V
Video Conference Gateway
Equipment

IP Backbone PBX Phones
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Introduction to VoIP Networks 13

VoIP, on the other hand, in addition to digitizing voice, requires IP-based signaling (for call
routing, admission control, setup, maintenance, status, teardown, and so on). Also, VoIP requires
conversion of analog voice into IP packets and transport using IP-based protocols such as Real-
time Transport Protocol (RTP). Many organizations might not be using VoIP (packet telephony)
but have been enjoying the benefits of voice digitization technologies such as PBX and T1 lines.
Converting analog voice signals to digital voice signals and back is almost always done. But VoIP
signaling and VoIP encapsulation and transport happen only in packet telephony networks. In
Figure 1-1, all phone calls made with the IP phones from the main local branch are IP dependent
and need IP signaling, IP encapsulation, and transportation in addition to the initial digitization.

You might ask if a packet telephony network always includes and needs a gateway. The answer is
this: If the IP phones need to make calls and receive them from PBX phones or the phones on the
PSTN network, or if certain calls have to leave the LAN and go through a WAN to reach non-IP
phones (such as analog or PBX phones) at remote locations, a gateway is definitely necessary. In
Figure 1-1, a phone call made from an IP phone in the local branch to another IP phone within the
local branch does not require the services of a voice gateway.

Analog Interfaces
A gateway can have many types of analog interfaces: FXS (Foreign Exchange Station), FXO
(Foreign Exchange Office), and E&M (Earth and Magneto or Ear and Mouth).

An FX connection has a station and an office end. The office end (FXO) provides services such as
battery, dial tone, digit collection, and ringing to the other end, namely the station (FXS).

The FXS interface of a gateway is meant for analog phones, fax machines, and modems. To those
devices, the gateway acts like the PSTN central office (CO) switch.

The FXO interface of a gateway can connect to a regular phone jack to be connected to the PSTN
CO switch. The FXO interface acts as a regular analog device such as a legacy analog phone, and
it expects to receive battery, dial tone, digit collection, ringing, and other services from the other
side, namely the PSTN CO switch. In many small branch offices, at least one FXO interface on a
gateway is dedicated to and connected to the PSTN for emergency 911 call purposes.

The E&M connections traditionally provided PBX-to-PBX analog trunk connectivity. However,
any two of gateways, PBX switches, or PSTN CO switches may be connected using an E&M
connection with E&M interfaces present. Five different types of E&M types exist based on the
circuitry, battery present, wiring, and signaling used.

Figure 1-2 shows a gateway with a fax machine plugged into its FXS interface. Its FXO interface
is connected to the PSTN CO switch, and its E&M interface is connected to a PBX switch. The
gateway has connectivity to the IP phones through the LAN switch, and it provides connectivity
to the other branches through the IP backbone (WAN).
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14 Chapter 1: Cisco VoIP Implementations

Figure 1-2 Gateway Analog Interfaces
Local Branch

Phone/Fax
IP Phones Application Servers Machine

IP

FXS

Call Agent LAN Switch Gateway
LAN
FXO
Interface
T1 CO PSTN
V Switch
E&M

PBX
MCU

WAN
Interface
Video Conference
Equipment PBX Phones

Remote
Branches
IP Backbone

Digital Interfaces
Gateways can also connect to telco and PBX switches using digital interfaces. A gateway can have
BRI or T1/E1 digital interfaces. Using a T1 connection is common in North America, whereas E1
lines are more common in Europe. You can configure the T1/E1 interface controller as an ISDN
PRI or as Channelized T1/E1 and use channel associated signaling (CAS).

BRI and PRI interfaces use common channel signaling (CCS), where a D (Delta) channel is
dedicated to a messaging style of signaling, such as Q931 (or QSIG). You can configure a T1
controller to perform channel associated signaling (CAS) instead. T1 CAS does not dedicate a D
channel to signaling. Each T1 CAS channel gives up a few data bits to perform signaling;
therefore, T1 CAS is also referred to as robbed bit signaling. You can also configure an E1
interface to perform CAS, but because E1 CAS still dedicates a channel to signaling, data channels
do not lose bits to signaling.
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Introduction to VoIP Networks 15

Table 1-2 lists and compares the BRI, PRI, and CT1/CE1 digital interfaces.

Table 1-2 Summary of Digital Interfaces

64 Kbps Data/ Framing
Interface Voice Channels Signaling Overhead Total Bandwidth
BRI 2 16 kbps 48 kbps 192 kbps

(D channel)
T1 CAS 24 In-band (robbed bits) 8 kbps 1544 kbps
T1 CCS 23 64 kbps 8 kbps 1544 kbps

(D Channel)
E1 CAS 30 64 kbps 64 kbps 2048 kbps
E1 CCS 30 64 kbps 64 kbps 2048 kbps

(D Channel)

Stages of a Phone Call
The three most popular VoIP signaling and control protocols are H.323, which is an ITU standard;
Media Gateway Control Protocol (MGCP), which is an Internet Engineering Task Force (IETF)
standard; and Session Initiation Protocol (SIP), also an IETF standard. Regardless of the signaling
protocol used, a phone call has three main stages: call setup, call maintenance, and call teardown.

During call setup, the destination telephone number must be resolved to an IP address, where the
call request message must be sent; this is called call routing. Call admission control (CAC) is an
optional step that determines whether the network has sufficient bandwidth for the call. If bandwidth
is inadequate, CAC sends a message to the initiator indicating that the call cannot get through
because of insufficient resources. (The caller usually hears a fast busy tone.)

If call routing and CAC succeed, a call request message is sent toward the destination. If the
destination is not busy and it accepts the call, some parameters for the call must be negotiated
before voice communication begins. Following are a few of the important parameters that must be
negotiated:

■ The IP addresses to be used as the destination and source of the VoIP packets between the call
end points

■ The destination and source User Datagram Protocol (UDP) port numbers that the RTP uses at
each call end point

■ The compression algorithm (codec) to be used for the call; for example, whether G.729,
G.711, or another standard will be used
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16 Chapter 1: Cisco VoIP Implementations

Call maintenance collects statistics such as packets exchanged, packets lost, end-to-end delay, and
jitter during the VoIP call. The end points (devices such as IP phones) that collect this information
can locally analyze this data and display the call quality information upon request, or they can
submit the results to another device for centralized data analysis. Call teardown, which is usually
due to either end point terminating the call, or to put it simply, hanging up, sends appropriate
notification to the other end point and any control devices so that the resources can be made free
for other calls and purposes.

Distributed Versus Centralized Call Control
Two major call control models exist: distributed call control and centralized call control. The
H.323 and SIP protocols are classified as distributed, whereas the MGCP protocol is considered
as a centralized call control VoIP signaling protocol.

In the distributed model, multiple devices are involved in setup, maintenance, teardown, and other
aspects of call control. The voice-capable devices that perform these tasks have the intelligence
and proper configuration to do so.

Figure 1-3 shows a simple case in which two analog phones are plugged into the FXS interfaces
of two Cisco voice gateways that have connectivity over an IP network and use the H.323 signaling
protocol (distributed model). From the time that the calling device goes off-hook to the time that
the called device receives the ring, seven steps are illustrated within this distributed call control
model:

1. The calling phone goes off-hook, and its voice gateway (R1) provides a dial tone and waits
for digits.
2. The calling phone sends digits, and its voice gateway (R1) collects them.
3. The voice gateway (R1) determines whether it can route the call, or whether it has an IP
destination configured for the collected digits. In this case, the voice gateway (R1) determines
the other voice gateway (R2) as the destination. This is called call routing; the R1 is capable
of doing that in the distributed model.
4. R1 sends a call setup message to R2 along with information such as the dialed number.
5. R2 receives the call setup message from R1 along with the information sent.
6. R2 determines whether it has a destination mapped to the called number. In this case, the
called number maps to a local FXS interface. R2 takes care of this call routing in the
distributed model.
7. If the determined FXS port on R2 is not busy and it is not configured to reject this call, R2
sends an AC ringing voltage to the FXS port, and the phone plugged into that interface rings.
If the ringing phone on the FXS of R2 goes off-hook, the call is considered answered, and
voice traffic starts flowing between the calling and called parties.
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Introduction to VoIP Networks 17

Figure 1-3 Call Setup Example for Distributed Call Control
1. Phone 1 goes off-hook and
receives dial tone from R1.
2. Digits 7. Ringing

R1 IP Network R2
Phone 1 Phone 2
4. Call Setup Message

V V
3. Call Routing 5. R2 Receives Call Setup
6. Call Routing

While the call is in progress, endpoints can monitor the quality of the call based on the number
of packets sent, received, and dropped, and the amount of delay and jitter experienced. In the
distributed model, the end points might have the intelligence and configuration to terminate a call
if its quality is not acceptable.

If either phone on R1 or R2 hangs up (goes on-hook), the corresponding router sends a call
termination message to its counterpart. Both routers release all resources that are dedicated to
the call. Notice that in this distributed model example, the end-point gateways handled the call
teardown in addition to the other tasks.

In the example used here, no call routing, call setup, call maintenance, or call teardown tasks
depended on a centralized intelligent agent. The gateways at both ends had the intelligence and
configuration to handle all the tasks involved in the end-to-end call. You must note, though, that if
there were thousands of end devices, each would need the intelligence and configuration to be able
to make and maintain calls to all other destinations (not necessarily at the same time). Naturally,
a fully distributed model is not scalable; imagine if the telephone in your home needed the
intelligence and configuration to be able to call every other phone number in the world, without
the services of telco switches!

For large-scale deployments of H.323 or SIP, which are distributed call control protocols, special
devices are added to offer a scalable and manageable solution. For example, the H.323 gatekeeper
can be utilized to assist H.323 terminals or gateways with call routing. In SIP environments,
special SIP servers such as Registrar, Location, Proxy, and Redirect can be utilized to facilitate
scalability and manageability, among other benefits.

Centralized call control relieves the gateways and end points from being responsible for tasks such
as call routing, call setup, CAC, and call teardown. MGCP end points do not have the intelligence
and configuration to perform those tasks, and they are expected to receive those services from
CAs. Analog voice digitization, encapsulation of digitized voice in IP packets, and transporting
(sending) the IP packets from one end to the other remain the responsibility of the DSPs of the
MGCP gateways and end points. Therefore, when the call is set up, VoIP packet flow does not
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18 Chapter 1: Cisco VoIP Implementations

involve the CA. When either end point terminates the call, the CA is notified, and the CA in turn
notifies both parties to release resources and essentially wait until the next call is initiated.

Figure 1-4 shows a simple case in which two analog phones are plugged into the FXS interfaces
of two Cisco voice gateways that have connectivity over an IP network and are configured to use
the MGCP signaling protocol (centralized model), using the services of a CA. The sequence of
events from the time that the calling phone goes off-hook to the time that the called phone rings is
listed here:

1. The phone plugged into the FXS port of R1 goes off-hook. R1 detects this event (service
request) and notifies the CA.
2. The CA instructs R1 to provide a dial tone on that FXS port, collect digits one at a time, and
send them to the CA.
3. R1 provides a dial tone, collects dialed digits, and sends them to the CA one at a time.
4. The CA, using its call routing table and other information, determines that the call is for an
FXS port on R2. It is assumed that R2 is also under the control of this CA, and that is why the
CA had such detailed information about the R2 port and associated numbers. The CA must
also determine if that FXS interface is free and whether the call is allowed. Note that the call
routing capability of the CA not only determines that R2 is the destination end device, but it
also informs which interface on R2 the call is for. In other words, neither R1 nor R2 have to
know how to perform call routing tasks.
5. Upon successful call routing, availability, and restrictions checks, the CA notifies R2 of the
incoming call for its FXS interface. R2 then sends an AC ringing voltage to the appropriate
FXS port.

Figure 1-4 Call Setup Example for Centralized Call Control
4. Call Routing
Call Agent

k
oo
ff-H tion
O a s
1. otific on
N cti
s tru 5.
Digits In C Ringing
2. s
git M all S
Di es
ed sa etu
Phone 1 R1 ec
t ge p R2 Phone 2
ll
Co
3.
V VoIP Packets (Active Call Traffic) V
Call Routing Call Routing
IP Network
1763fm.book Page 19 Monday, April 23, 2007 8:58 AM

Digitizing and Packetizing Voice 19

While the call is in progress, the end points (R1 and R2 in this example) collect and analyze the
call statistics, such as packets sent and lost, and delay and jitter incurred (Theoretically, if the
quality of the call is unacceptable, the CA is notified, and the CA instructs both parties to terminate
the call.) If either phone hangs up, the gateway it is connected to (R1 or R2) notifies the CA of this
event. The CA instructs both parties that call termination procedures must be performed and call
resources must be released.

In the centralized call control model, the end points are not responsible for call control functions;
therefore, they are simpler devices to build, configure, and maintain. On the other hand, the CA is
a critical component within the centralized model and, to avoid a single point of failure, it requires
deployment of fault-tolerance technologies. It is easier to manage a centralized model than to
manage the distributed model, because only the CAs need to be configured and maintained.
Implementing new services, features, and policies is also easier in the centralized model.

Digitizing and Packetizing Voice
Upon completion of this section, you will be able to identify the steps involved in converting an
analog voice signal to a digital voice signal, explain the Nyquist theorem, the reason for taking
8000 voice samples per second; and explain the method for quantization of voice samples.
Furthermore, you will be familiar with standard voice compression algorithms, their bandwidth
requirements, and the quality of the results they yield. Knowing the purpose of DSP in voice
gateways is the last objective of this section.

Basic Voice Encoding: Converting Analog to Digital
Converting analog voice signal to digital format and transmitting it over digital facilities (such as
T1/E1) had been created and put into use before Bell (a North American telco) invented VoIP
technology in 1950s. If you use digital PBX phones in your office, you must realize that one of the
first actions that these phones perform is converting the analog voice signal to a digital format.
When you use your regular analog phone at home, the phone sends analog voice signal to the telco
CO. The Telco CO converts the analog voice signal to digital format and transmits it over the
public switched telephone network (PSTN). If you connect an analog phone to the FXS interface
of a router, the phone sends an analog voice signal to the router, and the router converts the analog
signal to a digital format. Voice interface cards (VIC) require DSPs, which convert analog voice
signals to digital signals, and vice versa.

Analog-to-digital conversion involves four major steps:

1. Sampling
2. Quantization
3. Encoding
4. Compression (optional)
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20 Chapter 1: Cisco VoIP Implementations

Sampling is the process of periodic capturing and recording of voice. The result of sampling is
called a pulse amplitude modulation (PAM) signal. Quantization is the process of assigning
numeric values to the amplitude (height or voltage) of each of the samples on the PAM signal
using a scaling methodology. Encoding is the process of representing the quantization result for
each PAM sample in binary format. For example, each sample can be expressed using an 8-bit
binary number, which can have 256 possible values.

One common method of converting analog voice signal to digital voice signal is pulse code
modulation (PCM), which is based on taking 8000 samples per second and encoding each sample
with an 8-bit binary number. PCM, therefore, generates 64,000 bits per second (64 Kbps); it does
not perform compression. Each basic digital channel that is dedicated to transmitting a voice call
within PSTN (DS0) has a 64-kbps capacity, which is ideal for transmitting a PCM signal.

Compression, the last step in converting an analog voice signal to digital, is optional. The purpose
of compression is to reduce the number of bits (digitized voice) that must be transmitted per
second with the least possible amount of voice-quality degradation. Depending on the
compression standard used, the number of bits per second that is produced after the compression
algorithm is applied varies, but it is definitely less than 64 Kbps.

Basic Voice Encoding: Converting Digital to Analog
When a switch or router that has an analog device such as a telephone, fax, or modem connected
to it receives a digital voice signal, it must convert the analog signal to digital or VoIP before
transmitting it to the other device. Figure 1-5 shows that router R1 receives an analog signal and
converts it to digital, encapsulates the digital voice signal in IP packets, and sends the packets to
router R2. On R2, the digital voice signal must be de-encapsulated from the received packets.
Next, the switch or router must convert the digital voice signal back to analog voice signal and
send it out of the FXS port where the phone is connected.

Figure 1-5 Converting Analog Signal to Digital and Digital Signal to Analog
Analog Signal Digital Signal Digital Signal Analog Signal

Phone 1 R1 Encapsulation De-Encapsulation R2 Phone 2
IP Packet IP Packet
FXS FXS FXS
V V
1. Sampling IP Network 1. Decompression
2. Quantization 2. Decoding
3. Encoding 3. Filtering and
4. Compression Reconstructing the
Analog Signal
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Digitizing and Packetizing Voice 21

Converting digital signal back to analog signal involves the following steps:

1. Decompression (optional)
2. Decoding and filtering
3. Reconstructing the analog signal
If the digitally transmitted voice signal was compressed at the source, at the receiving end, the
signal must first be decompressed. After decompression, the received binary expressions are
decoded back to numbers, which regenerate the PAM signal. Finally, a filtering mechanism
attempts to remove some of the noise that the digitization and compression might have introduced
and regenerates an analog signal from the PAM signal. The regenerated analog signal is hopefully
very similar to the analog signal that the speaker at the sending end had produced. Do not forget
that DPS perform digital-to-analog conversion, similar to analog to digital conversion.

The Nyquist Theorem
The number of samples taken per second during the sampling stage, also called the sampling rate,
has a significant impact on the quality of digitized signal. The higher the sampling rate is, the
better quality it yields; however, a higher sampling rate also generates higher bits per second that
must be transmitted. Based on the Nyquist theorem, a signal that is sampled at a rate at least twice
the highest frequency of that signal yields enough samples for accurate reconstruction of the signal
at the receiving end.

Figure 1-6 shows the same analog signal on the left side (top and bottom) but with two sampling
rates applied: the bottom sampling rate is twice as much as the top sampling rate. On the right side
of Figure 1-6, the samples received must be used to reconstruct the original analog signal. As you
can see, with twice as many samples received on the bottom-right side as those received on the
top-right side, a more accurate reconstruction of the original analog signal is possible.

Human speech has a frequency range of 200 to 9000 Hz. Hz stands for Hertz, which specifies the
number of cycles per second in a waveform signal. The human ear can sense sounds within a
frequency range of 20 to 20,000 Hz. Telephone lines were designed to transmit analog signals
within the frequency range of 300 to 3400 Hz. The top and bottom frequency levels produced by
a human speaker cannot be transmitted over a phone line. However, the frequencies that are
transmitted allow the human on the receiving end to recognize the speaker and sense his/her tone
of voice and inflection. Nyquist proposed that the sampling rate must be twice as much as the
highest frequency of the signal to be digitized. At 4000 Hz, which is higher than 3400 Hz (the
maximum frequency that a phone line was designed to transmit), based on the Nyquist theorem,
the required sampling rate is 8000 samples per second.
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22 Chapter 1: Cisco VoIP Implementations

Figure 1-6 Effect of Higher Sampling Rate

Quantization
Quantization is the process of assigning numeric values to the amplitude (height or voltage) of
each of the samples on the PAM signal using a scaling methodology. A common scaling method
is made of eight major divisions called segments on each polarity (positive and negative) side.
Each segment is subdivided into 16 steps. As a result, 256 discrete steps (2 × 8 × 16) are possible.

The 256 steps in the quantization scale are encoded using 8-bit binary numbers. From the 8 bits,
1 bit represents polarity (+ or –), 3 represent segment number (1 through 8), and 4 bits represent
the step number within the segment (1 through 16). At a sampling rate of 8000 samples per second,
if each sample is represented using an 8-bit binary number, 64,000 bits per second are generated
for an analog voice signal. It must now be clear to you why traditional circuit-switched telephone
networks dedicated 64 Kbps channels, also called DS0s (Digital Signal Level 0), to each telephone
call.

Because the samples from PAM do not always match one of the discrete values defined by
quantization scaling, the process of sampling and quantization involves some rounding. This
rounding creates a difference between the original signal and the signal that will ultimately be
reproduced at the receiver end; this difference is called quantization error. Quantization error or
quantization noise, is one of the sources of noise or distortion imposed on digitally transmitted
voice signals.
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Digitizing and Packetizing Voice 23

Figure 1-7 shows two scaling models for quantization. If you look at the graph on the top, you will
notice that the spaces between the segments of that graph are equal. However, the spaces between
the segments on the bottom graph are not equal: the segments closer to the x-axis are closer to each
other than the segments that are further away from the x-axis. Linear quantization uses graphs with
segments evenly spread, whereas logarithmic quantization uses graphs that have unevenly spread
segments. Logarithmic quantization yields smaller signal-to-noise quantization ratio (SQR),
because it encounters less rounding (quantization) error on the samples (frequencies) that human
ears are more sensitive to (very high and very low frequencies).

Figure 1-7 Linear Quantization and Logarithmic Quantization
Y-axis

X-axis Equidistant Segments

Linear Quantization

Y-axis
Logarithmic Quantization

Segments are
NOT X-axis
Equidistant

Two variations of logarithmic quantization exist: A-Law and µ-Law. Bell developed µ-Law
(pronounced me-you-law) and it is the method that is most common in North America and Japan.
ITU modified µ-Law and introduced A-Law, which is common in countries outside North America
(except Japan). When signals have to be exchanged between a µ-Law country and an A-Law
country in the PSTN, the µ-Law country must change its signaling to accommodate the A-Law
country.
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24 Chapter 1: Cisco VoIP Implementations

Compression Bandwidth Requirements and Their Comparative Qualities
Several ITU compression standards exist. Voice compression standards (algorithms) differ based
on the following factors:

■ Bandwidth requirement

■ Quality degradation they cause

■ Delay they introduce

■ CPU overhead due to their complexity

Several techniques have been invented for measuring the quality of the voice signal that has been
processed by different compression algorithms (codecs). One of the standard techniques for
measuring quality of voice codecs, which is also an ITU standard, is called mean opinion score
(MOS). MOS values, which are subjective and expressed by humans, range from 1 (worst) to 5
(perfect or equivalent to direct conversation). Table 1-3 displays some of the ITU standard codecs
and their corresponding bandwidth requirements and MOS values.

Table 1-3 Codec Bandwidth Requirements and MOS Values

Codec Associated Bit Rate Quality Based on
Standard Acronym Codec Name (BW) MOS
G.711 PCM Pulse Code Modulation 64 Kbps 4.10
G.726 ADPCM Adaptive Differential PCM 32, 24, 3.85 (for 32 Kbps)
16 Kbps
G.728 LDCELP Low Delay Code Exited 16 Kbps 3.61
Linear Prediction
G.729 CS-ACELP Conjugate Structure 8 Kbps 3.92
Algebraic CELP
G.729A CS-ACELP Conjugate Structure 8 Kbps 3.90
Annex a Algebraic CELP Annex A

MOS is an ITU standard method of measuring voice quality based on the judgment of several
participants; therefore, it is a subjective method. Table 1-4 displays each of the MOS ratings along
with its corresponding interpretation, and a description for its distortion level. It is noteworthy that
an MOS of 4.0 is deemed to be Toll Quality.
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Digitizing and Packetizing Voice 25

Table 1-4 Mean Opinion Score

Rating Speech Quality Level of Distortion
5 Excellent Imperceptible
4 Good Just perceptible but not annoying
3 Fair Perceptible but slightly annoying
2 Poor Annoying but not objectionable
1 Unsatisfactory Very annoying and objectionable

Perceptual speech quality measurement (PSQM), ITU’s P.861 standard, is another voice quality
measurement technique implemented in test equipment systems offered by many vendors. PSQM
is based on comparing the original input voice signal at the sending end to the transmitted voice
signal at the receiving end and rating the quality of the codec using a 0 through 6.5 scale, where 0
is the best and 6.5 is the worst.

Perceptual analysis measurement system (PAMS) was developed in the late 1990s by British
Telecom. PAMS is a predictive voice quality measurement system. In other words, it can predict
subjective speech quality measurement methods such as MOS.

Perceptual evaluation of speech quality (PESQ), the ITU P.862 standard, is based on work done
by KPN Research in the Netherlands and British Telecommunications (developers of PAMS).
PESQ combines PSQM and PAMS. It is an objective measuring system that predicts the results of
subjective measurement systems such as MOS. Various vendors offer PESQ-based test equipment.

Digital Signal Processors
Voice-enabled devices such as voice gateways have special processors called DSPs. DSPs are
usually on packet voice DSP modules (PVDM). Certain voice-enabled devices such as voice
network modules (VNM) have special slots for plugging PVDMs into them. Figure 1-8 shows a
network module high density voice (NM-HDV) that has five slots for PVDMs. The NM in Figure
1-8 has four PVDMs plugged into it . Different types of PVDMs have different numbers of DSPs,
and each DSP handles a certain number of voice terminations. For example, one type of DSP can
handle tasks such as codec and transcoding for up to 16 voice channels if a low-complexity codec
is used, or up to 8 voice channels if a high-complexity codec is used.
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26 Chapter 1: Cisco VoIP Implementations

Figure 1-8 Network Module with PVDMs
PVDM2 Slots
(Two on Each Side, Total of
Four)

Onboard T1/E1– Ports

DSPs provide three major services:

■ Voice termination

■ Transcoding

■ Conferencing

Calls to or from voice interfaces of a voice gateway are terminated by DSPs. DSP performs
analog-to-digital and digital-to-analog signal conversion. It also performs compression (codec),
echo cancellation, voice activity detection (VAD), comfort noise generation (CNG), jitter handling,
and some other functions.

When the two parties in an audio call use different codecs, a DSP resource is needed to perform
codec conversion; this is called transcoding. Figure 1-9 shows a company with a main branch and
a remote branch with an IP connection over WAN. The voice mail system is in the main branch,
and it uses the G.711 codec. However, the branch devices are configured to use G.729 for VoIP
communication with the main branch. In this case, the edge voice router at the main branch needs
to perform transcoding using its DSP resources so that the people in the remote branch can retrieve
their voice mail from the voice mail system at the main branch.

DSPs can act as a conference bridge: they can receive voice (audio) streams from the participants
of a conference, mix the streams, and send the mix back to the conference participants. If all the
conference participants use the same codec, it is called a single-mode conference, and the DSP
does not have to perform codec translation (called transcoding). If conference participants use
different codecs, the conference is called a mixed-mode conference, and the DSP must perform
transcoding. Because mixed-mode conferences are more complex, the number of simultaneous
mixed-mode conferences that a DSP can handle is less than the number of simultaneous single-
mode conferences it can support.
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Encapsulating Voice Packets 27

Figure 1-9 DSP Transcoding Example
Main Branch Remote Branch

IP

IP WAN
(G.729 Only)
G.729

G.711 DSP
Voice Mail Server Transcoding
(G.711 Only)

Encapsulating Voice Packets
This section explains the protocols and processes involved in delivering VoIP packets as opposed
to delivering digitized voice over circuit-switched networks. It also explains the RTP as the
transport protocol of choice for voice and discusses the benefits of RTP header compression
(cRTP).

End-to-End Delivery of Voice
To review the traditional model of voice communication over the PSTN, imagine a residential
phone that connects to the telco CO switch using an analog telephone line. After the phone goes
off-hook and digits are dialed and sent to the CO switch, the CO switch, using a special signaling
protocol, finds and sends call setup signaling messages to the CO that connects to the line of the
destination number. The switches within the PSTN are connected using digital trunks such as
T1/E1 or T3/E3. If the call is successful, a single channel (DS0) from each of the trunks on the
path that connects the CO switches of the caller and called number is dedicated to this phone call.
Figure 1-10 shows a path from the calling party CO switch on the left to the called party CO switch
on the right.
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28 Chapter 1: Cisco VoIP Implementations

Figure 1-10 Voice Call over Traditional Circuit-Switched PSTN
Analog Residential
Phone

Analog
PSTN Residential
Line

Analog-to-
Digital
CO Conversion
Vice Versa

Digital
Trunks

Digital
Trunks

Analog-to-
Digital CO
Conversion
Vice Versa

Analog Residential Line

Analog Residential
Phone

After the path between the CO switches at each end is set up, while the call is active, analog voice
signals received from the analog lines must be converted to digital format, such as G.711 PCM,
and transmitted over the DS0 that is dedicated to this call. The digital signal received at each CO
must be converted back to analog before it is transmitted over the residential line. The bit trans-
mission over DS0 is a synchronous transmission with guaranteed bandwidth, low and constant
end-to-end delay, plus no chance for reordering. When the call is complete, all resources and the
DS0 channel that is dedicated to this call are released and are available to another call.

If two analog phones were to make a phone call over an IP network, they would each need to be
plugged into the FXS interface of a voice gateway. Figure 1-11 displays two such gateways (R1
and R2) connected over an IP network, each of which has an analog phone connected to its FXS
interface.
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Encapsulating Voice Packets 29

Figure 1-11 Voice Call over IP Networks
Analog
Phone 1

D

FXS
D Analog
Analog-to-Digital
R1 Conversion
V
IP Network & Vice Versa,
D V
Plus VoIP
Encapsulation and
De-Encapsulation

D V

D V

Analog-to- D V
Digital
Conversion &
Vice Versa,
Plus VoIP
Encapsulation R2 D
and V
De-Encapsulation
V LEGEND:
Data Over IP: D
FXS D
Voice Over IP: V

D

Analog
Phone 2

Assume that phone 1 on R1 goes off-hook and dials a number that R1 maps to R2. R1 will send a
VoIP signaling call setup message to R2. If the call is accepted and it is set up, each of R1 and R2
will have to do the following:

■ Convert the analog signal received from the phone on the FXS interface to digital (using a
codec such as G.711).

■ Encapsulate the digital voice signal into IP packets.

■ Route the IP packets toward the other router.
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30 Chapter 1: Cisco VoIP Implementations

■ De-encapsulate the digital voice from the received IP packets.

■ Convert the digital voice to analog and transmit it out of the FXS interface.

Notice that in this case, in contrast to a call made over the circuit-switched PSTN network, no end-
to-end dedicated path is built for the call. IP packets that encapsulate digitized voice (20 ms of
audio by default) are sent independently over the IP network and might arrive out of order and
experience different amounts of delay. (This is called jitter.) Because voice and data share the IP
network with no link or circuit dedicated to a specific flow or call, the number of data and voice
calls that can be active at each instance varies. Also, it affects the amount of congestion, loss, and
delay in the network.

Protocols Used in Voice Encapsulation
Even though the term VoIP implies that digitized voice is encapsulated in IP packets, other
protocol headers and mechanisms are involved in this process. Although the two major TCP/IP
transport layer protocols, namely TCP and UDP, have their own merits, neither of these protocols
alone is a suitable transport protocol for real-time voice. RTP, which runs over UDP using UDP
ports 16384 through 32767, offers a good transport layer solution for real-time voice and video.
Table 1-5 compares TCP, UDP, and RTP protocols with respect to reliability, sequence numbering
(re-ordering), time-stamping, and multiplexing.

Table 1-5 Comparing Suitability of TCP/IP Transport Protocols for Voice

Feature Required for Voice TCP Offers UDP Offers RTP Offers
Reliability No Yes No No
Sequence numbering Yes Yes No Yes
and reordering
Time-stamping Yes No No Yes
Multiplexing Yes Yes Yes No

TCP provides reliability by putting sequence numbers on the TCP segments sent and expecting
acknowledgements for the TCP segment numbers arriving at the receiver device. If a TCP segment
is not acknowledged before a retransmission timer expires, the TCP segment is resent. This model
is not suitable for real-time applications such as voice, because the resent voice arrives too late for
it to be useful. Therefore, reliability is not a necessary feature for a voice transport protocol. UDP
and RTP do not offer reliable transport. Please note, however, that if the infrastructure capacity,
configuration, and behavior are such that there are too many delayed or lost packets, the quality of
voice and other real-time applications will deteriorate and become unacceptable.

Data segmentation, sequence numbering, reordering, and reassembly of data are services that the
transport protocol must offer, if the application does not or cannot perform those tasks. The
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Encapsulating Voice Packets 31

protocol to transport voice must offer these services. TCP and RTP offer those services, but pure
UDP does not.

Voice or audio signal is released at a certain rate from its source. The receiver of the voice or
audio signal must receive it at the same rate that the source has released it; otherwise, it will sound
different or annoying, or it might even become incomprehensible. Putting timestamps on the
segments encapsulating voice, at source, enables the receiving end to release the voice at the same
rate that it was released at the source. RTP adds timestamps in the segments at source, but TCP
and UDP do not.

Both TCP and UDP allow multiple applications to simultaneously use their services to transport
application data, even if all the active flows and sessions originate and terminate on the same pair
of IP devices. The data from different applications is distinguished based on the TCP or UDP port
number that is assigned to the application while it is active. This capability of the TCP and UDP
protocols is called multiplexing. On the other hand, RTP flows are differentiated based on the
unique UDP port number that is assigned to each of the RTP flows. UDP numbers 16384 through
32767 are reserved for RTP. RTP does not have a multiplexing capability.

Knowing that RTP runs over UDP, considering the fact that neither UDP nor RTP offers the
unneeded reliability and overhead offered by TCP, and that RTP uses sequence numbers and time-
stamping, you can conclude that RTP is the best transport protocol for voice, video, and other real-
time applications. Please note that even though the reliability that TCP offers might not be useful
for voice applications, it is desirable for certain other applications.

RTP runs over UDP; therefore, a VoIP packet has IP (20 bytes), UDP (8 bytes), and RTP (12 bytes)
headers added to the encapsulated voice payload. DSPs usually make a package out of 10-ms
worth of analog voice, and two of those packages are usually transported within one IP packet. (A
total of 20-ms worth of voice in one IP packet is common.) The number of bytes resulting from
20 ms (2 × 10 ms) worth of analog voice directly depends on the codec used. For instance, G.711,
which generates 64 Kbps, produces 160 bytes from 20 ms of analog voice, whereas G.729, which
generates 8 Kbps, produces 20 bytes for 20 ms of analog voice signal. The RTP, UDP, and IP
headers, which total 40 bytes, are added to the voice bytes (160 bytes for G.711 and 20 bytes for
G.729) before the whole group is encapsulated in the Layer 2 frame and transmitted.

Figure 1-12 displays two VoIP packets. One packet is the result of the G.711 codec, and the other
is the result of the G.729 codec. Both have the RTP, UDP, and IP headers. The Layer 2 header is
not considered here. The total number of bytes resulting from IP, UDP, and RTP is 40. Compare
this 40-byte overhead to the size of the G.711 payload (160 bytes) and of the G.729 payload (20
bytes). The ratio of overhead to payload is 40/160, or 25 percent, when G.711 is used; however,
the overhead-to-payload ratio is 40/20, or 200 percent, when G.729 is used!
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32 Chapter 1: Cisco VoIP Implementations

Figure 1-12 Voice Encapsulation Utilizing G.711 and G.729
64000 bps ⫻ 20/1000 sec ⫻ 1 Byte/8 Bits

20 8 12 160
Bytes Bytes Bytes Bytes
IP UDP RTP Digitized Voice

20 ms of Digitized Voice Using G.711

8000 bps ⫻ 20/1000 sec ⫻ 1 Byte/8 Bits

20 8 12 20
Bytes Bytes Bytes Bytes
IP UDP RTP Digitized Voice

20 ms of Digitized Voice Using G.729

If you ignore the Layer 2 overhead for a moment, just based on the overhead imposed by RTP,
UDP, and IP, you can recognize that the required bandwidth is more than the bandwidth that is
needed for the voice payload. For instance, when the G.711 codec is used, the required bandwidth
for voice only is 64 Kbps, but with 25 percent added overhead of IP, UDP, and RTP, the required
bandwidth increases to 80 Kbps. If G.729 is used, the bandwidth required for pure voice is only 8
Kbps, but with the added 200 percent overhead imposed by IP, UDP, and RTP, the required
bandwidth jumps to 24 Kbps. Again, note that the overhead imposed by the Layer 2 protocol and
any other technologies such as tunneling or security has not even been considered.

Reducing Header Overhead
An effective way of reducing the overhead imposed by IP, UDP, and RTP is Compressed RTP
(cRTP). cRTP is also called RTP header compression. Even though its name implies that cRTP
compresses the RTP header only, the cRTP technique actually significantly reduces the overhead
imposed by all IP, UDP, and RTP protocol headers. cRTP must be applied on both sides of a link,
and essentially the sender and receiver agree to a hash (number) that is associated with the 40 bytes
of IP, UDP, and TCP headers. Note that cRTP is applied on a link-by-link basis.

The premise of cRTP is that most of the fields in the IP, UDP, and RTP headers do not change
among the elements (packets) of a common packet flow. After the initial packet with all the
headers is submitted, the following packets that are part of the same packet flow do not carry the
40 bytes of headers. Instead, the packets carry the hash number that is associated with those 40
bytes (sequence number is built in the hash). The main difference among the headers of a packet
flow is the header checksum (UDP checksum). If cRTP does not use this checksum, the size of the
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Encapsulating Voice Packets 33

overhead is reduced from 40 bytes to only 2 bytes. If the checksum is used, the 40 bytes overhead
is reduced to 4 bytes. If, during transmission of packets, a cRTP sender notices that a packet header
has changed from the normal pattern, the entire header instead of the hash is submitted.

Figure 1-13 displays two packets. The top packet has a 160-byte voice payload because of usage
of the G.711 codec, and a 2-byte cRTP header (without checksum). The cRTP overhead-to-voice
payload ratio in this case is 2/160, or 1.25 percent. Ignoring Layer 2 header overhead, because
G.711 requires 64 Kbps for the voice payload, the bandwidth needed for voice and the cRTP
overhead together would be 64.8 Kbps (without header checksum). The bottom packet has a
20-byte voice payload because of usage of the G.729 codec and a 2-byte cRTP header (without
checksum). The cRTP overhead-to-voice payload ratio in this case is 2/20, or 10 percent. Ignoring
Layer 2 header overhead, because G.729 requires 8 Kbps for the voice payload, the bandwidth
needed for voice and the cRTP overhead together would be 8.8 Kbps (without header checksum).

Figure 1-13 RTP Header Compression (cRTP)
64000 bps ⫻ 20/1000 sec ⫻ 1 Byte/8 Bits

2 Bytes Without Checksum
4 Bytes With Checksum
160
Bytes
cRTP Digitized Voice

20 ms of Digitized Voice Using G.711

8000 bps ⫻ 20/1000 sec ⫻ 1 Byte/8 Bits

2 Bytes Without Checksum
4 Bytes With Checksum
20
Bytes
cRTP Digitized Voice

20 ms of Digitized Voice Using G.729

The benefit of using cRTP with smaller payloads (such as digitized voice) is more noticeable than
it is for large payloads. Notice that with cRTP, the total bandwidth requirement (without Layer 2
overhead considered) dropped from 80 Kbps to 64.8 Kbps for G.711, and it dropped from 24 Kbps
to 8.8 Kbps for G.729. The relative gain is more noticeable for G.729. You must, however, consider
factors before enabling cRTP on a link:

■ cRTP does offer bandwidth saving, but it is only recommended for use on slow links (links
with less than 2 Mbps bandwidth). More accurately, Cisco recommends cRTP on 2 Mbps
links only if the cRTP is performed in hardware. cRTP is only recommended on the main
processor if the link speed is below 768 kbps.
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34 Chapter 1: Cisco VoIP Implementations

■ cRTP has a processing overhead, so make sure the device where you enable cRTP has enough
resources.

■ The cRTP process introduces a delay due to the extra computations and header replacements.

■ You can limit the number of cRTP sessions on a link. By default, Cisco IOS allows up to only
16 concurrent cRTP sessions. If enough resources are available on a device, you can increase
this value.

Bandwidth Calculation
Computing the exact amount of bandwidth needed for each VoIP call is necessary for planning and
provisioning sufficient bandwidth in LANs and WANs. The previous section referenced parts of
this computation, but this section thoroughly covers the subject of VoIP bandwidth calculation.
The impact of packet size, Layer 2 overhead, tunneling, security, and voice activity detection are
considered in this discussion.

Impact of Voice Samples and Packet Size on Bandwidth
DSP coverts analog voice signal to digital voice signal using a particular codec. Based on the
codec used, the DSP generates so many bits per second. The bits that are generated for 10
milliseconds (ms) of analog voice signal form one digital voice sample. The size of the digital
voice sample depends on the codec used. Table 1-6 shows how the digital voice sample size
changes based on the codec used. The number of voice bytes for two digital voice samples using
different codecs is shown in the last column.

Table 1-6 Examples of Voice Payload Size Using Different Codecs

Size of 10 ms Size of Two
Codec: Size of Digital Voice Sample for Digitized Voice Digital Voice
Bandwidth 10 ms of Analog Voice in Bits in Bytes Samples (20 ms)
G.711: 64 Kbps 64,000 bps × 10/1000 sec = 640 bits 80 bytes 2 × 80 = 160 bytes
G.726 r32: 32 Kbps 32,000 bps × 10/1000 sec = 320 bits 40 bytes 2 × 40 = 80 bytes
G.726 r24: 24 Kbps 24,000 bps × 10/1000 sec = 240 bits 30 bytes 2 × 30 = 60 bytes
G.726 r16: 16 Kbps 16,000 bps × 10/1000 sec = 160 bits 20 bytes 2 × 20 = 40 bytes
G.728: 16 Kbps 16,000 bps × 10/1000 sec = 160 bits 20 bytes 2 × 20 = 40 bytes
G.729: 8 Kbps 8000 bps × 10/1000 sec = 80 bits 10 bytes 2 × 10 = 20 bytes
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Bandwidth Calculation 35

The total size of a Layer 2 frame encapsulating a VoIP packet depends on the following factors:

■ Packet rate and packetization size—Packet rate, specified in packets per seconds (pps), is
inversely proportional to packetization size, which is the amount of voice that is digitized and
encapsulated in each IP packet. Packetization size is expressed in bytes and depends on the
codec used and the amount of voice that is digitized. For example, if two 10-ms digitized
voice samples (total of 20 ms voice) are encapsulated in each IP packet, the packet rate will
be 1 over 0.020, or 50 packets per second (pps), and if G.711 is used, the packetization size
will be 160 bytes. (See Table 1-6.)

■ IP overhead—IP overhead refers to the total number of bytes in the RTP, UDP, and IP
headers. With no RTP header compression, the IP overhead is 40 bytes. If cRTP with no
header checksum is applied to a link, the IP overhead drops to 2 bytes, and with header
checksum, the IP header checksum is 4 bytes.

■ Data link overhead—Data link layer overhead is always present, but its size depends on the
type of encapsulation (frame type) and whether link compression applied. For instance, the
data link layer overhead of Ethernet is 18 bytes (it is 22 bytes with 802.1Q).

■ Tunneling overhead—Tunneling overhead is only present if some type of tunneling is used.
Generic routing encapsulation (GRE), Layer 2 Tunneling Protocol (L2TP), IP security
(IPsec), QinQ (802.1Q), and Multiprotocol Label Switching (MPLS) are common tunneling
techniques with their own usage reasons and benefits. Each tunneling approach adds a specific
number of overhead bytes to the frame.

Codecs are of various types. The size of each VoIP packet depends on the codec type used and the
number of voice samples encapsulated in each IP packet. The number of bits per second that each
codec generates is referred to as codec bandwidth. The following is a list of some ITU codec
standards, along with a brief description for each:

■ G.711 is PCM—Based on the 8000 samples per second rate and 8 bits per sample, PCM
generates 64,000 bits per second, or 64 Kbps. No compression is performed.

■ G.726 is adaptive differential pulse code modulation (ADPCM)—Instead of constantly
sending 8 bits per sample, fewer bits per sample, which only describe the change from the
previous sample, are sent. If the number of bits (that describe the change) sent is 4, 3, or 2,
G.726 generates 32 Kbps, 24 Kbps, or 16 Kbps respectively, and it is correspondingly called
G.726 r32, G.726 r24, or G.726 r16.

■ G.722 is wideband speech encoding standard—G.722 divides the input signal into two
subbands and encodes each subband using a modified version of ADPCM. G.722 supports a
bit rate of 64 Kbps, 56 Kbps, or 48 Kbps.
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36 Chapter 1: Cisco VoIP Implementations

■ G.728 is low delay code exited linear prediction (LDCELP)—G.728 uses codes that
describe voice samples generated by human vocal cords, and it utilizes a prediction technique.
Wave shapes of five samples (equivalent of 40 bits in PCM) are expressed with 10-bit codes;
therefore, the G.728 bandwidth drops to 16 Kbps.

■ G.729 is conjugate structure algebraic code exited linear prediction (CS-ACELP)—
G.729 also uses codes from a code book; however, 10 samples (equivalent of 80 PCM bits)
are expressed with 10-bit codes. Therefore, the G.729 is only 8 Kbps.

DSPs produce one digital voice sample for 10 milliseconds (ms) of analog voice signal. It is
common among Cisco voice-enabled devices to put two digital voice samples in one IP packet,
but it is possible to put three or four samples in one IP packet if desired. The packetization period
is the amount of analog voice signal (expressed in milliseconds) that is encapsulated in each IP
packet (in digitized format). The merit of more voice samples in a packet—longer packetization
period, in other words—is reduction in the overhead-to-payload ratio.

The problem, though, with putting too many digital voice samples in one IP packet is that when a
packet is dropped, too much voice is lost. That loss has a more noticeable negative effect on the
quality of the call when packets are dropped. The other drawback of a longer packetization period
(more than two or three digital voice samples in one IP packet) is the extra packetization delay it
introduces. More voice bits means a larger IP packet, and a larger IP packet means a longer
packetization period.

Table 1-7 shows a few examples to demonstrate the combined effect of codec used and packet-
ization period (number of digitized 10-ms voice samples per packet) on the voice encapsulating
IP packet (VoIP) size and on the packet rate. The examples in Table 1-7 do not use compressed
RTP and make no reference to the effects of Layer 2 and tunneling overheads.

Table 1-7 Packet Size and Packet Rate Variation Examples

Codec and Packetization Period Voice Payload Total IP Packet
(Number of Encapsulated Codec (Packetization) IP (VoIP) Rate
Digital Voice Samples) Bandwidth Size Overhead Packet Size (pps)
G.711 with 20-ms packetization 64 Kbps 160 bytes 40 bytes 200 bytes 50 pps
period (two 10-ms samples)
G.711 with 30-ms packetization 64 Kbps 240 bytes 40 bytes 280 bytes 33.33 pps
period (three 10-ms samples)
G.729 with 20 ms packetization 8 Kbps 20 bytes 40 bytes 60 bytes 50 pps
period (two 10-ms samples)
G.729 with 40 ms packetization 8 Kbps 40 bytes 40 bytes 80 bytes 25 pps
period (four 10-ms samples)
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Bandwidth Calculation 37

Data Link Overhead
Transmitting an IP packet over a link requires encapsulation of the IP packet in a frame that is
appropriate for the data link layer protocol provisioned on that link. For instance, if the data link
layer protocol used on a link is PPP, the interface connected to that link must be configured for
PPP encapsulation. In other words, any packet to be transmitted out of that interface must be
encapsulated in a PPP frame. When a router routes a packet, the packet can enter the router via an
interface with a certain encapsulation type such as Ethernet, and it can leave the router through
another interface with a different encapsulation such as PPP. After the Ethernet frame enters the
router via the ingress interface, the IP packet is de-encapsulated. Next, the routing decision directs
the packet to the egress interface. The packet has to be encapsulated in the frame proper for the
egress interface data link protocol before it is transmitted.

Different data link layer protocols have a different number of bytes on the frame header; for VoIP
purposes, these are referred to as data link overhead bytes. Data link overhead bytes for Ethernet,
Frame Relay, Multilink PPP (MLP), and Dot1Q (802.1Q) are 18, 6, 6, and 22 bytes in that order,
to name a few. During calculation of the total bandwidth required for a VoIP call, for each link type
(data link layer protocol or encapsulation), you must consider the appropriate data link layer
overhead.

Security and Tunneling Overhead
IPsec is an IETF protocol suite for secure transmission of IP packets. IPsec can operate in two
modes: Transport mode or Tunnel mode. In Transport mode, encryption is applied only to the
payload of the IP packet, whereas in Tunnel mode, encryption is applied to the whole IP packet,
including the header. When the IP header is encrypted, the intermediate routers can no longer
analyze and route the IP packet. Therefore, in Tunnel mode, the encrypted IP packet must be
encapsulated in another IP packet, whose header is used for routing purposes. The new and extra
header added in Transport mode means 20 extra bytes in overhead. In both Transport mode and
Tunnel mode, either an Authentication Header (AH) or an Encapsulating Security Payload (ESP)
header is added to the IP header. AH provides authentication only, whereas ESP provides
authentication and encryption. As a result, ESP is used more often. AH, ESP, and the extra IP
header of the Tunnel mode are the IPsec overheads to consider during VoIP bandwidth calculation.
IPsec also adds extra delay to the packetization process at the sending and receiving ends.

Other common tunneling methods and protocols are not focused on security. IP packets or data
link layer frames can be tunneled over a variety of protocols; the following is a short list of
common tunneling protocols:

■ GRE—GRE transports Layer 3 (network layer) packets, such as IP packets, or Layer 2 (data
link) frames, over IP.

■ Layer 2 Forwarding (L2F) and L2TP—L2F and L2TP transport PPP frames over IP.
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38 Chapter 1: Cisco VoIP Implementations

■ PPP over Ethernet (PPPoE)—PPPoE transports PPP frames over Ethernet frames.

■ 802.1Q tunneling (QinQ)—An 802.1Q frame with multiple 802.1Q headers is called QinQ.
Layer 2 switching engines forward the QinQ frame based on the VLAN number in the top
802.1Q header. When the top header is removed, forwarding of the frame based on the VLAN
number in the lower 802.1Q header begins.

Whether one of the preceding tunneling protocols, IPsec in Tunnel mode, or any other tunneling
protocol is used, the tunnel header is always present and is referred to as tunneling overhead. If
any tunneling protocol is used, the tunneling overhead must be considered in VoIP bandwidth
calculation. Table 1-8 shows the tunneling overhead—in other words, the tunnel header size—for
a variety of tunneling options.

Table 1-8 IPsec and Main Tunneling Protocols Overheads

Protocol Header Size
IPsec Transport Mode 30 to 37 bytes

With ESP header utilizing DES or 3DES for encryption and MD5 or SHA-1 for
authentication. (DES and 3DES require the payload size to be multiples of 8
bytes; therefore, 0 to 7 bytes padding may be necessary.)
IPsec Transport Mode 38 to 53 bytes

With ESP header utilizing AES for encryption and AES-XCBC for
authentication. (AES requires the payload size to be multiples of 16 bytes;
therefore, 0 to 15 bytes of padding might be necessary.)
IPsec Tunnel Mode 50 to 57 bytes

Extra 20 bytes must be added to the IPsec transport mode header size for the extra or
IP header in Tunnel mode
58 to 73 bytes
L2TP 24 bytes
GRE 24 bytes
MPLS 4 bytes
PPPoE 8 bytes

If a company connects two of its sites over the public Internet using IPsec in Tunnel mode (also
called IPsec VPN), you must be able to calculate the total size of the IP packet encapsulating voice
(VoIP). To do that, you need to know the codec used, the packetization period, and whether
compressed RTP is used. The fictitious company under discussion uses the G.729 codec for site-
to-site IP Telephony and a 20-ms packetization period (two 10-ms equivalent digital voice samples
per packet); it does not utilize cRTP. For IPsec, assume tunnel mode with ESP header utilizing
3DES for encryption and SHA-1 for authentication. The voice payload size with G.729 and 20-ms
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Bandwidth Calculation 39

packetization period will be 20 bytes. IP, UDP, and RTP headers add 40 bytes to the voice payload,
bringing the total to 60 bytes. Because 60 is not a multiple of 8, 4 bytes of padding are added to
bring the total to 64 bytes. Finally, the ESP header of 30 bytes and the extra IP header of 20 bytes
bring the total packet size to 114 byes. The ratio of total IP packet size to the size of the voice
payload is 114 over 20—more than 500 percent! Notice that without IPsec (in Tunnel mode), the
total size of the IP packet (VoIP) would have been 60 bytes.

Calculating the Total Bandwidth for a VoIP Call
Calculating the bandwidth that a VoIP call consumes involves consideration for all the factors
discussed thus far. Some fields and protocols are required, each of which might offer
implementation alternatives. Other protocols and fields are optional. You use the bandwidth
consumed by each VoIP call to calculate the total bandwidth required for the aggregate of
simultaneous VoIP calls over LAN and WAN connections. This information is required for the
following purposes:

■ Designing and planning link capacities

■ Deployment of CAC

■ Deployment of quality of service (QoS)

QoS can be defined as the ability of a network to provide services to different applications as per
their particular requirements. Those services can include guarantees to control end-to-end delay,
packet loss, jitter, and guaranteed bandwidth based on the needs of each application. CAC is used
to control the number of concurrent calls to prevent oversubscription of the resources guaranteed
for VoIP calls.

Computing the bandwidth consumed by a VoIP call involves six major steps:

Step 1 Determine the codec and the packetization period. Different codecs
generate different numbers of bits per second (also called codec bandwidth),
and they generally range from 5.3 Kbps to 64 Kbps. The number of digital
voice samples (each of which is equivalent to 10 ms of analog voice)
encapsulated in each IP packet determines the packetization period. A
packetization period of 20 ms, which is the default in Cisco voice-enabled
devices, means that each VoIP packet will encapsulate two 10-ms digital
voice samples.
Step 2 Determine the link-specific information; this includes discovering whether
cRTP is used and what the data link layer protocol (encapsulation type) is.
You must also find out if any security or tunneling protocols and features
are used on the link.
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40 Chapter 1: Cisco VoIP Implementations

Step 3 Calculate the packetization size or, in other words, calculate the size of
voice payload based on the information gathered in Step 1. Multiplying the
codec bandwidth by the packetization period and dividing the result by 8
results in the size of voice payload in bytes. Please note that the packet-
ization period is usually expressed in milliseconds, so you first must divide
this number by 1000 to convert it to seconds. If G.729 with the codec
bandwidth of 8 Kbps is used and the packetization period is 20 ms, the
voice payload size will equal 20 bytes. 8000 (bps) multiplied by 0.020
(seconds) and divided by 8 (bits per byte) yields 20 bytes.
Step 4 Calculate the total frame size. Add the size of IP, UDP, and RTP headers, or
cRTP header if applied, plus the optional tunneling headers and the data
link layer header determined in Step 2, to the size of voice payload (packet-
ization size) determined in Step 3. The result is the total frame size. If the
voice payload size is 20 bytes, adding 40 bytes for RTP, UDP, and IP, and
adding 6 bytes for PPP will result in a frame size of 66 bytes (without usage
of cRTP and any tunneling or security features).
Step 5 Calculate the packet rate. The packet rate is inversed packetization period
(converted to seconds). For example, if the packetization period is 20 ms,
which is equivalent to 0.020 seconds, the packet rate is equal to 1 divided
by 0.020, resulting in a packet rate of 50 packets per second (pps).
Step 6 Calculate the total bandwidth. The total bandwidth consumed by one VoIP
call is computed by multiplying the total frame size (from step 4) converted
to bits multiplied by the packet rate (from step 5). For instance, if the total
frame size is 66 bytes, which is equivalent to 528 bits, and the packet rate
is 50 pps, multiplying 528 by 50 results in a total bandwidth of 26400 bits
per second, or 26.4 Kbps.
Figure 1-14 shows VoIP framing and two methods for computing the bandwidth required for a
VoIP call. Method 1 displayed in Figure 1-14 is based on the six-step process just discussed.

The second method for calculating voice bandwidth is shown as Method 2 in Figure 1-14. This
method is based on the ratio shown on the bottom of Figure 1-14: The ratio of total bandwidth over
voice payload is equal to the ratio of total frame size over voice payload size. If G.729 is used and
the packetization period is 20 milliseconds, the voice payload size will be 20 bytes. With PPP
encapsulation and no cRTP, security, or tunneling, the total frame size adds up to 66 bytes. The
ratio of total frame size to voice payload size is 66 over 20, which is equal to the ratio of voice
bandwidth over codec bandwidth (8 Kbps for G.729). This 66 multiplied by 8 Kbps and divided
by 20 results in voice bandwidth of 26.4 Kbps.
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Bandwidth Calculation 41

Figure 1-14 Computing the VoIP Bandwidth Requirement
E D C B A
Bytes Bytes Bytes Bytes Bytes
Either Digitized Voice
IP+UDP+RTP The size of this section depends
Possible Possible
Layer 2 Header on the codec type and the
Tunnel Security
Header or amount (msec) of analog voice
Header Header
cRTP that is digitized and encapsulated
Header in each IP packet.

VoIP Bandwidth Calculation Method 1:
A = Amount of digitized voice per packet (Bytes)
= CODEC Bandwidth (bps) x Packetization Period (in Sec) / 8 (bytes)
F = Total Frame Size (bits) = 8 x (E + D + C + B + A)
R = Packet Rate = 1/(Packetization Period in Seconds)
Bandwidth per call (kbps) = F x R divided by 1000

VoIP Bandwidth Calculation Method 2:
A = Amount of digitized voice per packet (bytes)
= CODEC Bandwidth (bps) x Packetization Period (in Sec) / 8 (bytes)
F = Total Frame Size (bytes) = E + D + C + B + A
Bandwidth per call = codec bandwidth multiplied by F divided by A
Total Frame Size Total Bandwidth Requirement
=
Voice Payload Size codec Bandwidth (also called Nominal
Bandwidth Requirement)

After you compute the bandwidth for one voice call, you can base the total bandwidth for VoIP on
the maximum number of concurrent VoIP calls you expect or are willing to allow using CAC. The
bandwidth required by VoIP and other applications (non-VoIP) added together generally should
not exceed 75 percent of any bandwidth link. VoIP signaling also consumes bandwidth, but it takes
much less bandwidth than actual VoIP talk (audio) packets. QoS tools and techniques treat VoIP
signaling and VoIP data (audio) packets differently, so VoIP signaling bandwidth and QoS
considerations need special attention.

Effects of VAD on Bandwidth
VAD is a feature that is available in voice-enabled networks. VAD detects silence (speech pauses)
and one-way audio and does not generate data; as a result, it produces bandwidth savings. This
does not happen in circuit-switched voice networks such as the PSTN, where a channel (usually a
64 Kbps DS0) is dedicated to a call regardless of the amount of activity on that circuit.

It is common for about one-third of a regular voice call to be silence; therefore, the concept of VAD
for bandwidth saving is promising. One instance of a modern-day situation is when a caller is put
on hold and listens to music on hold (MOH); in this situation, audio flows in one direction only,
and it is not necessary to send data from the person on hold to anywhere.
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42 Chapter 1: Cisco VoIP Implementations

The amount of bandwidth savings experienced based on VAD depends on the following factors:

■ Type of audio—During a regular telephone call, only one person speaks at a time (usually!);
therefore, no data needs to be sent from the silent party toward the speaking party. The same
argument applies when a caller is put on hold or when the person gets MOH.

■ Background noise level—If the background noise is too loud, VAD does not detect silence
and offers no savings. In other words, the background noise is transmitted as regular audio.

■ Other factors—Differences in language and culture and the type of communication might
vary the amount of bandwidth savings due to VAD. During a conference, or when one person
is lecturing other(s), the listeners remain silent, and VAD certainly takes advantage of that.

Studies have shown that even though VAD can produce about 35 percent bandwidth savings, its
results depend heavily on the fore-mentioned factors. The 35 percent bandwidth savings is based
on distribution of different call types; this is only realized if at least 24 active voice calls are on a
link. If you expect fewer than 24 calls, the bandwidth savings due of VAD should not be included
in the bandwidth calculations. Most conservative people do not count on the VAD savings; in other
words, even though they use the VAD feature, they do not include the VAD bandwidth savings in
their calculations.

Implementing VoIP Support in an Enterprise Network
This section is intended to give you an overview of telephony deployment models and their
necessary elements and components in an enterprise network. It briefly introduces Cisco Unified
CallManager, and it discusses a few different implementation options for CallManager clusters.
The last part of this section includes a simple configuration for a Cisco voice gateway and
concludes with a brief discussion of CAC.

Enterprise Voice Implementations
The main telephony elements of an enterprise Cisco VoIP implementation are gateway, gatekeeper,
Cisco Unified CallManager, and Cisco IP phones. Cisco IP phones need CallManager, because it
acts as an IP PBX for the Cisco IP phones. The gateways provide connectivity between analog,
digital, and IP-based telephony devices and circuits. Gatekeeper is an H.323 device that provides
call routing or CAC services.

Enterprise voice implementations can vary based on many factors. One of those factors is the
number of sites, and the preferred method of data and voice connectivity (primary and backup)
between the sites. Some sites might not have VoIP implemented; other sites might have VoIP
connectivity but no IP phones or other IP Telephony services. The sites with IP phones and
services might have the control components, such as Cisco Unified CallManager cluster, locally
present, or they might have to communicate with the control devices that reside at another branch
or site. Figure 1-15 displays an enterprise with three branches: Branch A, Branch B, and Branch C.
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Implementing VoIP Support in an Enterprise Network 43

Figure 1-15 VoIP Implementation Within an Enterprise
Branch A

Workstations,
PCs, Laptops
Application
Servers
LAN
Switch
Cisco Unified PSTN
CallManager Cluster
CO
T1/E1
IPIP WAN Router &
IP
Voice Gateway V

Branch C

Branch B PBX
PSTN
MAN
PBX
IP
Phones
WAN

SRST
V V
FXO

FXO

PSTN

At Branch A, IP Telephony services and IP phones have been deployed. Branch A has a Cisco
Unified CallManager cluster, and all employees use IP phones. Branch A is connected to Branch
B using a metropolitan-area network (MAN) connection such as Metro Ethernet; voice calls
between Branch A and Branch B must use this path. The Branch A connection to Branch C is over
a WAN, such as legacy Frame Relay or ATM (a modern connection would be an MPLS VPN
connection); voice calls between Branch A and Branch C must use this path. If WAN or MAN
connections are down, voice calls must be rerouted via PSTN; if there is congestion, using the
automated alternate routing (AAR) feature, voice calls are again rerouted via PSTN. Note that at
Branch A, voice calls to and from people outside the enterprise are naturally through PSTN.

At Branch C, on the other hand, the old PBX system and phones are still in use. A voice gateway
at Branch C provides connectivity between the Branch C PBX system (and phones) to the PSTN
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44 Chapter 1: Cisco VoIP Implementations

and all other branch phones over the WAN connection. Again, the preferred path for voice calls
between Branch C and the other branches is over the WAN connection; however, when the WAN
connection is down or is utilized at full capacity, voice calls are rerouted over the PSTN. All
outside calls to and from Branch C are through the PSTN. The enterprise is planning to deploy IP
phones in Branch C, but they are planning to buy a voice gateway with Cisco CallManager Express
instead of installing a full Cisco Unified CallManager cluster at that branch. Cisco CallManager
Express runs on a Cisco gateway instead of a server, and it is ideal for smaller branches that want
IP Telephony without dependence on another branch over a WAN connection.

Branch B is connected to Branch A over a high-speed MAN. IP phones at Branch B are under
control of the Cisco Unified CallManager cluster at Branch A. Voice calls between Branch B and
Branch A must go over the MAN connection. Voice calls between Branch B and Branch C go over
MAN to get to Branch A and then over the WAN to get to Branch C. Voice calls from Branch C to
Branch B take the reverse path. If the MAN connection goes down, survivable remote site
telephony (SRST) deployed on the Branch B gateway allows Branch B IP phones to call each
other, but calls to anywhere else are limited to one at a time and are sent over PSTN. That is
because the gateway at Branch B has two FXO interfaces, which are connected using two analog
phone lines to the PSTN. One of the analog lines is reserved exclusively for 911 emergency calls;
that leaves only one line for any other out-of-branch call (when MAN is down). When the MAN
connection between Branch B and Branch A is up, all of the Branch B outside calls, except the
911 emergency calls, are sent over the MAN connection to Branch A and then through the Branch
A gateway to PSTN.

Voice Gateway Functions on a Cisco Router
The Cisco family of voice gateways, including integrated services routers (ISR), provide
connectivity between analog interfaces, digital interfaces, and IP Telephony devices. Examples of
analog interfaces are FXS and FXO. Examples of analog devices are analog phones, fax machines,
and modems. T1/E1 and BRI are examples of digital interfaces. A PBX is usually connected to a
gateway using T1/E1 interfaces, even though using an E&M interface is also possible. You can set
up a gateway connection to the PSTN CO switch over a T1/E1 or an E&M connection. You can
configure a gateway T1/E1 for CCS, where one channel is dedicated to signaling such as ISDN
Q.931 or QSIG, and the rest of the channels are available for data or digital voice signals. You can
also configure a gateway T1/E1 as CAS. When configured for CAS, a T1 interface can have all 24
channels available for data/digital voice, but each channel loses a few bits to signaling; for this
reason, CAS is also referred to as robbed bit signaling (RBS). A gateway can have one or more
LAN and WAN interfaces, such as Fast Ethernet, synchronous Serial interface, and ATM.

Gateways convert analog signals to digital and digital signals to analog. They might also be able
to handle several different types of codecs. These capabilities depend on the DSPs installed in
that gateway and its IOS feature set. DSPs also allow gateways to provide transcoding and
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Implementing VoIP Support in an Enterprise Network 45

conferencing services. Cisco IOS routers (gateways) support the most common VoIP gateway
signaling protocols, namely H.323, SIP, and MGCP.

SRST is a useful IOS feature on gateways at remote sites with no CallManager servers. The IP
phones at these types of sites communicate with and receive services from CallManager servers
at another branch, such as a central branch. If the IP connectivity between the central and remote
branch is lost, the IP phones at the remote branch are dysfunctional, unless the gateway of the
remote site has the SRST feature. With SRST, the IP phones at the remote site survive, can call
among themselves, and have limited features such as hold and transfer. However, the gateway with
SRST has to route all other calls to the PSTN.

The IOS on certain Cisco routers and switches has the Cisco Unified CallManager Express feature.
This feature allows the gateway to act as a complete CA (CallManager) for the IP phones at a
branch. This is not disaster recovery, but a permanent solution or option for smaller branches.

In addition to the features listed, the Cisco gateways offer fax relay, modem relay, and DTMF
relay services. Other features such as Hot Standby Routing Protocol (HSRP), Virtual Router
Redundancy Protocol (VRRP), and Gateway Load Balancing Protocol (GLBP) provide fault
tolerance and load sharing among redundant gateways.

Cisco Unified CallManager Functions
Cisco CallManager (CCM) is call processing software; it is the main component of the Cisco
Unified Communication System. CCM supports the MGCP, H.323, SIP, and SCCP IP Telephony
signaling protocols. Within the MGCP context, CCM acts as the CA and controls MGCP
gateways, and within the SCCP context, it controls the IP phones (Skinny Clients). CCM interacts
with H.323 and SIP devices. Cisco CallManager version 5.0 supports SIP clients, such as SIP-
based IP phones. CallManager servers form a cluster that provides the means for load sharing and
fault tolerance through redundancy. Some of the important services and functions that Cisco
Unified CallManager provides are these:

■ Call processing—CCM performs call routing, signaling, and accounting; furthermore, it has
bandwidth management and class of service (CoS) capabilities. (Class of service in this
context means enforcing call restrictions.)

■ Dial plan administration—CCM acts as the CA for MGCP gateways and IP phones;
therefore, the dial plan is administered, implemented, and enforced on CCM, and its clients
do not and need not have that information or capability.

■ Signaling and device control—Acting as the CA for MGCP gateways and IP phones, CCM
performs signaling for these devices and fully controls their configuration and behavior.
When an event occurs, the device informs CCM (the CA), and CCM in turn instructs the
device as to the action it should take in response to that event.
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46 Chapter 1: Cisco VoIP Implementations

■ Phone feature administration—IP phone configuration files are stored on the Cisco
CallManager server; therefore, IP phone administration is centralized. At the time of bootup
or when it is manually reset, an IP phone loads its configuration file from its own CallManager
server.

■ Directory and XML services—Directory services can be made available on Cisco
CallManager; IP phones can then perform lookup on the available directories. XML
applications can be administered as IP phone services on CCM.

■ Programming interface to external applications—Cisco Systems provides an application
programming interface (API) so that applications software can be written to work and
communicate with Cisco Unified CallManager. Examples of such applications already
developed are Cisco IP Communicator (a computer-based soft IP phone), Cisco Interactive
Voice Response System (IVR), Cisco Attendant Console, and Cisco Personal Assistant.

Enterprise IP Telephony Deployment Models
Many IP Telephony deployment options, utilizing Cisco Unified CallManager, are available. The
option that is suitable for an enterprise depends on the organization of that enterprise, its business
strategy, budget, and objectives. You can deploy the options presented here in combination (hybrid
models) or slightly differently. The four main options are as follows:

■ Single site

■ Multisite with centralized call processing

■ Multisite with distributed call processing

■ Clustering over WAN

Single-Site Model
In the single-site model, as the name implies, the enterprise has one site, and within that site it has
a Cisco CallManager cluster deployed. The local IP phones and perhaps MGCP gateways are
under the control of CCM, and CCM can communicate with H.323 and SIP devices. Calls that are
external to and from the site are routed through a gateway to the PSTN. The gateway DSPs can
provide codec, compression, transcoding, or conferencing resources. If the site has a WAN
connection to another place, the WAN connection is not used for IP Telephony purposes in this
model.

Multisite with Centralized Call Processing Model
In the multisite with centralized call processing model, the Cisco Unified CallManager (CCM)
cluster and application servers are placed at one of the sites—usually a main or central site. This
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Implementing VoIP Support in an Enterprise Network 47

IP Telephony solution spans multiple sites; in other words, all devices such as IP phones and
MGCP gateways at all sites are under the control of the CCM cluster at the central site. Notice that
even though call processing is centralized, DSP resources can be distributed.

If network connectivity, such as IP WAN, exists between sites, it carries signaling messages to and
from remote sites. Even if a device in a remote site calls another device within the same site,
signaling traffic must go through the WAN connection. However, VoIP packets (not signaling) go
through the WAN connection only for intersite calls.

Usually, each site has a PSTN connection that serves two purposes: It allows the site to make
outside calls, and it can act as an alternate route for when the WAN is down or is utilized to its
limit. CAC is used to prohibit too many active intersite calls from hindering data communications
or making the quality of calls drop. Administrators decide how many concurrent intersite calls
over the WAN connection are viable and configure CAC to deny permission to any new calls over
the WAN when the number of active intersite calls reaches that level. In those situations, a new
intersite call can either fail (reorder tone or annunciator message), or it can be transparently
rerouted through PSTN by means of automated alternate routing (AAR).

If a remote site temporarily loses its WAN connection to the central site, rendering its IP phones
useless, SRST is utilized on the gateway of that site. SRST is a feature available on Cisco gateways
that allows the IP phones at the remote site to stay active (in the absence of a path to their CCM
server) and be able to call each other within the site. SRST routes all calls through the PSTN when
the WAN connection is down.

Multisite with Distributed Call Processing Model
In the multisite with distributed call processing model, each site has its own Cisco Unified
CallManager cluster controlling all call processing aspects of that site—hence the term distributed
call processing. Application servers and DSP resources are also distributed at all sites. Sites, in
this case, do not depend on the call processing offered at another site. In distributed call
processing, each site has a CallManager cluster. Please note that the other resources (voice mail,
IPCC, IVR, DSP resources, etc.) can be centralized or distributed; while they’re normally
distributed, they do not have to be.

The WAN connection between the sites carries intersite data exchange, signaling, and VoIP
packets. However, when a device calls another device within its own site, no traffic is sent over the
WAN. CAC is still necessary to prohibit too many calls from going through the WAN connection.
Each site has PSTN connectivity, which serves two purposes: it allows outside enterprise calls for
each site, and it allows rerouting of intersite calls that cannot go through the WAN connection
(either due to CAC denial or WAN outage).
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48 Chapter 1: Cisco VoIP Implementations

This model is comparable to a legacy telephony model, where an enterprise would have a PBX
system at each site and, using telco services, the enterprise would connect each pair of PBX
systems at remote sites using tie-lines or trunks. In the distributed call processing model, an IP
Telephony trunk must be configured between each pair of CallManager clusters (IP PBX) to make
intersite calls possible. Examples of IP Telephony trunks that CCM supports are intercluster
trunks, H.323 trunks, and SIP trunks.

Clustering over WAN Model
This model uses only one Cisco CallManager cluster for all sites. However, not all servers of the
cluster are put in a single site together. Instead, the CCM servers, application servers, and DSP
resources are distributed to different locations to provide local service to their clients (such as IP
phones and gateways). The CCM servers need to communicate over the intersite IP WAN
connection to perform database synchronization and replication. For clustering over WAN to work
properly, the maximum round trip delay between each pair of servers within the cluster must be
less than 40 ms.

In this model, IP phones acquire services and are controlled by servers in the same site. IP WAN
carries signaling and voice packets only for intersite calls. CAC is needed to control the number
of calls utilizing the WAN connection. PSTN connection at each site is necessary for outside calls
and for AAR purposes.

Identifying Voice Commands in IOS Configurations
Cisco routers that have proper interfaces can be configured to provide connectivity between
analog or digital telephony devices over an IP network; they are called voice gateways in those
circumstances. Figure 1-16 shows two voice gateways, R1 and R2, each with an analog phone
connected to its FXS interface. To provide connectivity between the two phones over the IP
network, in addition to basic configurations, each of the routers (gateways) needs one plain old
telephone service (POTS) and one VoIP dial peer configured.

Figure 1-16 Two Sample Voice Gateways with Analog Phones Connected to Their FXS Interfaces
R1 R2
1/1/1 192.168.1.1 192.168.2.2 2/0/0
IP
FXS V V FXS

Extension 11 Extension 22

A dial peer is a Cisco IOS configuration that links or binds a telephone number to a local POTS
interface such as FXS or to a remote IP address; therefore, one POTS dial peer and one VoIP dial
peer exist. The series of dial peers configured on a gateway together form its VoIP call routing
table. The configurations of R1 and R2 shown in Example 1-1 and Example 1-2 take advantage of
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Implementing VoIP Support in an Enterprise Network 49

the default VoIP signaling protocol on Cisco gateways (H.323). If the phone on R1 goes off-hook
and, after receiving the dial tone, number 22 is dialed, R1 sends H.323 signaling (call setup)
messages to the R2 IP address 192.168.2.2. After the message from R1 is received and processed,
based on the dialed number 22, R2 sends a ring signal to interface 2/0/0 (the FXS port), and the
phone on R2 rings.

Example 1-1 R1 VoIP Configuration
Dial-peer voice 1 pots
destination-pattern 11
port 1/1/1

Dial-peer voice 2 voip
destination-pattern 22
session target ipv4:192.168.2.2

Example 1-2 R2 VoIP Configuration
Dial-peer voice 1 pots
destination-pattern 22
port 2/0/0

Dial-peer voice 2 voip
destination-pattern 11
session target ipv4:192.168.1.1

Call Admission Control (CAC)
Call admission control is a feature that is configured to limit the number of concurrent calls.
Usually, because the bandwidth of the WAN link is much less than LAN links, CAC is configured
so that WAN bandwidth does not get oversubscribed by VoIP calls. CAC complements QoS
configurations. For instance, if a strict priority queue with enough bandwidth for three voice calls
is configured on all routers between two phones, although there are fewer than four concurrent
calls, all will be good quality. What would happen if ten calls went active concurrently? If all the
VoIP traffic packets (RTP) must share the strict priority queue that is provisioned with enough
bandwidth for three calls, routers will drop many VoIP packets when there are ten active calls. The
packets that will be dropped belong to any or all active calls, indiscriminately. It is wrong to
believe that only packets associated to the calls beyond the third one will be dropped. As a result,
all calls can and probably will experience packet drops and, naturally, poor call quality. When
there are available and reserved resources for a certain number of concurrent calls, CAC must be
configured so that no more calls than the limit can go active. QoS features such as classification,
marking, congestion avoidance, congestion management, and so on provide priority services to
voice packets (RTP) but do not prevent their volume from exceeding the limit; for that, you
need CAC.
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50 Chapter 1: Cisco VoIP Implementations

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Benefits of packet telephony networks include usage of common infrastructure for voice and data,
lower transmission costs, more efficient usage of bandwidth, higher employee productivity, and
access to new communication devices. Main packet telephony components are phones, video end
points, gateways, MCUs, application servers, gatekeepers, and call agents. Voice gateways can
have analog interfaces such as FXS, FXO, and E&M; they may have digital interfaces such as BRI,
CT1/PRI, or CE1/PRI.

The main stages of a phone call are call setup, call maintenance, and call teardown. Call control
has two main types: centralized call control and distributed call control. H.323 and SIP are
examples of distributed VoIP call control protocol, whereas MGCP is an example of a centralized
VoIP call control protocol.

The steps involved in analog-to-digital voice conversion are sampling, quantization, encoding, and
compression. Digital-to-analog voice conversion steps include decompression, decoding, and
reconstruction of analog signal from pulse amplitude modulation (PAM) signal. Based on the
Nyquist theorem, the sampling rate must be at least twice the maximum analog audio signal
frequency. Quantization is the process of expressing the amplitude of a sampled signal by a binary
number. Several different ITU coding, decoding, and compression standards (called codecs) exist,
each of which requires a specific amount of bandwidth per call and yields a different quality.
Digital signal processors (DSP) convert analog voice signal to digital and vice versa; DSPs are
also voice termination points on voice gateways and are responsible for transcoding and
conferencing. Digitized voice is encapsulated in IP packets, which are routed and transported over
IP networks. RTP, UDP, and IP headers are added to digitized voice, and the data link layer header
is added to form a frame that is ready for transmission over media. Compressed RTP (cRTP)
can reduce or compress the RTP/UDP/IP headers when configured on the router interfaces on
both sides of a link; the reduction in overhead produced by cRTP is mainly beneficial and
recommended on links with less than 2 Mbps bandwidth.

The factors that influence the bandwidth requirement of each VoIP call over a link are packet rate,
packetization size, IP overhead, data link overhead, and tunneling overhead. The amount of voice
that is encapsulated in an IP packet affects the packet size and the packet rate. Smaller IP packets
mean more of them will be present, so the IP overhead elevates. Different data link layer protocols
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Foundation Summary 51

have varying amounts of header size and hence overhead. Tunneling and Security (IPsec) also add
overhead and hence increase the bandwidth demand for VoIP. Computing the total bandwidth
required on a link for each VoIP flow includes knowledge of the codec used, packetization period,
and all the overheads that will be present. Voice activity detection (VAD) can reduce bandwidth
requirements of VoIP calls and produce bandwidth savings of up to 35 percent.

The main components of enterprise voice implementations are IP phones, gateways, gatekeepers,
and Cisco Unified CallManager (CCM). Gateway, call agent, and DSP are among the capabilities
offered by Cisco integrated services routers (ISRs). CCM provides call processing, dial plan
administration, signaling and device control, phone feature administration, and access to applications
from IP phones. Enterprise IP Telephony deployment models are single site, multisite with
centralized call processing, multisite with distributed call processing, and clustering over WAN.
Dial peers are created with Cisco IOS commands configured on gateways to implement a local
dial plan. Call admission control (CAC) is configured to limit the number of concurrent VoIP calls.
It is required even in the presence of good QoS configurations so that WAN resources (bandwidth)
do not become oversubscribed.
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52 Chapter 1: Cisco VoIP Implementations

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. List at least three benefits of packet telephony networks.
2. List at least three important components of a packet telephony (VoIP) network.
3. List three types of analog interfaces through which legacy analog devices can connect to a
VoIP network.
4. List at least two digital interface options to connect VoIP equipment to PBXs or the PSTN.
5. List the three stages of a phone call.
6. What are the two main models of call control?
7. List the steps for converting analog signals to digital signals.
8. List the steps for converting digital signals to analog signals.
9. Based on the Nyquist theorem, what should be the minimum sampling rate of analog signals?
10. What are the two main quantization techniques?
11. Name and explain the quantization methods used in North America and in other countries.
12. Name at least three main codec/compression standards, and specify their bandwidth
requirements.
13. What is MOS?
14. What is a DSP?
15. Which TCP/IP protocols are responsible for transporting voice? What are the sizes of those
protocol headers?
16. What features does RTP provide to complement UDP?
17. What is cRTP?
18. List at least three factors that influence bandwidth requirements of VoIP.
19. What is the relationship between the packet rate and the packetization period?
20. What are the sizes of Ethernet, 802.1Q, Frame Relay, and Multilink PPP (MLP) overheads?
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Q&A 53

21. Name at least three tunneling and security protocols and their associated overheads.
22. Briefly list the steps necessary to compute the total bandwidth for a VoIP call.
23. What is VAD?
24. List at least three important components of enterprise voice implementations.
25. List at least three voice gateway functions on a Cisco router.
26. List the main functions of Cisco Unified CallManager.
27. List the four main enterprise IP Telephony deployment models.
28. What is CAC?
29. With QoS features in place, there can be up to ten concurrent VoIP calls over a company WAN
link. Is there a need for CAC? With no CAC, what will happen when there are more than ten
concurrent calls?
1763fm.book Page 54 Monday, April 23, 2007 8:58 AM

This part covers the following ONT exam topics. (To view the ONT exam
overview, visit http://www.cisco.com/web/learning/le3/current_exams/
642-845.html.)

■ Explain the necessity of QoS in converged networks (e.g., bandwidth, delay,
loss, etc.).
■ Describe strategies for QoS implementations (e.g. QoS Policy, QoS Models,
etc.).
■ Describe classification and marking (e.g., CoS, ToS, IP Precedence, DSCP, etc.).
■ Describe and configure NBAR for classification.
■ Explain congestion management and avoidance mechanisms (e.g., FIFO, PQ,
WRR, WRED, etc.).
■ Describe traffic policing and traffic shaping (i.e., traffic conditioners).
■ Describe Control Plane Policing.
■ Describe WAN link efficiency mechanisms (e.g., Payload/Header
Compression, MLP with interleaving, etc.).
■ Describe and configure QoS Pre-Classify.
■ Explain the functions and operations of AutoQoS.
■ Describe the SDM QoS Wizard.
■ Configure, verify, and troubleshoot AutoQoS implementations (i.e., MQC).
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Part II: Quality of Service

Chapter 2 IP Quality of Service

Chapter 3 Classification, Marking, and NBAR

Chapter 4 Congestion Management and Queuing

Chapter 5 Congestion Avoidance, Policing, Shaping, and Link Efficiency
Mechanisms

Chapter 6 Implementing QoS Pre-Classify and Deploying End-to-End QoS

Chapter 7 Implementing AutoQoS
1763fm.book Page 56 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:

■ Introduction to QoS

■ Identifying and Comparing QoS Models

■ QoS Implementation Methods
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CHAPTER 2
IP Quality of Service

This chapter provides the essential background, definitions, and concepts for you to start
learning IP quality of service (QoS). The following two chapters complement this one and
provide more coverage of this topic. It is probably safe to expect about 20 percent of the ONT
exam questions from this chapter.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 20-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 2-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 2-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“Introduction to QoS” 1–7
“Identifying and Comparing QoS Models” 8–13
“QoS Implementation Methods” 14–20
Total Score (20 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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58 Chapter 2: IP Quality of Service

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 15 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 16–17 overall score—Begin with the “Foundation Summary” section and then follow up
with the “Q&A” section at the end of the chapter.

■ 18 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following items is not considered one of four major issues and challenges facing
converged enterprise networks?
a. Available bandwidth
b. End-to-end delay
c. Delay variation (jitter)
d. Packet size
2. Which of the following is defined as the maximum bandwidth of a path?
a. The bandwidth of the link within the path that has the largest bandwidth
b. The bandwidth of the link within the path that has the smallest bandwidth
c. The total of all link bandwidths within the path
d. The average of all the link bandwidths within the path
3. Which of the following is not considered one of the main methods to tackle the bandwidth
availability problem?
a. Increase (upgrade) the link bandwidth.
b. Classify and mark traffic and deploy proper queuing mechanisms.
c. Forward large packets first.
d. Use compression techniques.
4. Which of the following is not considered a major delay type?
a. Queuing delay
b. CEF (Cisco Express Forwarding) delay
c. Serialization delay
d. Propagation delay
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“Do I Know This Already?” Quiz 59

5. Which of the following does not reduce delay for delay-sensitive application traffic?
a. Increasing (upgrade) the link bandwidth
b. Prioritizing delay-sensitive packets and forwarding important packets first
c. Layer 2 payload encryption
d. Header compression
6. Which of the following approaches does not tackle packet loss?
a. Increase (upgrade) the link bandwidth.
b. Increase the buffer space.
c. Provide guaranteed bandwidth.
d. Eliminate congestion avoidance.
7. Which of the following is not a major step in implementing QoS?
a. Apply access lists to all interfaces that process sensitive traffic
b. Identify traffic types and their requirements
c. Classify traffic based on the requirements identified
d. Define policies for each traffic class
8. Which of following is not one of the three main QoS models?
a. MPLS QoS
b. Differentiated services
c. Best effort
d. Integrated services
9. Which two of the following items are considered drawbacks of the best-effort model?
a. Inability to scale
b. Lack of service guarantee
c. Lack of service differentiation
d. Difficulty in implementing (complexity)
10. Which of the following is not a function that IntServ requires to be implemented on the
routers along the traffic path?
a. Admission control and policing
b. Classification
c. Queuing and scheduling
d. Fast switching
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60 Chapter 2: IP Quality of Service

11. Which of the following is the role of RSVP within the IntServ model?
a. Routing
b. Switching
c. Signaling/Bandwidth Reservation
d. Caching
12. Which of the following is not considered a benefit of the IntServ model?
a. Explicit end-to-end resource admission control
b. Continuous signaling per active flow
c. Per-request policy admission control
d. Signaling of dynamic port numbers
13. Which of the following is not true about the DiffServ model?
a. Within the DiffServ model, QoS policies (are deployed to) enforce differentiated treat-
ment of the defined traffic classes.
b. Within the DiffServ model, classes of traffic and the policies are defined based on busi-
ness requirements; you choose the service level for each traffic class.
c. Pure DiffServ makes extensive use of signaling; therefore, it is called hard QoS.
d. DiffServ is a scalable model.
14. Which of the following is not a QoS implementation method?
a. Cisco IOS CLI
b. MQC
c. Cisco AVVID (VoIP and Enterprise)
d. Cisco SDM QoS Wizard
15. Which of the following is not a major step in implementing QoS with MQC?
a. Define traffic classes using the class map.
b. Define QoS policies for the defined traffic classes using the policy map.
c. Apply the defined policies to each intended interface using the service-policy com-
mand.
d. Enable AutoQoS.
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“Do I Know This Already?” Quiz 61

16. Which of the following is the simplest QoS implementation method with an option
specifically for VoIP?
a. AutoQoS (VoIP)
b. CLI
c. MQC
d. Cisco SDM QoS Wizard
17. Select the most time-consuming and the least time-consuming QoS implementation methods.
a. CLI
b. MQC
c. AutoQoS
d. Cisco SDM QoS Wizard
18. What is the most significant advantage of MQC over CLI?
a. It requires little time to implement.
b. It requires little expertise to implement.
c. It has a GUI and interactive wizard.
d. It separates traffic classification from policy definition.
19. Before you enable AutoQoS on an interface, which two of the following must you ensure have
been configured on that interface?
a. Cisco modular QoS is configured.
b. CEF is enabled.
c. The SDM has been enabled.
d. The correct bandwidth on the interface is configured.
20. Select the item that is not a main service obtained from SDM QoS.
a. It enables you to implement QoS on the network.
b. It enables you to fine-tune QoS on the network.
c. It enables you to monitor QoS on the network.
d. It enables you to troubleshoot QoS on the network.
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62 Chapter 2: IP Quality of Service

Foundation Topics

Introduction to QoS
This section introduces the concept of QoS and discusses the four main issues in a converged
network that have QoS implications, as well as the Cisco IP QoS mechanisms and best practices
to deal with those issues. This section also introduces the three steps in implementing a QoS policy
on a network.

Converged Network Issues Related to QoS
A converged network supports different types of applications, such as voice, video, and data,
simultaneously over a common infrastructure. Accommodating these applications that have
different sensitivities and requirements is a challenging task on the hands of network engineers.

The acceptable end-to-end delay for the Voice over IP (VoIP) packets is 150 to 200 milliseconds
(ms). Also, the delay variation or jitter among the VoIP packets must be limited so that the buffers
at the receiving end do not become exhausted, causing breakup in the audio flow. In contrast, a
data application such as a file download from an FTP site does not have such a stringent delay
requirement, and jitter does not impose a problem for this type of application either. When
numerous active VoIP and data applications exist, mechanisms must be put in place so that while
critical applications function properly, a reasonable number of voice applications can remain
active and function with good quality (with low delay and jitter) as well.

Many data applications are TCP-based. If a TCP segment is dropped, the source retransmits it after
a timeout period is passed and no acknowledgement for that segment is received. Therefore,
TCP-based applications have some tolerance to packet drops. The tolerance of video and voice
applications toward data loss is minimal. As a result, the network must have mechanisms in place
so that at times of congestion, packets encapsulating video and voice receive priority treatment and
are not dropped.

Network outages affect all applications and render them disabled. However, well-designed
networks have redundancy built in, so that when a failure occurs, the network can reroute packets
through alternate (redundant) paths until the failed components are repaired. The total time it takes
to notice the failure, compute alternate paths, and start rerouting the packets must be short enough
for the voice and video applications not to suffer and not to annoy the users. Again, data appli-
cations usually do not expect the network recovery to be as fast as video and voice applications
expect it to be. Without redundancy and fast recovery, network outage is unacceptable, and
mechanisms must be put in place to avoid it.
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Introduction to QoS 63

Based on the preceding information, you can conclude that four major issues and challenges face
converged enterprise networks:

■ Available bandwidth—Many simultaneous data, voice, and video applications compete over
the limited bandwidth of the links within enterprise networks.

■ End-to-end delay—Many actions and factors contribute to the total time it takes for data or
voice packets to reach their destination. For example, compression, packetization, queuing,
serialization, propagation, processing (switching), and decompression all contribute to the
total delay in VoIP transmission.

■ Delay variation (jitter)—Based on the amount of concurrent traffic and activity, plus the
condition of the network, packets from the same flow might experience a different amount of
delay as they travel through the network.

■ Packet loss—If volume of traffic exhausts the capacity of an interface, link, or device, packets
might be dropped. Sudden bursts or failures are usually responsible for this situation.

The sections that follow explore these challenges in detail.

Available Bandwidth
Packets usually flow through the best path from source to destination. The maximum bandwidth
of that path is equal to the bandwidth of the link with the smallest bandwidth. Figure 2-1 shows
that R1-R2-R3-R4 is the best path between the client and the server. On this path, the maximum
bandwidth is 10 Mbps because that is the bandwidth of the link with the smallest bandwidth on
that path. The average available bandwidth is the maximum bandwidth divided by the number of
flows.

Figure 2-1 Maximum Bandwidth and Average Available Bandwidth Along the Best Path (R1-R2-R3-R4)
Between the Client and Server
Bandwidth(Max) = Min(10 Mbps, 10 Mbps, 100 Mbps) = 10 Mbps

Bandwidth(Avail) = Bandwidth(Max)/Flows

10 Mbps

R2 R3
100 Mbps 10 Mbps 100 Mbps 100 Mbps

R1 R4
Client Server
1 Mbps 10 Mbps

R7 R5
10 Mbps 100 Mbps
R6
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64 Chapter 2: IP Quality of Service

Lack of sufficient bandwidth causes delay, packet loss, and poor performance for applications.
The users of real-time applications (voice and video) detect this right away. You can tackle the
bandwidth availability problem in numerous ways:

■ Increase (upgrade) link bandwidth—This is effective, but it is costly.

■ Classify and mark traffic and deploy proper queuing mechanisms—Forward important
packets first.

■ Use compression techniques—Layer 2 payload compression, TCP header compression, and
cRTP are some examples.

Increasing link bandwidth is undoubtedly beneficial, but it cannot always be done quickly, and it
has cost implications. Those who just increase bandwidth when necessary notice that their solution
is not very effective at times of heavy traffic bursts. However, in certain scenarios, increasing link
bandwidth might be the first action necessary (but not the last).

Classification and marking of the traffic, combined with congestion management, is an effective
approach to providing adequate bandwidth for enterprise applications.

Link compression, TCP header compression, and RTP header compression are all different
compression techniques that can reduce the bandwidth consumed on certain links, and therefore
increase throughput. Cisco IOS supports the Stacker and Predictor Layer 2 compression algorithms
that compress the payload of the packet. Usage of hardware compression is always preferred over
software-based compression. Because compression is CPU intensive and imposes yet another
delay, it is usually recommended only on slow links.

NOTE Most compression mechanisms must be configured on a link-by-link basis—in other
words, on both ends of each link. Classification, marking, compression, and advanced queuing
mechanisms are discussed in Chapters 3, 4, and 5 in detail.

End-to-End Delay
There are different types of delay from source to destination. End-to-end delay is the sum of those
different delay types that affect the packets of a certain flow or application. Four of the important
types of delay that make up end-to-end delay are as follows:

■ Processing delay

■ Queuing delay

■ Serialization delay

■ Propagation delay
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Introduction to QoS 65

Processing delay is the time it takes for a device such as a router or Layer 3 switch to perform all
the tasks necessary to move a packet from the input (ingress) interface to the output (egress)
interface. The CPU type, CPU utilization, switching mode, router architecture, and configured
features on the device affect the processing delay. For example, packets that are distributed-CEF
switched on a versatile interface processor (VIP) card cause no CPU interrupts.

Queuing delay is the amount of time that a packet spends in the output queue of a router interface.
The busyness of the router, the number of packets waiting in the queue, the queuing discipline, and
the interface bandwidth all affect the queuing delay.

Serialization delay is the time it takes to send all the bits of a frame to the physical medium for
transmission across the physical layer. The time it takes for the bits of that frame to cross the
physical link is called the propagation delay. Naturally, the propagation delay across different
media can be significantly different. For instance, the propagation delay on a high-speed optical
connection such as OC-192 is significantly lower than the propagation delay on a satellite-based
link.

NOTE In best-effort networks, while serialization and propagation delays are fixed, the
processing and queuing delays are variable and unpredictable.
Other types of delay exist, such as WAN delay, compression and decompression delay, and de-
jitter delay.

Delay Variation
The variation in delays experienced by the packets of the same flow is called delay variation or
jitter. Packets of the same flow might not arrive at the destination at the same rate that they were
released. These packets, individually and independent from each other, are processed, queued, de-
queued, and so on. Therefore, they might arrive out of sequence, and their end-to-end delays might
vary. For voice and video packets, it is essential that at the destination point, the packets are
released to the application in the correct order and at the same rate that they were released at the
source. The de-jitter buffer serves that purpose. As long as the delay variation is not too much, at
the destination point, the de-jitter buffer holds packets, sorts them, and releases them to the
application based on the Real-Time Transport Protocol (RTP) time stamp on the packets. Because
the buffer compensates the jitter introduced by the network, it is called the de-jitter buffer.

Average queue length, packet size, and link bandwidth contribute to serialization and propagation
delay. You can reduce delay by doing some or all of the following:

■ Increase (upgrade) link bandwidth—This is effective as the queue sizes drop and queuing
delays soar. However, upgrading link capacity (bandwidth) takes time and has cost implications,
rendering this approach unrealistic at times.
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66 Chapter 2: IP Quality of Service

■ Prioritize delay-sensitive packets and forward important packets first—This might
require packet classification or marking, but it certainly requires deployment of a queuing
mechanism such as weighted fair queuing (WFQ), class-based weighted fair queuing
(CBWFQ), or low-latency queuing (LLQ). This approach is not as costly as the previous
approach, which is a bandwidth upgrade.

■ Reprioritize packets—In certain cases, the packet priority (marking) has to change as the
packet enters or leaves a device. When packets leave one domain and enter another, this
priority change might have to happen. For instance, the packets that leave an enterprise
network with critical marking and enter a provider network might have to be reprioritized
(remarked) to best effort if the enterprise is only paying for best effort service.

■ Layer 2 payload compression—Layer 2 compression reduces the size of the IP packet (or
any other packet type that is the frame’s payload), and it frees up available bandwidth on that
link. Because complexity and delay are associated with performing the compression, you
must ensure that the delay reduced because of compression is more than the delay introduced
by the compression complexity. Note that payload compression leaves the frame header in
tact; this is required in cases such as frame relay connections.

■ Use header compression—RTP header compression (cRTP) is effective for VoIP packets,
because it greatly improves the overhead-to-payload ratio. cRTP is recommended on slow
(less than 2 Mbps) links. Header compression is less CPU-intensive than Layer 2 payload
compression.

Packet Loss
Packet loss occurs when a network device such as a router has no more buffer space on an interface
(output queue) to hold the new incoming packets and it ends up dropping them. A router may drop
some packets to make room for higher priority ones. Sometimes an interface reset causes packets
to be flushed and dropped. Packets are dropped for other reasons, too, including interface overrun.

TCP resends the dropped packets; meanwhile, it reduces the size of the send window and slows
down at times of congestion and high network traffic volume. If a packet belonging to a UDP-
based file transfer (such as TFTP) is dropped, the whole file might have to be resent. This creates
even more traffic on the network, and it might annoy the user. Application flows that do not use
TCP, and therefore are more drop-sensitive, are called fragile flows.

During a VoIP call, packet loss results in audio breakup. A video conference will have jerky
pictures and its audio will be out of synch with the video if packet drops or extended delays occur.
When network traffic volume and congestion are heavy, applications experience packet drops,
extended delays, and jitter. Only with proper QoS configuration can you avoid these problems or
at least limit them to low-priority packets.
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Introduction to QoS 67

On a Cisco router, at times of congestion and packet drops, you can enter the show interface
command and observe that on some or all interfaces, certain counters such as those in the
following list have incremented more than usual (baseline):

■ Output drop—This counter shows the number of packets dropped, because the output queue
of the interface was full at the time of their arrival. This is also called tail drop.

■ Input queue drop—If the CPU is overutilized and cannot process incoming packets, the
input queue of an interface might become full, and the number of packets dropped in this
scenario will be reported as input queue drops.

■ Ignore—This is the number of frames ignored due to lack of buffer space.

■ Overrun—The CPU must allocate buffer space so that incoming packets can be stored and
processed in turn. If the CPU becomes too busy, it might not allocate buffer space quickly
enough and end up dropping packets. The number of packets dropped for this reason is called
overruns.

■ Frame error—Frames with cyclic redundancy check (CRC) error, runt frames (smaller than
minimum standard), and giant frames (larger than the maximum standard) are usually
dropped, and their total is reported as frame errors.

You can use many methods, all components of QoS, to tackle packet loss. Some methods protect
packet loss from all applications, whereas others protect specific classes of packets from packet
loss only. The following are examples of approaches that packet loss can merit from:

■ Increase (upgrade) link bandwidth—Higher bandwidth results in faster packet departures
from interface queues. If full queue scenarios are prevented, so are tail drops and random
drops (discussed later).

■ Increase buffer space—Network engineers must examine the buffer settings on the interfaces
of network devices such as routers to see if their sizes and settings are appropriate. When
dealing with packet drop issues, it is worth considering an increase of interface buffer space
(size). A larger buffer space allows better handling of traffic bursts.

■ Provide guaranteed bandwidth—Certain tools and features such as CBWFQ and LLQ
allow the network engineers to reserve certain amounts of bandwidth for a specific class of
traffic. As long as enough bandwidth is reserved for a class of traffic, packets of such a class
will not become victims of packet drop.

■ Perform congestion avoidance—To prevent a queue from becoming full and starting tail
drop, you can deploy random early detection (RED) or weighted random early detection
(WRED) to drop packets from the queue before it becomes full. You might wonder what the
merit of that deployment would be. When packets are dropped before a queue becomes full,
the packets can be dropped from certain flows only; tail drop loses packets from all flows.
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68 Chapter 2: IP Quality of Service

With WRED, the flows that lose packets first are the lowest priority ones. It is hoped that the
highest priority packet flows will not have drops. Drops due to deployment of RED/WRED
slow TCP-based flows, but they have no effect on UDP-based flows.

Most companies that connect remote sites over a WAN connection transfer both TCP- and UDP-
based application data between those sites. Figure 2-2 displays a company that sends VoIP traffic
as well as file transfer and other application data over a WAN connection between its remote
branch and central main branch. Note that, at times, the collection of traffic flows from the remote
branch intending to cross R2 and the WAN connection (to go to the main central branch) can reach
high volumes.

Figure 2-2 Solutions for Packet Loss and Extended Delay
Low
Bandwidth

Remote Main Branch
WAN
Branch LAN R2 R1 LAN

High
Volume
Congestion avoidance features such as WRED,
Low-Latency Queuing (LLQ), and
RTP Header Compression (cRTP) on R2
can ease or eliminate packet loss and extended delays
on this branch office edge (WAN) router.

Figure 2-2 displays the stated scenario that leads to extended delay and packet loss. Congestion
avoidance tools trigger TCP-based applications to throttle back before queues and buffers become
full and tail drops start. Because congestion avoidance features such as WRED do not trigger
UDP-based applications (such as VoIP) to slow down, for those types of applications, you must
deploy other features, including compression techniques such as cRTP and advanced queuing such
as LLQ.

Definition of QoS and the Three Steps to Implementing It
Following is the most recent definition that Cisco educational material provides for QoS:

QoS is the ability of the network to provide better or special service to a set of users or
applications or both to the detriment of other users or applications or both.

The earliest versions of QoS tools protected data against data. For instance, priority queuing made
sure packets that matched an access list always had the right of way on an egress interface. Another
example is WFQ, which prevents small packets from waiting too long behind large packets on an
egress interface outbound queue. When VoIP started to become a serious technology, QoS tools
were created to protect voice from data. An example of such a tool is RTP priority queue.
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Introduction to QoS 69

RTP priority queue is reserved for RTP (encapsulating voice payload). RTP priority queuing ensures
that voice packets receive right of way. If there are too many voice streams, data applications begin
experiencing too much delay and too many drops. Strict priority queue (incorporated in LLQ) was
invented to limit the bandwidth of the priority queue, which is essentially dedicated to voice
packets. This technique protects data from voice; too many voice streams do not downgrade the
quality of service for data applications. However, what if there are too many voice streams? All
the voice calls and streams must share the bandwidth dedicated to the strict priority queue that is
reserved for voice packets. If the number of voice calls exceeds the allocated resources, the quality
of those calls will drop. The solution to this problem is call admission control (CAC). CAC
prevents the number of concurrent voice calls from going beyond a specified limit and hurting the
quality of the active calls. CAC protects voice from voice. Almost all the voice requirements apply
to video applications, too; however, the video applications are more bandwidth hungry.

Enterprise networks must support a variety of applications with diverse bandwidth, drop, delay,
and jitter expectations. Network engineers, by using proper devices, Cisco IOS features, and
configurations, can control the behavior of the network and make it provide predictable service to
those applications. The existence of voice, video, and multimedia applications in general not only
adds to the bandwidth requirements in networks but also adds to the challenges involved in having
to provide granular and strictly controlled delay, jitter, and loss guarantees.

Implementing QoS
Implementing QoS involves three major steps:

Step 1 Identifying traffic types and their requirements
Step 2 Classifying traffic based on the requirements identified
Step 3 Defining policies for each traffic class
Even though many common applications and protocols exist among enterprise networks, within
each network, the volumes and percentages of those traffic types vary. Furthermore, each enter-
prise might have its own unique application types in addition to the common ones. Therefore, the
first step in implementing QoS in an enterprise is to study and discover the traffic types and define
the requirements of each identified traffic type. If two, three, or more traffic types have identical
importance and requirements, it is unnecessary to define that many traffic classes. Traffic
classification, which is the second step in implementing QoS, will define a few traffic classes,
not hundreds. The applications that end up in different traffic classes have different requirements;
therefore, the network must provide them with different service types. The definition of how each
traffic class is serviced is called the network policy. Defining and deploying the network QoS
policy for each class is Step 3 of implementing QoS. The three steps of implementing QoS on a
network are explained next.
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70 Chapter 2: IP Quality of Service

Step 1: Identifying Traffic Types and Their Requirements
Identifying traffic types and their requirements, the first step in implementing QoS, is composed
of the following elements or substeps:

■ Perform a network audit—It is often recommended that you perform the audit during the
busy hour (BH) or congestion period, but it is also important that you run the audit at other
times. Certain applications are run during slow business hours on purpose. There are scientific
methods for identifying the busy network moments, for example, through statistical sampling
and analysis, but the simplest method is to observe CPU and link utilizations and conduct the
audit during the general peak periods.

■ Perform a business audit and determine the importance of each application—The
business model and goals dictate the business requirements. From that, you can derive the
definition of traffic classes and the requirements for each class. This step considers whether
delaying or dropping packets of each application is acceptable. You must determine the
relative importance of different applications.

■ Define the appropriate service levels for each traffic class—For each traffic class, within
the framework of business objectives, a specific service level can define tangible resource
availability or reservations. Guaranteed minimum bandwidth, maximum bandwidth, guaranteed
end-to-end maximum delay, guaranteed end-to-end maximum jitter, and comparative drop
preference are among the characteristics that you can define for each service level. The final
service level definitions must meet business objectives and satisfy the comfort expectations
of the users.

Step 2: Classifying Traffic Based on the Requirements Identified
The definition of traffic classes does not need to be general; it must include the traffic (application)
types that were observed during the network audit step. You can classify tens or even hundreds of
traffic variations into very few classes. The defined traffic classes must be in line with business
objectives. The traffic or application types within the same class must have common requirements
and business requirements. The exceptions to this rule are the applications that have not been
identified or scavenger-class traffic.

Voice traffic has specific requirements, and it is almost always in its own class. With Cisco LLQ,
VoIP is assigned to a single class, and that class uses a strict priority queue (a priority queue with
strict maximum bandwidth) on the egress interface of each router. Many case studies have shown
the merits of using some or all of the following traffic classes within an enterprise network:

■ Voice (VoIP) class—Voice traffic has specific bandwidth requirements, and its delay and
drops must be eliminated or at least minimized. Therefore, this class is the highest priority
class but has limited bandwidth. VoIP packet loss should remain below 1% and the goal for
its end-to-end delay must be 150 ms.
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Introduction to QoS 71

■ Mission-critical traffic class—Critical business applications are put in one or two classes.
You must identify the bandwidth requirements for them.

■ Signaling traffic class—Signaling traffic, voice call setup and teardown for example, is often
put in a separate class. This class has limited bandwidth expectations.

■ Transactional applications traffic class—These applications, if present, include interactive,
database, and similar services that need special attention. You must also identify the bandwidth
requirements for them. Enterprise Resource Planning (ERP) applications such as Peoplesoft
and SAP are examples of these types of applications.

■ Best-effort traffic class—All the undefined traffic types are considered best effort and
receive the remainder of bandwidth on an interface.

■ Scavenger traffic class—This class of applications will be assigned into one class and be
given limited bandwidth. This class is considered inferior to the best-effort traffic class. Peer-
to-peer file sharing applications are put in this class.

Step 3: Defining Policies for Each Traffic Class
After the traffic classes have been formed based on the network audit and business objectives, the
final step of implementing QoS in an enterprise is to provide a network-wide definition for the QoS
service level that must be assigned to each traffic class. This is called defining a QoS policy, and
it might include having to complete the following tasks:

■ Setting a maximum bandwidth limit for a class

■ Setting a minimum bandwidth guarantee for a class

■ Assigning a relative priority level to a class

■ Applying congestion management, congestion avoidance, and many other advanced QoS
technologies to a class.

To provide an example, based on the traffic classes listed in the previous section, Table 2-2 defines
a practical QoS policy.

Table 2-2 Defining QoS Policy for Set Traffic Classes

Min/Max Special QoS
Class Priority Queue Type Bandwidth Technology
Voice 5 Priority 1 Mbps Min Priority queue

1 Mbps Max
Business mission 4 CBWFQ 1 Mbps Min CBWFQ
critical
continues
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72 Chapter 2: IP Quality of Service

Table 2-2 Defining QoS Policy for Set Traffic Classes (Continued)

Min/Max Special QoS
Class Priority Queue Type Bandwidth Technology
Signaling 3 CBWFQ 400 Kbps Min CBWFQ
Transactional 2 CBWFQ 1 Mbps Min CBWFQ
Best-effort 1 CBWFQ 500 Kbps Max CBWFQ

CB-Policing
Scavenger 0 CBWFQ Max 100 Kbps CBWFQ

+CB-Policing

WRED

Identifying and Comparing QoS Models
This section discusses the three main QoS models, namely best-effort, Integrated Services, and
Differentiated Services. The key features, and the benefits and drawbacks of each of these QoS
models, are explained in turn.

Best-Effort Model
The best-effort model means that no QoS policy is implemented. It is natural to wonder why this
model was not called no-effort. Within this model, packets belonging to voice calls, e-mails, file
transfers, and so on are treated as equally important; indeed, these packets are not even differentiated.
The basic mail delivery by the post office is often used as an example for the best-effort model,
because the post office treats all letters as equally important.

The best-effort model has some benefits as well as some drawbacks. Following are the main
benefits of this model:

■ Scalability—The Internet is a best-effort network. The best-effort model has no scalability
limit. The bandwidth of router interfaces dictates throughput efficiencies.

■ Ease—The best-effort model requires no special QoS configuration, making it the easiest and
quickest model to implement.

The drawbacks of the best-effort model are as follows:

■ Lack of service guarantee—The best-effort model makes no guarantees about packet
delivery/loss, delay, or available bandwidth.

■ Lack of service differentiation—The best-effort model does not differentiate packets that
belong to applications that have different levels of importance from the business perspective.
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Identifying and Comparing QoS Models 73

Integrated Services Model
The Integrated Services (IntServ) model, developed in the mid-1990s, was the first serious attempt
to provide end-to-end QoS, which was demanded by real-time applications. IntServ is based on
explicit signaling and managing/reserving network resources for the applications that need it and
demand it. IntServ is often referred to as Hard-QoS, because Hard-QoS guarantees characteristics
such as bandwidth, delay, and packet loss, thereby providing a predictable service level. Resource
Reservation Protocol (RSVP) is the signaling protocol that IntServ uses. An application that has a
specific bandwidth requirement must wait for RSVP to run along the path from source to destination,
hop by hop, and request bandwidth reservation for the application flow. If the RSVP attempt to
reserve bandwidth along the path succeeds, the application can begin operating. While the
application is active, along its path, the routers provide the bandwidth that they have reserved for
the application. If RSVP fails to successfully reserve bandwidth hop by hop all the way from
source to destination, the application cannot begin operating.

IntServ mimics the PSTN model, where every call entails end-to-end signaling and securing
resources along the path from source to destination. Because each application can make a unique
request, IntServ is a model that can provide multiple service levels. Within the Cisco QoS frame-
work, RSVP can act both as a signaling mechanism and as a CAC mechanism. If an RSVP attempt
to secure and reserve resources for a voice call fails, the call does not get through. Controlled
volume services within the Cisco IOS QoS feature set are provided by RSVP and advanced
queuing mechanisms such as LLQ. The Guaranteed Rate service type is offered by deploying
RSVP and LLQ. Controlled Load service is provided by RSVP and WRED.

For a successful implementation of IntServ, in addition to support for RSVP, enable the following
features and functions on the routers or switches within the network:

Admission control—Admission control responds to application requests for end-to-end
resources. If the resources cannot be provided without affecting the existing applications, the
request is turned down.

Classification—The traffic belonging to an application that has made resource reservations must
be classified and recognized by the transit routers so that they can furnish appropriate service to
those packets.

Policing—It is important to measure and monitor that applications do not exceed resource
utilization beyond their set profiles. Rate and burst parameters are used to measure the behavior
of an application. Depending on whether an application conforms to or exceeds its agreed-upon
resource utilizations, appropriate action is taken.

Queuing—It is important for network devices to be able to hold packets while processing and
forwarding others. Different queuing mechanisms store and forward packets in unique ways.
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74 Chapter 2: IP Quality of Service

Scheduling—Scheduling works in conjunction with queuing. If there are multiple queues on an
interface, the amount of data that is dequeued and forwarded from each queue at each cycle, hence
the relative attention that each queue gets, is called the scheduling algorithm. Scheduling is
enforced based on the queuing mechanism configured on the router interface.

When IntServ is deployed, new application flows are admitted until requested resources can no
longer be furnished. Any new application will fail to start because the RSVP request for resources
will be rejected. In this model, RSVP makes the QoS request for each flow. This request includes
identification for the requestor, also called the authorized user or authorization object, and the
needed traffic policy, also called the policy object. To allow all intermediate routers between
source and destination to identify each flow, RSVP provides the flow parameters such as IP
addresses and port numbers. The benefits of the IntServ model can be summarized as follows:

■ Explicit end-to-end resource admission control

■ Per-request policy admission control

■ Signaling of dynamic port numbers

Some drawbacks to using IntServ exist, the most important of which are these:

■ Each active flow has a continuous signaling. This overhead can become substantially large as
the number of flows grows. This is because of the stateful architecture of RSVP.

■ Because each flow is tracked and maintained, IntServ as a flow-based model is not considered
scalable for large implementations such as the Internet.

Differentiated Services Model
Differentiated Services (DiffServ) is the newest of the three QoS models, and its development has
aimed to overcome the limitations of its predecessors. DiffServ is not a guaranteed QoS model,
but it is a highly scalable one. The Internet Engineering Task Force (IETF) description and dis-
cussion on DiffServ are included in RFCs 2474 and 2475. Whereas IntServ has been called the
“Hard QoS” model, DiffServ has been called the “Soft QoS” model. IntServ, through usage of
signaling and admission control, is able to either deny application of requested resources or admit
it and guarantee the requested resources.

Pure DiffServ does not use signaling; it is based on per-hop behavior (PHB). PHB means that each
hop in a network must be preprogrammed to provide a specific level of service for each class of
traffic. PHB then does not require signaling as long as the traffic is marked to be identified as
one of the expected traffic classes. This model is more scalable because signaling and status
monitoring (overhead) for each flow are not necessary. Each node (hop) is prepared to deal with a
limited variety of traffic classes. This means that even if thousands of flows become active, they
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Identifying and Comparing QoS Models 75

are still categorized as one of the predefined classes, and each flow will receive the service level
that is appropriate for its class. The number of classes and the service level that each traffic class
should receive are decided based on business requirements.

Within the DiffServ model, traffic is first classified and marked. As the marked traffic flows
through the network nodes, the type of service it receives depends on its marking. DiffServ can
protect the network from oversubscription by using policing and admission control techniques as
well. For example, in a typical DiffServ network, voice traffic is assigned to a priority queue that
has reserved bandwidth (through LLQ) on each node. To prohibit too many voice calls from
becoming active concurrently, you can deploy CAC. Note that all the voice packets that belong to
the admitted calls are treated as one class.

The DiffServ model is covered in detail in Chapters 3, 4, and 5. Remember the following three
points about the DiffServ model:

■ Network traffic is classified.

■ QoS policies enforce differentiated treatment of the defined traffic classes.

■ Classes of traffic and the policies are defined based on business requirements; you choose the
service level for each traffic class.

The main benefit of the DiffServ model is its scalability. The second benefit of the DiffServ model
is that it provides a flexible framework for you to define as many service levels as your business
requirements demand. The main drawback of the DiffServ model is that it does not provide an
absolute guarantee of service. That is why it is associated with the term Soft QoS. The other
drawback of this model is that several complex mechanisms must be set up consistently on all the
elements of the network for the model to yield the desired results.

Following are the benefits of DiffServ:

■ Scalability

■ Ability to support many different service levels

The drawbacks of DiffServ are as follows:

■ It cannot provide an absolute service guarantee.

■ It requires implementation of complex mechanisms through the network.
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76 Chapter 2: IP Quality of Service

QoS Implementation Methods
This section explores the four main QoS implementation methods, namely CLI, MQC, Cisco
AutoQoS, and SDM QoS Wizard. A high-level explanation of each QoS implementation method
and the advantages and disadvantages of each are provided in turn.

Legacy Command-Line Interface (CLI)
Legacy CLI was the method used up to about six years ago to implement QoS on network devices.
Legacy CLI requires configuration of few to many lines of code that for the most part would have
to be applied directly at the interface level. Configuration of many interfaces required a lot of
typing or cutting and pasting. Maintaining consistency, minimizing errors, and keeping the
configuration neat and understandable were difficult to do using legacy CLI.

Legacy CLI configuration required the user to log into the router via console using a terminal (or
a terminal emulator) or via a virtual terminal line using a Telnet application. Because it was a
nonmodular method, legacy CLI did not allow users to completely separate traffic classification
from policy definition and how the policy is applied. Legacy CLI was also more error prone and
time consuming. Today, people still use CLI, but mostly to fine-tune the code generated by
AutoQoS, which will be discussed later.

You began legacy CLI configuration by identifying, classifying, and prioritizing the traffic. Next,
you had to select one of the available and appropriate QoS tools such as link compression or an
available queuing mechanism such as custom or priority queuing. Finally, you had to enter from
a few to several lines of code applying the selected QoS mechanisms for one or many interfaces.

Modular QoS Command-Line Interface (MQC)
Cisco introduced MQC to address the shortcomings of the legacy CLI and to allow utilization of
the newer QoS tools and features available in the modern Cisco IOS. With the MQC, traffic
classification and policy definition are done separately. Traffic policies are defined after traffic
classes. Different policies might reference the same traffic classes, thereby taking advantage of the
modular and reusable code. When one or more policies are defined, you can apply them to many
interfaces, promoting code consistency and reuse.

MQC is modular, more efficient, and less time consuming than legacy CLI. Most importantly,
MQC separates traffic classification from policy definition, and it is uniform across major Cisco
IOS platforms. With MQC, defined policies are applied to interfaces rather than a series of raw
CLI commands being applied to interfaces.
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QoS Implementation Methods 77

Implementing QoS with MQC involves three major steps:

Step 1 Define traffic classes using the class-map command. This step divides the
identified network traffic into a number of named classes.
Step 2 Define QoS policies for the defined traffic classes using the policy-map
command. This step involves QoS features being linked to traffic classes. It
defines the treatment of the defined classes of traffic.
Step 3 Apply the defined policies in the inbound or outbound direction to each
intended interface, subinterface, or circuit, using the service-policy
command. This step defines where the defined policies are applied.
Each class map, which has a case-sensitive name, is composed of one or more match statements.
One or all of the match statements must be matched, depending on whether class map contains
the match-any or the match-all command. When neither match-any nor match-all is specified
on the class-map statement, match-all applies by default.

Example 2-1 shows two class maps. The first class map is called VOIP. This class map specifies
that traffic matching access list 100 is classified as VOIP. The second class map is called Business-
Application. It specifies that traffic matching access-list 101 is classified as Business-Application.

Example 2-1 Class Maps
class-map VOIP
match access-group 100
!
class-map Business-Application
match access-group 101
!

In Example 2-1, note that both of the class maps have only one match statement, and neither
match-all nor match-any is specified, which defaults to match-all. When only one match
statement exists, match-all and match-any yield the same result. However, when more than one
match statement exists, using match-any or match-all makes a big difference. match-any means
only one of the match statements needs to be met, and match-all means all the match statements
must be met to bind the packet to the class.

NOTE The opposite of the match condition is the match not condition.

You create traffic policies by associating required QoS features to traffic classes defined by class
maps; you use the policy-map command to do that. A policy map has a case-sensitive name and
can associate QoS policies for up to 256 traffic classes (each defined by a class map). Example 2-2
exhibits a policy map called Enterprise-Policy. This policy map specifies that traffic classified as
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78 Chapter 2: IP Quality of Service

VOIP is assigned to a priority queue that has a bandwidth guarantee of 256 Kbps. Enterprise-
Policy also states that the traffic classified as Business-Application is assigned to a WFQ with a
bandwidth guarantee of 256 Kbps. According to this policy map, all other traffic, classified as
class-default, will be assigned to a queue that gets the rest of the available bandwidth, and a WFQ
policy will be applied to it.

Example 2-2 Policy Map
policy-map Enterprise-Policy
class VOIP
priority 256
class Business-Application
bandwidth 256
class class-default
fair-queue
!

If you configure a policy map that includes a class statement followed by the name of a nonexistent
class map, as long as the statement includes a condition, a class map is created and inserted into
the configuration with that name automatically. If, within a policy map, you do not refer to the
class-default (and do not configure it), any traffic that the defined classes do not match will still
be treated as class-default. The class-default gets no QoS guarantees and can use a FIFO or
a WFQ.

A policy map is applied on an interface (or subinterface, virtual template, or circuit) in the
outbound or inbound direction using the service-policy command (and the direction specified
using the input or output keywords). You can apply a defined and configured policy map to more
than one interface. Reusing class maps and policy maps is highly encouraged because it promotes
standardization and reduces the chance of errors. Example 2-3 shows that the policy map
Enterprise-Policy is applied to the serial 1/0 interface of a router on the outbound direction.

Example 2-3 Service-Policy
interface serial 1/0
service-policy output Enterprise-Policy
!

The following commands allow you to display and verify QoS classes and policies you have
configured using the MQC:

show class-map—This command displays all the configured class maps.

show policy-map—This command displays all the configured policy maps.
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QoS Implementation Methods 79

show policy-map interface interface—This command displays the policy map that is applied to
a particular interface using the service-policy command. This command also displays QoS
interface statistics.

AutoQoS
AutoQoS is a value-added feature of Cisco IOS. After it is enabled on a device, AutoQoS auto-
matically generates QoS configuration commands for the device. The initial release of AutoQoS
(Auto QoS VoIP) focused on generating commands that made the device ready for VoIP and IP
Telephony. Later, the AutoQoS Discovery feature was introduced. The next generation of
AutoQoS that takes advantage of AutoQoS discovery is called AutoQoS for the Enterprise.
AutoQoS Discovery, as its name implies, analyzes live network traffic for as long as you let it run
and generates traffic classes based on the traffic it has processed. Next, you enable the AutoQoS
feature. AutoQoS uses the traffic classes (class maps) formed by AutoQoS Discovery to generate
network QoS policy (policy map), and it applies the policy. Based on the interface type, AutoQoS
might also add features such as fragmentation and interleaving, multilink, and traffic shaping to
the interface configuration.

The main advantage of AutoQoS is that it simplifies the task of QoS configuration. Network
administrators who lack in-depth knowledge of QoS commands and features can use AutoQoS to
implement those features consistently and accurately. AutoQoS participates in all the main aspects
of QoS deployment:

■ Classification—AutoQoS for the Enterprise, through AutoQoS Discovery, automatically
discovers applications and protocols (using Network Based Application Recognition, or
NBAR). It uses Cisco Discovery Protocol (CDP) to check whether an IP phone is attached to
a switch port.

■ Policy generation—It provides appropriate treatment of traffic by the QoS policies that it
auto-generates. AutoQoS checks interface encapsulations, and accordingly, it considers usage
of features such as fragmentation, compression, and traffic shaping. Access lists, class maps,
and policy maps, which normally have to be entered manually, are automatically generated
by AutoQoS.

■ Configuration—It is enabled by entering only one command, auto qos, at the interface. In a
matter of seconds, proper commands to classify, mark, prioritize, preempt packets, and so on
are added to the configuration appropriately.

■ Monitoring and reporting—It generates system logging messages, SNMP traps, and
summary reports.

■ Consistency—The commands generated on different routers, using AutoQoS, are consistent
and interoperable.
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80 Chapter 2: IP Quality of Service

AutoQoS was introduced in Cisco IOS Software Release 12.2(15)T and provides a quick and
consistent way to enter the bulk of QoS commands. Network administrators can then modify those
commands and policies or optimize them using CLI. Cisco SDM QoS Wizard is a newer GUI tool
that generates QoS commands and policies; that tool will be discussed in the next section.

AutoQoS performs a series of functions on WAN devices and interfaces. It creates a traffic class
for voice payload (RTP), and it builds another class for voice signaling (Skinny, H.323, SIP, and
MGCP). Service policies for voice bearer and voice signaling are created and deployed using LLQ
with bandwidth guarantees. Voice traffic is assigned to the priority queue. On Frame Relay
connections, AutoQoS turns on Frame Relay traffic shaping (FRTS) and link fragmentation and
interleaving (LFI); on other types of links, such as PPP links, AutoQoS might turn on multilink
PPP (MLP) and compressed RTP (cRTP). AutoQoS also provides SNMP and syslog alerts for
VoIP packet drops.

In LAN environments, AutoQoS trust boundaries are set and enforced on the different types of
switch ports, such as access ports and uplinks. Expedited queuing (strict priority) and weighted
round-robin (WRR) are also enforced where required. Traffic is assigned to the proper queue
based on its marking or application recognition based on NBAR.

Using AutoQoS has some prerequisites. Before you enable AutoQoS on an interface, you must
ensure that the following tasks have been completed:

■ Cisco Express Forwarding (CEF) is enabled. CEF is the prerequisite for NBAR.

■ NBAR is enabled. AutoQoS for the Enterprise (not Auto QoS VoIP) uses NBAR for traffic
discovery and classification.

■ The correct bandwidth on the interface is configured. AutoQoS configures LLQ, cRTP, and
LFI based on the interface type and the interface bandwidth. On certain interfaces, such as
Ethernet, the bandwidth is auto-sensed; however, on other interfaces, such as synchronous
serial interface, if the bandwidth is not specified, the IOS assumes a bandwidth of 1544 Kbps.

After these tasks have been completed, AutoQoS can be configured (enabled) on the desired
interface. Example 2-4 shows a serial interface that has been configured with bandwidth, IP
address, CEF, and AutoQoS.

Example 2-4 Configuring AutoQoS on an Interface
ip cef
interface serial 1/0
bandwidth 256
ip address 10.1.1.1 255.255.255.252
auto qos voip
!
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QoS Implementation Methods 81

Note that in Example 2-4, the command auto qos voip is applied to interface serial 1/0. This
command represents the first generation of AutoQoS. The focus of auto qos voip was to automate
generation of QoS commands to get the device ready for VoIP traffic. In the second generation
AutoQoS for the Enterprise, you must first enter the auto discovery qos so that the router discovers
and analyzes network traffic entering the interface using NBAR. Next, you enter the auto qos
command. When you enter the auto qos command on an interface, the router builds class maps
(based on the results of discovery) and then creates and applies a policy map on the interface.
AutoQoS will be discussed in detail in Chapter 7, “Implementing AutoQoS.”

Router and Security Device Manager (SDM) QoS Wizard
Cisco SDM is a web-based device-management tool for Cisco routers. With SDM, router
deployment and troubleshooting of network and VPN connectivity issues becomes simpler.
Proactive management through performance monitoring is also accomplished using SDM.

Cisco SDM supports a range of Cisco IOS Software releases and is available on many Cisco router
models (from Cisco 830 Series to Cisco 7301); on several router models, SDM is preinstalled.
Cisco SDM offers smart wizards that provide step-by-step assistance for configuration of LAN
and WAN interfaces, Network Address Translation (NAT), firewall policy, IPS, IPsec VPN,
and QoS. Inexperienced users find the SDM GUI easier to use than the CLI and enjoy the
comprehensive online help and tutorials for SDM.

The QoS Wizard of SDM provides you with an easy-to-use user interface to define traffic classes
and configure QoS policies for your network. The SDM predefines three different application
categories: real-time, business-critical, and best-effort. SDM supports and uses NBAR to validate
the bandwidth consumed by different application categories. Additional features offered by the
SDM QoS Wizard include QoS policing and traffic monitoring. The SDM QoS Wizard enables
you to do three things:

■ Implement QoS

■ Monitor QoS

■ Troubleshoot QoS on your network

Figure 2-3 displays the main page of Cisco SDM. This page is comprised of two sections:

■ About Your Router

■ Configuration Overview
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82 Chapter 2: IP Quality of Service

Figure 2-3 Main Page of Cisco SDM

In the About Your Router section of the SDM main page you can find information about your
router’s hardware, software, and the available features. For example, you can see the router’s total
and available memory, flash capacity, IOS version, SDM version, and whether features such as IP,
firewall, VPN, IPS, and NAC are available. Further information can be seen through the More...
options in the hardware and software sections. The Configuration Overview section of the SDM
main page provides information about your router’s LAN and WAN interfaces, firewall policies,
VPN, routing, and IPS configurations. You can also see the router’s running configuration through
the View Running Config option. You can navigate to the main page by pressing the Home button
on the main tool bar of the Cisco SDM. The other two important buttons on the Cisco SDM main
tool bar are the Configure and Monitor buttons. The tasks available on the left side of the
Configure page are:

■ Interfaces and Connections

■ Firewall and ACL

■ VPN

■ Security Audit

■ Routing
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QoS Implementation Methods 83

■ NAT

■ Intrusion Prevention

■ Quality of Service

■ NAC

■ Additional Tasks

The tasks available on the left side of the Monitor page are:

■ Overview

■ Interface Status

■ Firewall Status

■ VPN Status

■ Traffic Status

■ NAC Status

■ Logging

■ IPS Status

If you select the Traffic Status task, you will have the option to view graphs about QoS or
application/protocol traffic.

The remainder of this section takes you through the steps necessary to create a QoS policy, apply
it to an interface, and monitor the QoS status using the Cisco SDM (GUI) Wizard. For each step
one or more figures are provided so that you are well prepared for the exam questions that might
be asked about creating QoS policy using the SDM Wizard.

To begin to create a QoS policy you must complete the following steps:

Step 1 Click the Configure button on the main toolbar of SDM.
Step 2 Click the Quality of Service button on the tasks toolbar on the left side of
the SDM window (in Configuration mode; see Figure 2-4).
Step 3 Click the Create QoS Policy tab in the middle section of the SDM window
(see Figure 2-4).
Step 4 Click the Launch QoS Wizard button on the bottom right side of the SDM
window (see Figure 2-4).
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84 Chapter 2: IP Quality of Service

Figure 2-4 Four Steps to Start Creating a QoS Policy with SDM

Now the SDM QoS Wizard page pops up on your computer screen (see Figure 2-5) and it informs
you that SDM by default creates QoS policy to handle two main types of traffic, namely Real-Time
and Business-Critical. To proceed press the Next button.

Figure 2-5 SDM QoS Wizard Initial Page
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QoS Implementation Methods 85

The QoS Wizard asks you to select an interface on which you want the QoS policy to be applied.
Figure 2-6 shows you this screen. After making your selection press the Next button on that screen
to proceed.

Figure 2-6 Interface Selection Page of SDM QoS Wizard

The SDM QoS Wizard asks you to enter the bandwidth percent for Real Time and Business-
Critical traffic (see Figure 2-7). SDM will then automatically compute the bandwidth percent for
the Best-Effort traffic and the actual bandwidth (kbps) for all three traffic classes.
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86 Chapter 2: IP Quality of Service

Figure 2-7 QoS Policy Generation Page of SDM QoS Wizard

After you press Next the new page shows a summary of the configuration applied to the interface
you have previously selected for the policy (see Figure 2-8). On this page you can scroll down and
up to see the policy generated (and to be applied) in its entirety. Once you press the Finish button.

After you press the Finish button on the SDM QoS summary of the configuration screen, a
Commands Delivery Status window appears (see Figure 2-9). This screen first informs you that
commands are being prepared, then it tells you that the commands are being submitted, and finally
it tells you that the commands have been delivered to the router. At this time, you can press the
OK button and the job is complete.
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QoS Implementation Methods 87

Figure 2-8 QoS Policy: Summary of the Configuration

Figure 2-9 QoS Policy: Commands Delivery Status

Upon completion of your QoS configuration tasks, SDM allows you to monitor the QoS status.
You must first click the Monitor button of the SDM main tool bar. Next, from the list of available
tasks you must select Traffic Status (see Figure 2-10). Note that in the ONT courseware, this
option is shown as QoS Status, probably due to SDM version differences. In the middle of the
Traffic Status screen, you will then notice a folder called Top N Traffic Flows with QoS and
Application/Protocol Traffic as two options displayed below it. If you click QoS (effectively
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88 Chapter 2: IP Quality of Service

requesting to see the QoS status), you can then choose any of the interfaces displayed in the Traffic
Status screen and see informative QoS-related graphs about the chosen interface.

Figure 2-10 SDM Monitor Traffic/QoS Status

When you select the QoS option of the Traffic Status, notice that on the top right corner of the
screen you can select the View Interval (Now, Every 1 Minute, Every 5 Minutes, Every 1
Hour). Furthermore, there is a small area with the “Select QoS Parameters for Monitoring”
title that allows you to select the Direction (input or output) of the traffic, and the Statistics
(bandwidth, byte, and packets dropped) for which you want to see graphs.
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Foundation Summary 89

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should help
solidify some key facts. If you are doing your final preparation before the exam, the information in
this section is a convenient way to review the day before the exam.

In a converged enterprise network, four major issues affect the performance and perceived quality
of applications:

■ Available bandwidth

■ End-to-end delay

■ Variation of delay (jitter)

■ Packet loss

Lack of sufficient bandwidth, high end-to-end delay, high variation in delay, and excessive packet
loss lower the quality of applications.

QoS is the ability of the network to provide better or “special” service to a set of users or app-
lications or both to the detriment of other users or applications or both. You can use several QoS
features, tools, and technologies to accomplish the QoS goals. Classification, marking, congestion
avoidance, congestion management, compression, shaping, and policing are examples of QoS
tools available in Cisco IOS. The three steps of implementing QoS in an enterprise network are as
follows:

Step 1 Identify the network traffic and its requirements
Step 2 Define traffic classes
Step 3 Define a QoS policy for each traffic class
The main QoS models of today are as follows:

■ Best-effort—The best-effort model requires no QoS configuration and mechanisms;
therefore, it is easy and scalable, but it provides no Differentiated Service to different
application types.

■ IntServ—IntServ provides guaranteed service (Hard QoS). It uses signaling to reserve and
guarantee resources for each traffic flow below it. RSVP is the common signaling protocol for
resource reservation signaling on IP networks. Per-flow signaling and monitoring escalate the
overhead of the IntServ model and make it nonscalable.
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90 Chapter 2: IP Quality of Service

■ DiffServ—DiffServ is the most modern of the three models. It requires traffic classification
and marking and providing differentiated service to each traffic class based on its marking.
DiffServ is scalable, but its drawback is that it requires implementation of complex QoS
features on network devices throughout the network.

Network administrators have four methods at their disposal to implement QoS on their network’s
Cisco devices:

■ Cisco IOS CLI—Configuring QoS using Cisco IOS CLI is the most complex and time-
consuming method. It requires that you learn different syntax for each QoS mechanism.
■ MQC—MQC is a modular command-line interface that is common across different Cisco
platforms, and it separates the task of defining different traffic classes from the task of
defining QoS policies.
■ Cisco AutoQoS—Because AutoQoS automatically generates QoS commands on your router
or switch, it is the simplest and fastest method among the four QoS implementation methods.
However, should you need to fine-tune the AutoQoS configuration results, you must use MQC
(or CLI) to do so. Fine-tuning of the commands that AutoQoS generates is seldom necessary.
■ Cisco Router and Security Device Manager (SDM) QoS Wizard—Cisco SDM offers
several wizards for implementing services, such as IPsec, VPN, and proactive management
through performance monitoring, in addition to the QoS Wizard. Cisco SDM QoS Wizard
allows you to remotely configure and monitor your Cisco routers without using the CLI. The
SDM GUI makes it simple for you to implement QoS services, features, and policies.
Table 2-3 compares Cisco IOS CLI, MQC, AutoQoS, and SDM with respect to how easy they are
to use, whether they allow you to fine-tune their results, how time consuming they are, and how
modular they are.

Table 2-3 Comparing QoS Implementation Methods

Method CLI MQC AutoQoS SDM
Ease of use Most difficult Easier than legacy CLI Simple Simple
Ability to fine-tune Yes (OK) Very well Limited Limited
Time consuming Most time Moderate time Least time Very little time
to implement consuming consumed (average) consuming consumed
(longest) (short)
Modularity Weakest Very modular Very Good
(poor) (excellent) modular
(excellent)

MQC is the recommended and the most powerful method for implementing QoS. It is modular, it
promotes re-use of written code, and it facilitates consistency of QoS configurations among your
Cisco devices. MQC also reduces the chances for errors and conflicts, while allowing you to take
advantage of the latest features and mechanisms offered by your version of Cisco IOS.
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Q&A 91

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. List the four key quality issues with converged networks.
2. Provide a definition for maximum available bandwidth and average available bandwidth per flow.
3. List at least three types of delay.
4. Provide at least three ways to reduce delay.
5. Provide at least two ways to reduce or prevent loss of important packets.
6. Provide a definition for QoS.
7. List the three key steps in implementing QoS on a network.
8. List the three main QoS models.
9. Provide a short description of the best-effort model.
10. What are the benefits and drawbacks of the best-effort model?
11. Provide a short description for the IntServ model.
12. Name the functions that the IntServ model requires on the network routers and switches.
13. What are the benefits and drawbacks of the IntServ model?
14. What are the main features of the DiffServ model?
15. What are the benefits and drawbacks of the DiffServ model?
16. What are the four QoS implementation methods?
17. Which of the four QoS implementation methods is nonmodular and the most time consuming?
18. What are the main benefits of MQC?
19. What is the most important advantage of AutoQoS?
20. What are the prerequisites for Auto QoS VoIP?
21. What are the prerequisites for Auto QoS for the enterprise?
22. Which of the four QoS implementation methods is the fastest?
23. What are the three main tasks that you can accomplish using the SDM QoS Wizard?
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This chapter covers the
following subjects:

■ Classification and Marking

■ The DiffServ Model, Differentiated
Services Code Point (DSCP), and Per-Hop
Behavior (PHB)

■ QoS Service Class

■ Trust Boundaries

■ Network Based Application Recognition
(NBAR)

■ Cisco IOS Commands to Configure NBAR
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CHAPTER 3
Classification,
Marking, and NBAR

Classification and marking are key IP QoS mechanisms used to implement the DiffServ QoS
model. This chapter defines classification and marking and explains the markings that are
available at the data link and network layers. This chapter also explains QoS service classes and
how to use them to create a service policy throughout a network. It also defines network trust
boundaries. Finally, it describes NBAR and PDLM and presents the IOS commands that are
required to configure NBAR.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 15-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 3-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics.

Table 3-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score

“Classification and Marking” 1–5

“The DiffServ Model, Differentiated Services Code Point 6–8
(DSCP), and Per-Hop Behavior (PHB)”

“QoS Service Class” 9

“Trust Boundaries” 10

“Network Based Application Recognition (NBAR)” 11–13

“Cisco IOS Commands to Configure NBAR” 14–15

Total Score (15 possible)
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94 Chapter 3: Classification, Marking, and NBAR

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this chapter.
If you do not know the answer to a question or are only partially sure of the answer, mark this
question wrong for purposes of the self-assessment. Giving yourself credit for an answer you
correctly guess skews your self-assessment results and might provide you with a false sense of
security.

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 9 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 10–12 overall score—Begin with the “Foundation Summary” section and then follow up
with the “Q&A” section at the end of the chapter.

■ 13 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following is not a valid classification traffic descriptor?
a. Incoming interface
b. Traffic path
c. IP precedence or DSCP value
d. Source or destination address
2. Which of the following is not considered a data link layer QoS marking field?
a. CoS
b. Frame Relay DE
c. DSCP
d. ATM CLP
3. Which of the following CoS values is reserved for internetwork and network control?
a. 0,1
b. 2,3
c. 4,5
d. 6,7
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“Do I Know This Already?” Quiz 95

4. Which of the following is the Frame Relay QoS marking field?
a. DE
b. CLP
c. CoS
d. EXP
5. Which of the following is true about the MPLS header and its EXP field size?
a. The MPLS header is 2 bytes and the EXP field is 3 bits long.
b. The MPLS header is 2 bytes and the EXP field is 6 bits long.
c. The MPLS header is 4 bytes and the EXP field is 6 bits long.
d. The MPLS header is 4 bytes and the EXP field is 3 bits long.
6. What is “an externally observable forwarding behavior of a network node toward a group of
IP packets that have the same DSCP value”?
a. BA
b. Prec
c. Service class
d. PHB
7. Which of the following is not a DSCP PHB?
a. Default PHB
b. Class selector PHB
c. Assured forwarding PHB
d. Cisco Express Forwarding PHB
8. Which of the following has the higher drop probability?
a. AF31.
b. AF32.
c. AF33.
d. They all have the same drop probability.
9. Which of the following is not a common voice and video service class?
a. Voice bearer (or payload)
b. Voice and video conferencing
c. Video payload
d. Voice and video signaling
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96 Chapter 3: Classification, Marking, and NBAR

10. At which of the following places is the trust boundary not implemented?
a. Core switch
b. Distribution switch
c. Access switch
d. End system
11. Which of the following is not a service that NBAR provides?
a. Protocol discovery
b. Collection of traffic statistics
c. Traffic classification
d. Traffic policing
12. Which of the following is true about loading a new PDLM?
a. You need to upgrade the IOS and reload your router.
b. You need to upgrade the IOS, but a reload is not necessary.
c. You do not need to upgrade the IOS, but a router reload is necessary.
d. You do not need to upgrade the IOS and do not need to reload either.
13. Which of the following is not an NBAR limitation?
a. NBAR can handle only up to 24 concurrent URLs.
b. NBAR analyzes only the first 400 bytes of the packet.
c. NBAR is not supported on interfaces in which tunneling or encryption is used.
d. NBAR is dependent on CEF.
14. Which of the following commands uses the NBAR classification feature within a class map?
a. match protocol protocol-name
b. match nbar protocol protocol-name
c. match protocol-name
d. match nbar protocol-name
15. What does the * character mean in a regular expression?
a. Match one of a choice of characters.
b. Match any zero or more characters in this position.
c. Match any one character in this position.
d. It means OR.
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Classification and Marking 97

Foundation Topics

Classification and Marking
With QoS, you intend to provide different treatments to different classes of network traffic.
Therefore, it is necessary to define traffic classes by identifying and grouping network traffic.
Classification does just that; it is the process or mechanism that identifies traffic and categorizes it
into classes. This categorization is done using traffic descriptors. Common traffic descriptors are
any of the following:

■ Ingress (or incoming) interface

■ CoS value on ISL or 802.1p frame

■ Source or destination IP address

■ IP precedence or DSCP value on the IP Packet header

■ MPLS EXP value on the MPLS header

■ Application type

In the past, you performed classification without marking. As a result, each QoS mechanism at
each device had to classify before it could provide unique treatments to each class of traffic. For
example, to perform priority queuing, you must classify the traffic using access lists so that you
can assign different traffic classes to various queues (high, medium, normal, or low). On the same
device or another, to perform queuing, shaping, policing, fragmentation, RTP header compression,
and so on, you must perform classification again so that different classes of traffic are treated
differently. Repeated classification in that fashion, using access-lists for example, is inefficient.
Today, after you perform the first-time classification, mark (or color) the packets. This way, the
following devices on the traffic path can provide differentiated service to packets based on packet
markings (colors): after the first-time classification is performed at the edge (which is mostly
based on deep packet inspection) and the packet is marked, only a simple and efficient
classification based on the packet marking is performed inside the network.

Classification has traditionally been done with access lists (standard or extended), but today the
Cisco IOS command class-map is the common classification tool. class-map is a component of
the Cisco IOS modular QoS command-line interface (MQC). The match statement within a class
map can refer to a traffic descriptor, an access list, or an NBAR protocol. NBAR is a classification
tool that will be discussed in this chapter. Please note that class-map does not eliminate usage of
other tools such as access lists. It simply makes the job of classification more sophisticated and
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98 Chapter 3: Classification, Marking, and NBAR

powerful. For example, you can define a traffic class based on multiple conditions, one of which
may be matching an access-list.

It is best to perform the initial classification (and marking) task as close to the source of traffic as
possible. The network edge device such as the IP phone, and the access layer switch would be the
preferable locations for traffic classification and marking.

Marking is the process of tagging or coloring traffic based on its category. Traffic is marked after
you classify it. What is marked depends on whether you want to mark the Layer 2 frame or cell or
the Layer 3 packet. Commonly used Layer 2 markers are CoS (on ISL or 802.1Q header), EXP
(on MPLS header, which is in between layers 2 and 3), DE (on Frame Relay header), and CLP (on
ATM cell header). Commonly used Layer 3 markers are IP precedence or DSCP (on IP header).

Layer 2 QoS: CoS on 802.1Q/P Ethernet Frame
The IEEE defined the 802.1Q frame for the purpose of implementing trunks between LAN
devices. The 4-byte 802.1Q header field that is inserted after the source MAC address on the
Ethernet header has a VLAN ID field for trunking purposes. A three-bit user priority field (PRI) is
available also and is called CoS (802.1p). CoS is used for QoS purposes; it can have one of eight
possible values, as shown in Table 3-2.

Table 3-2 CoS Bits and Their Corresponding Decimal Values and Definitions

CoS (bits) CoS (in Decimal) IETF RFC791 Application

000 0 Routine Best-Effort Data

001 1 Priority Medium Priority Data

010 2 Immediate High Priority Data

011 3 Flash Call Signaling

100 4 Flash-Override Video Conferencing

101 5 Critical Voice Bearer

110 6 Internet Reserved

(inter-network control)

111 7 Network Reserved

(network control)

Figure 3-1 shows the 4-byte 802.1Q field that is inserted into the Ethernet header after the source
MAC address. In a network with IP Telephony deployed, workstations connect to the IP phone
Ethernet jack (marked PC), and the IP phone connects to the access layer switch (marked Switch).
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Classification and Marking 99

The IP phone sends 802.1Q/P frames to the workgroup switch. The frames leaving the IP phone
toward the workgroup (access) switch have the voice VLAN number in the VLAN ID field, and
their priority (CoS) field is usually set to 5 (decimal), which is equal to 101 binary, interpreted as
critical or voice bearer.

Figure 3-1 802.1Q/P Field
Ethernet 802.1Q/P Frame

Preamble SFD DA SA 802.1Q/P Type Data FCS

TPID
PRI CFI VLAN ID
0×8100
3 bits 1 bit 12 bits
16 bits

CoS

Layer 2 QoS: DE and CLP on Frame Relay and ATM (Cells)
Frame Relay and ATM QoS standards were defined and used (by ITU-T and FRF) before Internet
Engineering Task Force (IETF) QoS standards were introduced and standardized. In Frame Relay,
for instance, the forward explicit congestion notification (FECN), backward explicit congestion
notification (BECN), and discard eligible (DE) fields in the frame header have been used to
perform congestion notification and drop preference notification. Neither Frame Relay frames nor
ATM cells have a field comparable to the 3-bit CoS field previously discussed on 802.1P frames.
A Frame Relay frame has a 1-bit DE, and an ATM cell has a 1-bit cell loss priority (CLP) field that
essentially informs the transit switches whether the data unit is not (DE or CLP equal 0) or whether
it is (DE or CLP equal 1) a good candidate for dropping, should the need for dropping arise. Figure
3-2 displays the position of the DE field in the Frame Relay frame header.

Figure 3-2 DE Field on Frame Relay Frame Header
Frame Relay Frame
Frame Relay
Flag Information FCS Flag
Header

DLCI C/R EA DLCI FECN BECN DE EA

Discard
Eligibility
(0 or 1)
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100 Chapter 3: Classification, Marking, and NBAR

Layer 2 1/2 QoS: MPLS EXP Field
MPLS packets are IP packets that have one or more 4-byte MPLS headers added. The IP packet
with its added MPLS header is encapsulated in a Layer 2 protocol data unit (PDU) such as
Ethernet before it is transmitted. Therefore, the MPLS header is often called the SHIM or layer
2 1/2 header. Figure 3-3 displays an MPLS-IP packet encapsulated in an Ethernet frame. The EXP
(experimental) field within the MPLS header is used for QoS purposes. The EXP field was
designed as a 3-bit field to be compatible with the 3-bit IP precedence field on the IP header and
the 3-bit PRI (CoS) field in the 802.1Q header.

Figure 3-3 EXP Field in the MPLS Header
MPLS Header

48 48 16 20 3 1 8 IP Packet
Bits Bits Bits Bits Bits Bit Bits
Type
DA SA Label Exp S TTL
×8847

Experimental
Field Used for
QoS Marking
Ethertype
0×8847
means
MPLS-IP-Unicast

By default, as an IP packet enters an MPLS network, the edge router copies the three most
significant bits of the type of service (ToS) byte of the IP header to the EXP field of the MPLS
header. The three most significant bits of the ToS byte on the IP header are called the IP precedence
bits. The ToS byte of the IP header is now called the DiffServ field; the six most significant bits of
the DiffServ field are called the DSCP.

Instead of allowing the EXP field of MPLS to be automatically copied from IP precedence, the
administrator of the MPLS edge router can configure the edge router to set the EXP to a desired
value. This way, the customer of an MPLS service provider can set the IP precedence or DSCP
field to a value he wants, and the MPLS provider can set the EXP value on the MPLS header to a
value that the service provider finds appropriate, without interfering with the customer IP header
values and settings.

The DiffServ Model, Differentiated Services Code Point (DSCP),
and Per-Hop Behavior (PHB)
The DiffServ model was briefly discussed in Chapter 2, “IP Quality of Service.” Within the
DiffServ architecture, traffic is preferred to be classified and marked as soon (as close to the
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The DiffServ Model, Differentiated Services Code Point (DSCP), and Per-Hop Behavior (PHB) 101

source) as possible. Marking of the IP packet was traditionally done on the three IP precedence
bits, but now, marking (setting) the six DSCP bits on the IP header is considered the standard
method of IP packet marking.

NOTE Some network devices cannot check or set Layer 3 header QoS fields (such as IP
precedence or DSCP). For example, simple Layer 2 wiring closet LAN switches can only check
and set the CoS (PRI) bits on the 802.1Q header.

Each of the different DSCP values—in other words, each of the different combinations of DSCP
bits—is expected to stimulate every network device along the traffic path to behave in a certain
way and to provide a particular QoS treatment to the traffic. Therefore, within the DiffServ
framework, you set the DSCP value on the IP packet header to select a per-hop behavior (PHB).
PHB is formally defined as an externally observable forwarding behavior of a network node
toward a group of IP packets that have the same DSCP value. The group of packets with a common
DSCP value (belonging to the same or different sources and applications), which receive similar
PHB from a DiffServ node, is called a behavior aggregate (BA). The PHB toward a packet,
including how it is scheduled, queued, policed, and so on, is based on the BA that the packet
belongs to and the implemented service level agreement (SLA) or policy.

Scalability is a main goal of the DiffServ model. Complex traffic classification is performed as
close to the source as possible. Traffic marking is performed subsequent to classification. If
marking is done by a device under control of the network administration, the marking is said to be
trusted. It is best if the complex classification task is not repeated, and the PHB of the transit
network devices will solely depend on the trusted traffic marking. This way, the DiffServ model
has a coarse level of classification, and the marking-based PHB is applied to traffic aggregates or
behavior aggregates (BAs), with no per-flow state in the core.

Application-generated signaling (IntServ style) is not part of the DiffServ framework, and this
boosts the scalability of the DiffServ model. Most applications do not have signaling and Resource
Reservation Protocol (RSVP) capabilities. The DiffServ model provides specific services and QoS
treatments to groups of packets with common DSCP values (BAs). These packets can, and in large
scale do, belong to multiple flows. The services and QoS treatments that are provided to traffic
aggregates based on their common DSCP values are a set of actions and guarantees such as queue
insertion policy, drop preference, and bandwidth guarantee. The DiffServ model provides
particular service classes to traffic aggregates by classifying and marking the traffic first, followed
by PHB toward the marked traffic within the network core.
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102 Chapter 3: Classification, Marking, and NBAR

IP Precedence and DSCP
The initial efforts on IP QoS were based on the specifications provided by RFC 791 (1981), which
had called the 3 most significant bits of the ToS byte on the IP header the IP precedence bits. The
3 IP precedence bits can have one of eight settings. The larger the IP precedence value, the more
important the packet and the higher the probability of timely forwarding. Figure 3-4 displays an
IP packet and focuses on the IP ToS byte, particularly on the IP precedence bits. The eight IP
precedence combinations and their corresponding decimal values, along with the name given to
each IP precedence value, are also displayed in Figure 3-4. The IP precedence values 6 and 7, called
Internetwork Control and Network Control, are reserved for control protocols and are not allowed to
be set by user applications; therefore, user applications have six IP precedence values available.

Figure 3-4 IP Header ToS Byte and IP Precedence Values
IP Header

Ver Length ToS Flags Checksum ...

8 Bits

3 Bits 4 Bits 1 Bit
IP Precedence IP Precedence IP Precedence
Decimal Binary Name
IP Precedence 0 000 Routine
1 001 Priority
2 010 Immediate
3 011 Flash
4 100 Flash-Override
5 101 Critical
6 110 Internetwork Control
7 111 Network Control

Redefining the ToS byte as the Differentiated Services (DiffServ) field, with the 6 most significant
bits called the DSCP, has provided much more flexibility and capability to the new IP QoS efforts.
The 2 least significant bits of the DiffServ field are used for flow control and are called explicit
congestion notification (ECN) bits. DSCP is backward compatible with IP Precedence (IPP),
providing the opportunity for gradual deployment of DSCP-based QoS in IP networks. The
current DSCP value definitions include four PHBs:

■ Class selector PHB—With the least significant 3 bits of the DSCP set to 000, the class
selector PHB provides backward compatibility with ToS-based IP Precedence. When DSCP-
compliant network devices receive IP packets from non-DSCP compliant network devices,
they can be configured only to process and interpret the IP precedence bits. When IP packets
are sent from DSCP-compliant devices to the non-DSCP-compliant devices, only the 3 most
significant bits of the DiffServ field (equivalent to IP precedence bits) are set; the rest of the
bits are set to 0.
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The DiffServ Model, Differentiated Services Code Point (DSCP), and Per-Hop Behavior (PHB) 103

■ Default PHB—With the 3 most significant bits of the DiffServ/DSCP field set to 000, the
Default PHB is used for best effort (BE) service. If the DSCP value of a packet is not mapped
to a PHB, it is consequently assigned to the default PHB.

■ Assured forwarding (AF) PHB—With the most significant 3 bits of the DSCP field set to
001, 010, 011, or 100 (these are also called AF1, AF2, AF3, and AF4), the AF PHB is used
for guaranteed bandwidth service.

■ Expedited forwarding (EF) PHB—With the most significant 3 bits of the DSCP field set to
101 (the whole DSCP field is set to 101110, decimal value of 46), the EF PHB provides low
delay service.

Figure 3-5 displays the DiffServ field and the DSCP settings for the class selector, default, AF, and
EF PHBs.

Figure 3-5 IP Header DS Field and DSCP PHBs
DS Field

6 DSCP Bits

0 ECN ECN

_ _ _ 0 0 0 Class Selector PHB
0 0 0 _ _ 0 Default PHB
0 0 1 _ _ 0
0 1 0 _ _ 0 Assured Forwarding
0 1 1 _ _ 0 (AF) PHB
1 0 0 _ _ 0
Expedited Forwarding
1 0 1 1 1 0
(EF) PHB

The EF PHB provides low delay service and should minimize jitter and loss. The bandwidth that
is dedicated to EF must be limited (capped) so that other traffic classes do not starve. The queue
that is dedicated to EF must be the highest priority queue so that the traffic assigned to it gets
through fast and does not experience significant delay and loss. This can only be achieved if the
volume of the traffic that is assigned to this queue keeps within its bandwidth limit/cap. Therefore,
successful deployment of EF PHB is ensured by utilizing other QoS techniques such as admission
control. You must remember three important facts about the EF PHB:

■ It imposes minimum delay.

■ It provides bandwidth guarantee.

■ During congestion, EF polices bandwidth.
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104 Chapter 3: Classification, Marking, and NBAR

Older applications (non-DSCP compliant) set the IP precedence bits to 101 (decimal 5, called Critical)
for delay-sensitive traffic such as voice. The most significant bits of the EF marking (101110) are 101,
making it backward compatible with the binary 101 IP precedence (Critical) setting.

The AF PHB as per the standards specifications provides four queues for four classes of traffic
(AFxy): AF1y, AF2y, AF3y, and AF4y. For each queue, a prespecified bandwidth is reserved. If
the amount of traffic on a particular queue exceeds the reserved bandwidth for that queue, the
queue builds up and eventually incurs packet drops. To avoid tail drop, congestion avoidance
techniques such as weighted random early detection (WRED) are deployed on each queue. Packet
drop is performed based on the marking difference of the packets. Within each AFxy class, y
specifies the drop preference (or probability) of the packet. Some packets are marked with
minimum probability/preference of being dropped, some with medium, and the rest with
maximum probability/preference of drop. The y part of AFxy is one of 2-bit binary numbers 01,
10, and 11; this is embedded in the DSCP field of these packets and specifies high, medium, and
low drop preference. Note that the bigger numbers here are not better, because they imply higher
drop preference. Therefore, two features are embedded in the AF PHB:

■ Four traffic classes (BAs) are assigned to four queues, each of which has a minimum reserved
bandwidth.

■ Each queue has congestion avoidance deployed to avoid tail drop and to have preferential drops.

Table 3-3 displays the four AF classes and the three drop preferences (probabilities) within each
class. Beside each AFxy within the table, its corresponding decimal and binary DSCP values are
also displayed for your reference.

Table 3-3 The AF DSCP Values

Drop Probability

Class Low Drop Medium Drop High Drop

Class 1 AF11 AF12 AF13

DSCP 10: (001010) DSCP 12: (001100) DSCP 14: (001110)

Class 2 AF21 AF22 AF23

DSCP 18: (010010) DSCP 20: (010100) DSCP 22: (010110)

Class 3 AF31 AF32 AF33

DSCP 26: (011010) DSCP 28: (011100) DSCP 30: (011110)

Class 4 AF41 AF42 AF43

DSCP 34: (100010) DSCP 36: (100100) DSCP 38: (100110)
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The DiffServ Model, Differentiated Services Code Point (DSCP), and Per-Hop Behavior (PHB) 105

You must remember a few important facts about AF:

■ The AF model has four classes: AF1, AF2, AF3, and AF4; they have no advantage over each
other. Different bandwidth reservations can be made for each queue; any queue can have more
or less bandwidth reserved than the others.

■ On a DSCP-compliant node, the second digit (y) of the AF PHB specifies a drop preference
or probability. When congestion avoidance is applied to an AF queue, packets with AFx3
marking have a higher probability of being dropped than packets with AFx2 marking, and
AFx2 marked packets have a higher chance of being dropped than packets with AFx1
marking, as the queue size grows.

■ You can find the corresponding DSCP value of each AFxy in decimal using this formula:

DSCP (Decimal) = 8x + 2y.
For example, the DSCP value for AF31 is 26 = (8 * 3) + (2 * 1).
■ Each AFx class is backward compatible with a single IP precedence value x. AF1y maps to
IP precedence 1, AF2y maps to IP precedence 2, AF3y maps to IP precedence 3, and AF4y
maps to IP precedence 4.

■ During implementation, you must reserve enough bandwidth for each AF queue to avoid
delay and drop in each queue. You can deploy some form of policing or admission control so
that too much traffic that maps to each AF class does not enter the network or node. The exact
congestion avoidance (and its parameters) that is applied to each AF queue is also dependent
on the configuration choices.

■ If there is available bandwidth and an AF queue is not policed, it can consume more
bandwidth than the amount reserved.

Most of the fields within the IP packet header in a transmission do not change from source to
destination. (However, TTL, checksum, and sometimes the fragment-related fields do change.)
The Layer 3 QoS marking on the packet can be preserved, but the Layer 2 QoS marking must be
rewritten at every Layer 3 router because the Layer 3 router is responsible for rewriting the Layer
2 frame. The packet marking is used as a classification mechanism on each ingress interface of a
subsequent device. The BA of the service class that the traffic maps to must be committed. To
guarantee end-to-end QoS, every node in the transmission path must be QoS capable. QoS
differentiated service in MPLS networks is provided based on the EXP bits on the MPLS header.
As a result, it is important that at certain points in the network, such as at edge devices, mapping
is performed between IP precedence, DSCP, CoS, MPLS, or other fields that hold QoS markings.
The mapping between 802.1Q/P CoS, MPLS EXP, and IP precedence is straightforward because
all of them are based on the old-fashioned 3-bit specifications of the 1980s. Mapping the DSCP
PHBs to those 3-bit fields requires some administrative decisions and compromises.
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106 Chapter 3: Classification, Marking, and NBAR

QoS Service Class
Planning and implementing QoS policies entails three main steps:

Step 1 Identify network traffic and its requirements.
Step 2 Divide the identified traffic into classes.
Step 3 Define QoS policies for each class.
In Step 1, you use tools such as NBAR to identify the existing traffic in the network. You might
discover many different traffic types. In Step 1, you must then recognize and document the
relevance and importance of each recognized traffic type to your business.

In Step 2, you group the network traffic into traffic or service classes. Each traffic or service class,
composed of one or more traffic types, receives a specific QoS treatment. Each service class is
created for one or more traffic types (a single group) that is called a BA. A common model used
by service providers, called the customer model, defines four service classes:

■ Mission critical

■ Transactional

■ Best-effort

■ Scavenger

A traffic class can be defined based on many factors. For example, these criteria, should they be
appropriate, can also be used to define traffic classes: an organization or department, a customer
(or a set of them), an application (or a group of applications, such as Telnet, FTP, SAP, Oracle), a
user or group of users (by location, job description, workstation MAC address), a traffic
destination, and so on.

Step 3 in planning and implementing QoS policies using QoS service classes is defining policies
for each service class. This step requires an understanding of the QoS needs of the traffic and
applications that are within your network. When you design the policies, be careful not to make
too many classes and make the matter too complex and over-provisioned. Limiting the service
classes to four or five is common. Also, do not assign too many applications and traffic to the high-
priority and mission-critical classes, because assigning a large percentage of traffic to those classes
will ultimately have a negative effect. Some of the existing common traffic classes are as follows:

■ Voice applications (VoIP)

■ Mission-critical applications, such as Oracle and SAP

■ Transactional/Interactive applications, such as Telnet and SSH
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QoS Service Class 107

■ Bulk applications such as FTP and TFTP

■ Best-effort applications, such as WWW and e-mail

■ Scavenger applications, such as Napster and Kazaa

You can find many sources of information and recommendations on QoS design and implementation;
however, each network is unique and requires special attention. It is important to implement the
QoS policies throughout the network and in a consistent way. Keep in mind the following two
important points:

■ If you do not implement QoS policies in certain parts of the network, the QoS offering of your
network will be incomplete, unpredictable, and inadequate.

■ Because not all network devices have consistent and complete capabilities and features, you
must map QoS techniques and features well. That way, the behavior of the diverse devices
within your network will be consistent and in-line with your policies.

One required task during the QoS policy implementation stage is mapping and translating between
CoS, DSCP, IP precedence, and MPLS EXP markings. Table 3-4 shows the Cisco recommended
mappings between Layer 2 CoS, IP precedence, DSCP, PHB and Class Selector Name, and their
corresponding traffic types.

Table 3-4 Mapping Different Markings to Different Traffic Types

Cisco AutoQoS Layer 2 CoS or IP DSCP Value DSCP Value
Class Precedence in Decimal in Binary Code Name

Best Effort 0 0 000000 BE

(Best Effort)

Scavenger 1 8 001000 CS1

(Class Selector 1)

Bulk Data 1 10 001010 AF11

12 001100 AF12

14 001110 AF13

Network 2 16 010000 CS2
Management
(Class Selector 2)
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108 Chapter 3: Classification, Marking, and NBAR

Table 3-4 Mapping Different Markings to Different Traffic Types (Continued)

Cisco AutoQoS Layer 2 CoS or IP DSCP Value DSCP Value
Class Precedence in Decimal in Binary Code Name

Telephony 3 26 011010 AF31
Signaling

Local Mission 3 28 011100 AF32
Critical
30 011110 AF33

Streaming Media 4 32 100000 CS4
Traffic
(Class Selector 4)

Interactive Video 4 34 100010 AF41
Traffic
36 100100 AF42

38 100110 AF43

Interactive Voice 5 46 101110 EF
Bearer Traffic

Trust Boundaries
End-system devices such as personal computers, IP phones, IP conference devices, and video
conference gateways, plus switches and routers at different levels of the network hierarchy, can
mark the IP packets or the encapsulating frames such as 802.1Q/P. One of the design and policy
decisions you have to make is where to place your network trust boundary. The trust boundary
forms a perimeter on your network; your network respects and trusts (does not override) the
markings that the devices on or inside this perimeter (trust boundary) make. Markings that devices
make outside the trust boundary are often reset, or at least checked and modified if necessary. The
devices that check and reset the markings of the traffic received from the untrusted devices
(devices outside the trust boundary), form the trust boundary of the network. The devices that form
the trust boundary are the first set of devices that are trusted because they forward traffic toward
the network core. It is considered good practice to place the trust boundary as close to the traffic
source (and away from the network core) as possible.

You should certainly try to place the trust boundary as close to the network edge as possible.
However, two other factors can affect your decision. First, the trusted device must be under your
administration and control; at the very least, you should be confident that its marking is in-line
with your QoS policies. Second, different devices have different capabilities and feature sets with
respect to the ability to check and set/reset various QoS markings such as CoS and DSCP. With all
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Trust Boundaries 109

facts considered, the trust boundary is implemented at one of the following network hierarchy
layers:

■ End system

■ Access switch

■ Distribution switch

Figure 3-6 depicts three scenarios with the trust boundary placed on the IP phone, the access
switch, and the distribution switch. The end systems, except for telephony and conference
systems, are generally recommended not to be trusted. New microcomputer operating systems
such as the Linux and Microsoft operating systems make it possible to set the DSCP or CoS field
on the transmitted traffic. Access switches, if they have the capability, are generally configured to
(or by default do) trust the markings set by the IP phone only. If the access switch does not have
any or enough QoS capabilities, you might have to shift the trust boundary to the distribution layer
switch.

Figure 3-6 Trust Boundary Placement Choices
Trust Boundary Access
Distribution
PC 1 Switch
Switch
IP Network
Access 802.1Q/p Trunk Core
Connection

Trust Boundary
PC 2

Network
Access Trunk Core
Connection

Trust Boundary
PC 3

Network
Access Trunk Core
Connection

In the first scenario displayed in Figure 3-6, the trust boundary is placed on the Cisco IP phone.
The phone sets/resets the CoS field to 0 (000 binary) for the frames it receives from the PC as it
forwards them to the switch. The CoS value on the IP phone-generated frames that are carrying
voice signaling is set to 3 (011 binary), and it is set to 5 (101 binary) for those that are carrying
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110 Chapter 3: Classification, Marking, and NBAR

voice. The access switch is configured to trust the markings of the traffic received on the port that
the Cisco IP phone is connected to. But how does the switch know that a Cisco IP phone, and not
another IP device such as a PC, is connected to that port? The switch discovers that a Cisco IP
phone is connected to its port by means of the Cisco Discovery Protocol version 2 (CDP v2) that
both the switch and the IP phone are supposed to have enabled. If the switch does not discover an
IP phone, it does not extend the trust boundary to the end device and dynamically shifts the trust
boundary to itself (the access switch).

In the second scenario, the PC is connected to the access switch, the trusted device. The access
switch must be configured to check (and reset if necessary) the CoS field in case it receives
802.1Q/P frames from the PC (rare case). Some access switches are capable of checking (and
setting) the IP header QoS fields (ToS field’s IP precedence or DSCP). When the traffic from the
PC is forwarded toward the distribution switch, because the connection between the access switch
and distribution switch is usually an 802.1Q/P trunk, the access switch can set the CoS field (and
the DSCP field, if the switch has the capability) of the outgoing traffic to certain values based on
QoS policies and the traffic type. For instance, the PC can run several different applications,
including Cisco IP Communicator. In that case, if the marking of the traffic coming from the PC
is not trusted, classification and marking of the traffic must happen on the trusted access switch.
Network QoS treatments and PHBs are based on the markings that happen at the trusted boundary.

The third scenario in Figure 3-6 shows the trust boundary placed on the distribution switch. This
usually happens when the access switch does not have enough or complete QoS classification,
policing, or marking capabilities. It is also possible that the access switch is not under your
administrative control; this is quite common in data center environments. For instance, the access
switch might be able to set or reset the CoS field of the 802.1Q/P header but might not be able to
set or reset the DSCP field on the IP packet header. The distribution switch has QoS capabilities
and features so that it can do classification, policing, and marking based on CoS or DSCP (or IP
precedence).

Network Based Application Recognition (NBAR)
NBAR is a Cisco IOS feature that can be used to perform three tasks:

■ Protocol discovery

■ Traffic statistics collection

■ Traffic classification

Because NBAR can discover which applications and protocols are running on your network and
display volume and statistics about them, you can use it as a powerful yet simple tool to form the
definitions of your network traffic classes (BAs). You can also use NBAR within class-based (CB)
marking or other MQC-based tools to classify packets for purposes such as marking, policing, and
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Network Based Application Recognition (NBAR) 111

queuing. NBAR is a powerful protocol discovery and classification tool, but the overhead it
imposes is considered small or medium. The amount of CPU utilization increase that a router
running NBAR experiences depends on the amount of traffic and the router CPU type and speed.

NBAR recognizes a limited number of protocols. However, you can expand the list of recognized
protocols by loading new Packet Description Language Modules (PDLMs), published by Cisco
systems, into your device (flash memory) and making a reference to the new PDLM in the device
configuration. PDLMs are files that Cisco Systems publishes; these files contain rules that NBAR
uses to recognize protocols and applications. A new PDLM can be loaded in the flash memory of
the Cisco device and then referenced within its configuration without a need to perform an IOS
upgrade or reload the device. Cisco Systems makes up-to-date PDLMs available to registered
users on Cisco Connection Online (CCO) at www.cisco.com/cgi-bin/tablebuild.pl/pdlm.

Before you can design a classification and marking scheme for your network, you need to identify
and recognize the existing traffic for your network. The NBAR protocol-discovery feature
provides a simple way to discover and report the applications and protocols that transit (in and out)
a particular interface of a network device you choose. Protocol discovery discovers and reports on
the protocols and applications that NBAR supports (plus those added by the loaded PDLMs). Key
statistics are also reported on the discovered protocols and applications. Examples of the statistics
that NBAR protocol discovery reports on each protocol are the total number of input and output
packets and bytes and the input and output bit rates. The list of discovered protocols and
applications, plus the associated statistics, which NBAR reports, are valuable when you want to
define your traffic classes and their QoS policies.

NBAR can classify traffic by inspecting bytes beyond the network and transport layer headers.
This is called subport classification. This means that NBAR looks into the segment (TCP or UDP)
payload and classifies based on that content. For example, NBAR can classify HTTP traffic based
on the URL; it can also classify based on MIME type.

NBAR has some limitations. First, it does not function on the Fast EtherChannel logical interface.
Second, NBAR can only handle up to 24 concurrent URLs, hosts, or MIME types. Third, NBAR
only analyzes the first 400 bytes of the packet. Fourth, it only supports CEF and does not work if
another switching mode is used. It does not support multicast packets, fragmented packets, and
packets that are associated with secure HTTP (URL, host, or MIME classification). NBAR does
not analyze or recognize the traffic that is destined to or emanated from the router where NBAR
is running.

Configuring classification without NBAR is mostly dependent on writing and maintaining access
lists. Using NBAR for classification is not only simpler than using access lists, but NBAR also
offers capabilities beyond those offered by access lists. NBAR can do stateful inspection of flows.
This means that it can discover the dynamic TCP or UDP port numbers that are negotiated at
connection establishment time by inspecting the control session packets. For example, a TFTP
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112 Chapter 3: Classification, Marking, and NBAR

session is initiated using the well-known UDP port 69, but the two ends of the session negotiate
other ports for the remainder of the session traffic. NBAR also supports some non-IP and non-
TCP/non-UDP protocols and applications such as Internetwork Packet Exchange (IPX), IPsec,
and GRE. Finally, as stated already, NBAR is able to discover and classify by deep packet
inspection, too. This means that NBAR can inspect the payload of TCP and UDP segments (up to
the 400th byte of the packet) and classify. HTTP sessions can be classified by URL, hostname, or
MIME type.

Cisco IOS Commands to Configure NBAR
To enhance the list of protocols that NBAR recognizes through a PDLM, download the PDLM
from CCO and copy it into the flash or on a TFTP server. Next, enter the following command,
which refers to the PDLM name in URL format:

Router(config)# ip nbar pdlm pdlm-name

The URL, for example, can be flash://citrix.pdlm, referring to the citrix.pdlm file in flash memory.
The URL can also refer to a file on a TFTP server, such as tftp://192.168.19.66/citrix.pdlm.

To modify the port number that NBAR associates to a protocol name or to add a port to the list of
ports associated to a protocol name, use this command:

Router(config)# ip nbar port-map protocol-name [t
tcp | udp] port-number

The preceding command configures NBAR to search for a protocol or protocol name using a port
number other than the well-known one. You can specify up to 16 additional port numbers.

To see the current NBAR protocol-to-port mapping, use the following show command:

Router# show ip nbar port-map [protocol-name]

Example 3-1 displays partial sample output of the preceding command.

Example 3-1 Displaying NBAR Protocol-to-Port Mapping
Router# show ip nbar portmap

port-map bgp tcp 179
port-map dhcp udp 67 68
port-map dns udp 53
port-map dns tcp 53
...

To enable NBAR protocol discovery on a router interface, first ensure that CEF is enabled on that
interface. CEF is turned on using the IP CEF command from Cisco IOS global configuration
mode. Next, enter the following command in the interface configuration mode:

Router(config-if)# ip nbar protocol-discovery
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Cisco IOS Commands to Configure NBAR 113

To display the discovered protocols and the statistics gathered for each discovered protocol, enter
the following show command. Note that unless you specify an interface, the output will include
the statistics gathered for all interfaces (back to back):

Router# show ip nbar protocol-discovery

Sample output of the preceding command is shown in Example 3-2.

Example 3-2 Displaying NBAR protocol-discovery Results
Router# show ip nbar protocol-discovery
Ethernet 0/0/0
Input Output
Protocol Packet Count Packet Count
Byte Count Byte Count
5 minute bit rate (bps) 5 minute bit rate (bps)
---------------- ------------------------ -----------------------------
eigrp 60 0
3600 0
0 0
bgp 0 0
0 0
0 0
...

You can use NBAR to recognize and classify protocols that use static port numbers; NBAR can
do the same for protocols that dynamically negotiate port numbers. If you want NBAR to classify
network traffic based on protocol and subsequently apply certain QoS policies to each traffic class,
use MQC class map and refer to the desired NBAR protocol with a match statement. The
following is the syntax for the match statement within a class map:

Router(config-cmap)# match protocol protocol-name

The protocol-name that is referred by the class map match protocol statement is an NBAR-
supported protocol such as ip, arp, compressed tcp, cdp, dlsw, ipx, and so on. Do not forget that
you can specify additional ports (besides the well-known ports) for each protocol by configuring
the previously introduced ip nbar port-map command. Also, to expand the list of NBAR-
supported protocols, you can load new PDLMs in your device, as discussed earlier in this section.
To use NBAR for classification and marking of traffic belonging to static-port protocols and to
apply the policy to an interface, you have to perform the following tasks:

■ Enable NBAR protocol discovery.

■ Configure a traffic class using the MQC class map.

■ Configure a QOS policy using the MQC policy map.
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114 Chapter 3: Classification, Marking, and NBAR

■ Apply the policy to the interface(s).

■ Expand the NBAR protocol ports or PDLM protocols if needed.

Example 3-3 shows partial configuration of a router with a policy called www-ltd-bw (implying
limited bandwidth for web browsing or HTTP protocol) applied to its serial 1/1 interface. The first
line shows that TCP ports 80 and 8080 are defined for HTTP. The configured class map defines a
traffic class called www, which includes all traffic classified by NBAR as http. The policy map
called www-ltd-bw is applied to the outgoing traffic of the serial 1/1 interface using the service-
policy output command. The policy map www-ltd-bw specifies that the traffic classified as www
is assigned to a queue with a 512-Kbps bandwidth reservation.

Example 3-3 Implementing QoS Policy Using NBAR for Static Protocols
ip nbar port-map http tcp 80 8080
!
class-map www
match protocol http
!
policy-map www-ltd-bw
class www
bandwidth 512
!
interface serial 1/1
ip nbar protocol-discovery
service-policy output www-ltd-bw
!

In Example 3-3, the command ip nbar protocol-discovery is applied to the serial 1/1 interface.
In the past (earlier Cisco IOS releases), you had to apply this command to the interface before you
could apply a service policy that used NBAR (through the match protocol name command);
however, as of Cisco IOS 12.2T, this is no longer necessary. The ONT course does not mention
this fact in its initial release, so for examination purposes, you might want to do it the old-
fashioned way and apply the ip nbar protocol-discovery command to the interface.

You can also use NBAR to do traffic classification for stateful protocols, those that negotiate the
data session port numbers during the initial control session. You still need to take three steps:

1. Configure a traffic class using MQC class map.
(Within the class map, the match statement references the stateful protocol such as TFTP).
2. Configure a QOS policy using MQC policy map.
3. Apply the policy to the interface(s).
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Cisco IOS Commands to Configure NBAR 115

One of the most attractive and powerful NBAR features is its ability to do deep packet inspection.
Four popular uses of NBAR deep packet inspection are as follows:

■ Classifying traffic based on the hostname or the URL after the hostname in the HTTP GET
requests

■ Classifying traffic based on the MIME type

■ Classifying traffic belonging to fast-track protocols file transfers using regular expressions
that match strings

■ Classifying traffic based on the RTP payload type or CODEC

The match protocol commands required within MQC class map, to classify traffic according to
the preceding criteria, are as follows:

Router(config-cmap)# match protocol http url url-string
Router(config-cmap)# match protocol http host host-name
Router(config-cmap)# match protocol http mime mime-type
Router(config-cmap)# match protocol fasttrack file-transfer regular-expression
Router(config-cmap)# match protocol rtp [a
audio | video |
payload-type payload-type-string]

Example 3-4 shows three class maps: from-cisco, whats-up, and cool-jpegs. The class map from-
cisco matches any HTTP GET request from hosts whose names begin with cisco. cisco* is a
regular expression that matches any string that begins with characters cisco (followed by zero or
more characters). Special characters such as *, which means zero or more characters (wildcard),
make writing regular expressions a lot easier. The class map whats-up matches HTTP packets
based on any URL containing the string /latest/whatsnew followed by zero or more characters. The
last class map in Example 3-4, cool-jpegs, classifies packets based on the Joint Photographics
Expert Group (JPEG) MIME type.

Example 3-4 Using NBAR to Match HTTP Hostname, URL, and MIME Type
!
class-map from-cisco
match protocol http host cisco*
!
class-map whats-up
match protocol http url /latest/whatsnew*
!
class-map cool-jpegs
match protocol http mime “*jpeg”
!
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116 Chapter 3: Classification, Marking, and NBAR

For your reference only (not for the purpose of exam preparation), Table 3-5 presents a few useful
special characters you can use within regular expressions of the class map match statement.

Table 3-5 Special Strings and Characters for Regular Expressions

Character or String Description

* Match zero or more characters in this position.

? Match any one character in this position.

It means OR. Match one of a choice of characters on either side of the |
| symbol.

Match one of a choice of characters inside the parentheses on either side of
(|) the | symbol. For example, xyz.(gif|jpg) matches either xyz.gif or xyz.jpg.

Match any character in the range specified, or one of the special characters.
For example, [0-9] is any single digit; [*] matches the * character, and [[]
[] matches the [ character.

You can also use NBAR deep packet inspection to match traffic from FastTrack peer-to-peer
protocols such as Kazaa and Grokster. To configure NBAR to match FastTrack peer-to-peer traffic,
use the following command in class map configuration mode:

Router(config-cmap)# match protocol fasttrack file-transfer reg-exp

Please note that the preceding command syntax expects a regular expression to identify a specific
FastTrack traffic. Gnutella traffic can be classified similarly using NBAR, by changing the
keyword FastTrack to Gnutella.

Example 3-5 shows three class maps. The class map called fasttrack1 configures NBAR to match
all FastTrack traffic. In the second class map, all FastTrack files that have the .mpeg extension are
classified into traffic class fasttrack2. Class map fasttrack3 specifies that all FastTrack traffic that
contains the string “cisco” is part of the traffic class called fasttrack3.

Example 3-5 Using NBAR to Match FastTrack Protocol Traffic
!
class-map fasttrack1
match protocol fasttrack file-transfer “*”
!
class-map fasttrack2
match protocol fasttrack file-transfer “*.mpeg”
!
class-map fasttrack3
match protocol fasttrack file-transfer “*cisco*”
!
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Cisco IOS Commands to Configure NBAR 117

The Real-Time Transport Protocol (RTP) is considered the transport protocol of choice for real-
time audio and video. It adds a header above the UDP header to include information such as
reconstruction timestamp and sequence number, plus security and content identification. RTP has
a control protocol sister called Real-Time Protocol Control Protocol (RTCP). Whereas RTP uses
the UDP even-numbered ports (starting with 16384 by default), RTCP uses the UDP odd-number
ports. NBAR deep packet inspection allows you to do classification based on RTP payload type
(audio or video) or do a deeper classification based on audio or video CODEC type. The syntax to
configure NBAR to match RTP traffic in class map configuration mode is as follows:

Router(config-cmap)# match protocol rtp [a
audio | video |
payload-type payload-type-string]

In the preceding command syntax, the optional keyword audio specifies matching by audio
payload type. (Values in the range of 0 to 23 are reserved for audio traffic.) Similarly, the optional
keyword video specifies matching by video payload type. (Values in the range of 24 to 33 are
reserved for video traffic.) If you use the optional keyword payload-type, you can specify (using
a string) matching by a specific payload type value, providing more granularity than is available
with the audio or video keywords. A payload string argument can contain commas to separate
payload type values and hyphens to indicate a range of payload type values.

Example 3-6 shows two class maps. The first class map is called voice, and as the name implies,
it matches the RTP audio protocol. The class map called video matches the RTP video protocol.

Example 3-6 Using NBAR to Match RTP Protocol Traffic
!
class-map voice
match protocol RTP audio
!
class-map video
match protocol RTP video
!
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118 Chapter 3: Classification, Marking, and NBAR

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Table 3-6 summarizes the major topics in this chapter.

Table 3-6 Summary of Classification, Marking, and NBAR

Topic Summary

Purpose of packet Packet classification is a QoS mechanism that distinguishes and
classification divides network traffic into traffic classes or behavior aggregates
(BAs).

Purpose of packet marking Packets, frames, and some other protocol data units (PDUs) have a
special field designed for QoS purposes. Marking is a QoS
mechanism that sets this field to a common value on packets that
belong to the same traffic/service class (BA) and sets them to
different values on packets that belong to different classes.

Classification and marking at Different data link layer protocol data units (PDUs) have different
the data link layer fields for QoS classification and marking purposes. On 802.1Q/P or
ISL frames, the 3-bit PRI (CoS) field is used for that purpose. On
Frame Relay frames, the DE bit is used for that purpose, and on
AMT cells, the CLP bit is used. On the MPLS header (layer 2∫) the
3-bit EXP field is used for QoS purposes.

PHB A per-hop behavior (PHB) is an externally observable forwarding
behavior applied at a DiffServ-compliant node to a DiffServ BA.

Class selector PHB (DSCP) The class-selector PHB is a set of DSCP values that make DSCP
backward compatible with IPP (IP precedence). The least significant
bits of the class selectors (CS1 through CS7) are 000.

AF PHB The assured forwarding (AF) PHB provides four queues for four
classes of traffic. Bandwidth reservation can be made for each AF
queue. Each AF has three DSCP values associated to it so that
differentiated drop policy can be applied to the packets in the same
AF queue.

EF PHB The expedited forwarding (EF) PHB provides a priority queue with
guaranteed but policed bandwidth. EF PHB is ideal for delay-
sensitive traffic as long as this type of traffic is not oversubscribed.
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Foundation Summary 119

Table 3-6 Summary of Classification, Marking, and NBAR (Continued)

Topic Summary

QoS service class QoS service class is a logical grouping of packets that, as per the
administrative policy definitions, are required to receive the same
QoS treatment.

Trust boundary Marking is recommended to take place as close to the ingress edge
of the network as possible. Marking, however, must be done by a
trusted device. The ingress edge/perimeter of the network where the
trusted devices reside and perform marking is called the trust
boundary.

NBAR NBAR is a protocol discovery and a classification tool/feature.
Within a class map, you can configure a match statement that refers
to an NBAR protocol.

NBAR Protocol Discovery To discover the network traffic mix that transits through an interface
(both input and output), apply the NBAR protocol discovery feature
to that interface. NBAR protocol discovery also reports traffic
statistics such as total number of input/output packets and bytes and
input/output bit rates.

NBAR PDLMs The NBAR Packet Description Language Modules (PDLM) are files
provided by Cisco Systems that you can load into your network
device to extend the NBAR list of supported protocols or enhance
the NBAR existing protocol-recognition capability. Loading a new
PDLM does not require a router reload.

NBAR application support NBAR can discover and classify both types of applications: those
that use static ports and those that use dynamically assigned ports.
NBAR can do classification through deep packet inspection; for
example, it can classify based on URL, MIME type, and RTP
payload type. CEF must be enabled on device interfaces for NBAR
to function.
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120 Chapter 3: Classification, Marking, and NBAR

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. Define and explain classification.
2. Define and explain marking.
3. What is the marker field on the 802.1Q/P frame called?
4. What are the names and definitions for CoS values 0 through 7?
5. Which one of the DSCP PHBs provides backward compatibility with ToS-based IP
precedence?
6. What are the four DiffServ (DSCP) PHBs?
7. How is compatibility between MPLS and network layer QoS achieved?
8. What is a QoS service class?
9. What is a trust boundary?
10. What is NBAR?
11. Name at least three limitations of NBAR.
12. List application support for NBAR.
13. What is PDLM?
14. What types of RTP payload classification does NBAR offer?
15. Which match command within a class map allows you to identify FastTrack peer-to-peer
protocols?
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This chapter covers the
following subjects:

■ Introduction to Congestion Management
and Queuing”

■ First-In-First-Out, Priority Queuing,
Round-Robin, and Weighted Round-
Robin Queuing”

■ Weighted Fair Queuing

■ Class-Based Weighted Fair Queuing

■ Low-Latency Queuing
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CHAPTER 4
Congestion Management
and Queuing

This chapter starts by defining what congestion is and why it happens. Next, it explains the need
for queuing or congestion management and describes the router queuing components. The rest
of this chapter is dedicated to explaining and providing configuration and monitoring
commands for queuing methods, namely FIFO, PQ, RR, WRR, WFQ, CBWFQ, and LLQ.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 13-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 4-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 4-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“Introduction to Congestion Management and Queuing” 1–4
“First-In-First-Out, Priority Queuing, Round-Robin, and 5–7
Weighted Round-Robin Queuing”
“Weighted Fair Queuing” 8–11
“Class-Based Weighted Fair Queuing” 12
“Low-Latency Queuing” 13
Total Score (13 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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124 Chapter 4: Congestion Management and Queuing

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 9 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 10–11 overall score—Begin with the “Foundation Summary” section and then follow up
with the “Q&A” section at the end of the chapter.

■ 12 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following is not a common reason for congestion?
a. Speed mismatch
b. Aggregation
c. Confluence
d. Queuing
2. Which of the following is a congestion management tool?
a. Aggregation
b. Confluence
c. Queuing
d. Fast Reroute
3. Which of the following is not a function within a queuing system?
a. Creating one or more queues
b. CEF
c. Assigning arriving packets to queues
d. Scheduling departure of packets from queues
4. How many queuing subsystems exist in an interface queuing system?
a. One
b. Two: a software queue and a hardware queue
c. Three: a software, a transmit, and a hardware queue
d. Four: a software, a hold, a transmit, and a hardware queue
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“Do I Know This Already?” Quiz 125

5. What is the default queuing discipline on all but slow serial interfaces?
a. FIFO
b. WFQ
c. CQ
d. WRR
6. How many queues does PQ have?
a. One
b. Two: High and Low
c. Three: High, Medium, and Low
d. Four: High, Medium, Normal, and Low
7. Custom queuing is a modified version of which queuing discipline?
a. WFQ
b. PQ
c. FIFO
d. WRR
8. Which of the following is not a goal or objective of WFQ?
a. Provide high bandwidth to high-volume traffic
b. Divide traffic into flows
c. Provide fair bandwidth allocation to the active flows
d. Provide faster scheduling to low-volume interactive flows
9. Which of the following is not used to recognize and differentiate flows in WFQ?
a. Source and destination IP address
b. Packet size
c. Source and destination TCP/UDP port number
d. Protocol number and type of service
10. Which of the following is an advantage of WFQ?
a. WFQ does not starve flows and guarantees throughput to all flows.
b. WFQ drops/punishes packets from most aggressive flows first.
c. WFQ is a standard queuing mechanism that is supported on most Cisco platforms.
d. All of the above.
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126 Chapter 4: Congestion Management and Queuing

11. Which of the following is not a disadvantage of WFQ?
a. WFQ does not offer guarantees such as bandwidth and delay guarantees to traffic flows.
b. FQ classification and scheduling are not configurable and modifiable.
c. You must configure flow-based queues for WFQ, and that is a complex task.
d. Multiple traffic flows may be assigned to the same queue within the WFQ system.
12. Which of the following is not true about CBWFQ?
a. CBWFQ allows creation of user-defined classes.
b. CBWFQ allows minimum bandwidth reservation for each queue.
c. CBWFQ addresses all of the shortcomings of WFQ.
d. Each of the queues in CBWFQ is a FIFO queue that tail drops by default.
13. Which of the following is not true about LLQ?
a. LLQ includes a strict-priority queue.
b. The LLQ strict priority queue is given priority over other queues.
c. The LLQ strict-priority queue is policed.
d. LLQ treats all traffic classes fairly.
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Introduction to Congestion Management and Queuing 127

Foundation Topics

Introduction to Congestion Management and Queuing
Congestion happens when the rate of input (incoming traffic switched) to an interface exceeds the
rate of output (outgoing traffic) from an interface. Why would this happen? Sometimes traffic
enters a device from a high-speed interface and it has to depart from a lower-speed interface; this
can cause congestion on the egress lower-speed interface, and it is referred to as the speed
mismatch problem. If traffic from many interfaces aggregates into a single interface that does not
have enough capacity, congestion is likely; this is called the aggregation problem. Finally, if
joining of multiple traffic streams causes congestion on an interface, it is referred to as the
confluence problem.

Figure 4-1 shows a distribution switch that is receiving traffic destined to the core from many
access switches; congestion is likely to happen on the interface Fa 0/1, which is the egress interface
toward the core. Figure 4-1 also shows a router that is receiving traffic destined to a remote office
from a fast Ethernet interface. Because the egress interface toward the WAN and the remote office
is a low-speed serial interface, congestion is likely on the serial 0 interface of the router.

Figure 4-1 Examples of Why Congestion Can Occur on Routers and Switches
Aggregating traffic from
access switches may
cause congestion here.

Fa0/1

Distribution Core

Switch

Access
Switches
Speed mismatch may
cause congestion here.

Fa0
Remote
S0 WAN
Traffic to Office
Remote
Office
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128 Chapter 4: Congestion Management and Queuing

A network device can react to congestion in several ways, some of which are simple and some of
which are sophisticated. Over time, several queuing methods have been invented to perform
congestion management. The solution for permanent congestion is often increasing capacity
rather than deploying queuing techniques. Queuing is a technique that deals with temporary
congestion. If arriving packets do not depart as quickly as they arrive, they are held and released.
The order in which the packets are released depends on the queuing algorithm. If the queue gets
full, new arriving packets are dropped; this is called tail drop. To avoid tail drop, certain packets
that are being held in the queue can be dropped so that others will not be; the basis for selecting
the packets to be dropped depends on the queuing algorithm. Queuing, as a congestion
management technique, entails creating a few queues, assigning packets to those queues, and
scheduling departure of packets from those queues. The default queuing on most interfaces, except
slow interfaces (2.048 Mbps and below), is FIFO. To entertain the demands of real-time, voice,
and video applications with respect to delay, jitter, and loss, you must employ more sophisticated
queuing techniques.

The queuing mechanism on each interface is composed of software and hardware components. If
the hardware queue, also called the transmit queue (TxQ), is not congested (full/exhausted), the
packets are not held in the software queue; they are directly switched to the hardware queue where
they are quickly transmitted to the medium on the FIFO basis. If the hardware queue is congested,
the packets are held in/by the software queue, processed, and released to the hardware queue based
on the software queuing discipline. The software queuing discipline could be FIFO, PQ, custom
queuing (CQ), WRR, or another queuing discipline.

The software queuing mechanism usually has a number of queues, one for each class of traffic.
Packets are assigned to one of those queues upon arrival. If the queue is full, the packet is dropped
(tail drop). If the packet is not dropped, it joins its assigned queue, which is usually a FIFO queue.
Figure 4-2 shows a software queue that is composed of four queues for four classes of traffic. The
scheduler dequeues packets from different queues and dispatches them to the hardware queue
based on the particular software queuing discipline that is deployed. Note that after a packet is
classified and assigned to one of the software queues, the packet could be dropped, if a technique
such as weighted random early detection (WRED) is applied to that queue.

As Figure 4-2 illustrates, when the hardware queue is not congested, the packet does not go
through the software queuing process. If the hardware queue is congested, the packet must be
assigned to one of the software queues (should there be more than one) based on classification of
the packet. If the queue to which the packet is assigned is full (in the case of tail-drop discipline)
or its size is above a certain threshold (in the case of WRED), the packet might be dropped. If the
packet is not dropped, it joins the queue to which it has been assigned. The packet might still be
dropped if WRED is applied to its queue and it is (randomly) selected to be dropped. If the packet
is not dropped, the scheduler is eventually going to dispatch it to the hardware queue. The
hardware queue is always a FIFO queue.
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Introduction to Congestion Management and Queuing 129

Figure 4-2 Router Queuing Components: Software and Hardware Components

HW FIFO
No
Queue Hardware Interface
Full? Queue (TxQ)

Yes
Software
Queue

Yes
Q1? Add/Drop Q1

No

Yes
Q2? Add/Drop Q2 Scheduler

No


Add/Drop Qn

Having both software and hardware queues offers certain benefits. Without a software queue, all
packets would have to be processed based on the FIFO discipline on the hardware queue. Offering
discriminatory and differentiated service to different packet classes would be almost impossible;
therefore, real-time applications would suffer. If you manually increase the hardware queue
(FIFO) size, you will experience similar results. If the hardware queue becomes too small, packet
forwarding and scheduling is entirely at the mercy of the software queuing discipline; however,
there are drawbacks, too. If the hardware queue becomes so small, for example, that it can hold
only one packet, when a packet is transmitted to the medium, a CPU interrupt is necessary to
dispatch another packet from the software queue to the hardware queue. While the packet is being
transferred from the software queue, based on its possibly complex discipline, to the hardware
queue, the hardware queue is not transmitting bits to the medium, and that is wasteful.
Furthermore, dispatching one packet at a time from the software queue to the hardware queue
elevates CPU utilization unnecessarily.

Many factors such as the hardware platform, the software version, the Layer 2 media, and the
particular software queuing applied to the interface influence the size of the hardware queue.
Generally speaking, faster interfaces have longer hardware queues than slower interfaces. Also, in
some platforms, certain QoS mechanisms adjust the hardware queue size automatically. The IOS
effectively determines the hardware queue size based on the bandwidth configured on the
interface. The determination is usually adequate. However, if needed, you can set the size of the
hardware queue by using the tx-ring-limit command from the interface configuration mode.
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130 Chapter 4: Congestion Management and Queuing

Remember that a too-long hardware queue imposes a FIFO style of delay, and a too-short
hardware queue is inefficient and causes too many undue CPU interrupts. To determine the size of
the hardware (transmit) queue on serial interfaces, you can enter the show controllers serial
command. The size of the transmit queue is reported by one of the tx_limited, tx_ring_limit, or
tx_ring parameters on the output of the show controllers serial command. It is important to know
that subinterfaces and software interfaces such as tunnel and dialer interfaces do not have their
own hardware (transmit) queue; the main interface hardware queue serves those interfaces. Please
note that the terms tx_ring and TxQ are used interchangeably to describe the hardware queue.

First-In-First-Out, Priority Queuing, Round-Robin, and Weighted
Round-Robin Queuing
FIFO is the default queuing discipline in most interfaces except those at 2.048 Mbps or lower (E1).
The hardware queue (TxQ) also processes packets based on the FIFO discipline. Each queue
within a multiqueue discipline is a FIFO queue. FIFO is a simple algorithm that requires no
configuration effort. Packets line up in a single FIFO queue; packet class, priority, and type play
no role in a FIFO queue. Without multiple queues and without a scheduling and dropping
algorithm, high-volume and ill-behaved applications can fill up the FIFO queue and consume all
the interface bandwidth. As a result, other application packets—for example, low volume and less
aggressive traffic such as voice—might be dropped or experience long delays. On fast interfaces
that are unlikely to be congested, FIFO is often considered an appropriate queuing discipline.

PQ, which has been available for many years, requires configuration. PQ has four queues
available: high-, medium-, normal-, and low-priority queues. You must assign packets to one of
the queues, or the packets will be assigned to the normal queue. Access lists are often used to
define which types of packets are assigned to which of the four queues. As long as the high-priority
queue has packets, the PQ scheduler forwards packets only from the high-priority queue. If the
high-priority queue is empty, one packet from the medium-priority queue is processed. If both the
high- and medium-priority queues are empty, one packet from the normal-priority queue is
processed, and if high-, medium-, and normal-priority queues are empty, one packet from the low-
priority queue is processed. After processing/de-queuing one packet (from any queue), the
scheduler always starts over again by checking if the high-priority queue has any packets waiting,
before it checks the lower priority queues in order. When you use PQ, you must both understand
and desire that as long as packets arrive and are assigned to the high-priority queue, no other queue
gets any attention. If the high-priority queue is not too busy, however, and the medium-priority
queue gets a lot of traffic, again, the normal- and low-priority packets might not get service, and
so on. This phenomenon is often expressed as a PQ danger for starving lower-priority queues.
Figure 4-3 shows a PQ when all four queues are holding packets.
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First-In-First-Out, Priority Queuing, Round-Robin, and Weighted Round-Robin Queuing 131

Figure 4-3 Priority Queuing
Packet

Yes
High? High-Priority Queue

No

Yes
Medium? Medium-Priority Queue

Scheduler Hardware Queue
No

No
Low? Normal-Priority Queue

Yes
Low-Priority Queue

In the situation depicted in Figure 4-3, until all the packets are processed from the high-priority
queue and forwarded to the hardware queue, no packets from the medium-, normal-, or low-
priority queues are processed. Using the Cisco IOS command priority-list, you define the traffic
that is assigned to each of the four queues. The priority list might be simple, or it might call an
access list. In this fashion, packets, based on their protocol, source address, destination address,
size, source port, or destination port, can be assigned to one of the four queues. Priority queuing
is often suggested on low-bandwidth interfaces in which you want to give absolute priority to
mission-critical or valued application traffic.

RR is a queuing discipline that is quite a contrast to priority queuing. In simple RR, you have a
few queues, and you assign traffic to them. The RR scheduler processes one packet from one queue
and then a packet from the next queue and so on. Then it starts from the first queue and repeats the
process. No queue has priority over the others, and if the packet sizes from all queues are (roughly)
the same, effectively the interface bandwidth is shared equally among the RR queues. If a queue
consistently has larger packets than other queues, however, that queue ends up consuming more
bandwidth than the other queues. With RR, no queue is in real danger of starvation, but the
limitation of RR is that it has no mechanism available for traffic prioritization.
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132 Chapter 4: Congestion Management and Queuing

A modified version of RR, Weighted Round Robin (WRR), allows you to assign a “weight” to
each queue, and based on that weight, each queue effectively receives a portion of the interface
bandwidth, not necessarily equal to the others. Custom Queuing (CQ) is an example of WRR, in
which you can configure the number of bytes from each queue that must be processed before it is
the turn of the next queue.

Basic WRR and CQ have a common weakness: if the byte count (weight) assigned to a queue is
close to the MTU size of the interface, division of bandwidth among the queues might not turn out
to be quite what you have planned. For example, imagine that for an interface with an MTU of
1500 bytes, you set up three queues and decide that you want to process 3000 bytes from each
queue at each round. If a queue holds a 1450-byte packet and two 1500-byte packets, all three of
those packets are forwarded in one round. The reason is that after the first two packets, a total of
2950 bytes have been processed for the queue, and more bytes (50 bytes) can be processed.
Because it is not possible to forward only a portion of the next packet, the whole packet that is
1500 bytes is processed. Therefore, in this round from this queue, 4450 bytes are processed as
opposed to the planned 3000 bytes. If this happens often, that particular queue consumes much
more than just one-third of the interface bandwidth. On the other hand, when using WRR, if the
byte count (weight) assigned to the queues is much larger than the interface MTU, the queuing
delay is elevated.

Weighted Fair Queuing
WFQ is a simple yet important queuing mechanism on Cisco routers for two important reasons:
first, WFQ is the default queuing on serial interfaces at 2.048 Mbps (E1) or lower speeds; second,
WFQ is used by CBWFQ and LLQ, which are two popular, modern and advanced queuing
methods. (CBWFQ and LLQ are discussed in the following sections of this chapter.) WFQ has the
following important goals and objectives:

■ Divide traffic into flows

■ Provide fair bandwidth allocation to the active flows

■ Provide faster scheduling to low-volume interactive flows

■ Provide more bandwidth to the higher-priority flows

WFQ addresses the shortcomings of both FIFO and PQ:

■ FIFO might impose long delays, jitter, and possibly starvation on some packets (especially
interactive traffic).

■ PQ will impose starvation on packets of lower-priority queues, and within each of the four
queues of PQ, which are FIFO based, dangers associated to FIFO queuing are present.
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Weighted Fair Queuing 133

WFQ Classification and Scheduling
WFQ is a flow-based queuing algorithm. Arriving packets are classified into flows, and each flow
is assigned to a FIFO queue. Flows are identified based on the following fields from IP and either
TCP or UDP headers:

■ Source IP address

■ Destination IP address

■ Protocol number

■ Type of service (ToS)

■ Source TCP/UDP port number

■ Destination TCP/UDP port number

A hash is generated based on the preceding fields. Because packets of the same traffic flow end up
with the same hash value, they are assigned to the same queue. Figure 4-4 shows that as a packet
arrives, the hash based on its header fields is computed. If the packet is the first from a new flow,
it is assigned to a new queue for that flow. If the packet hash matches an existing flow hash, the
packet is assigned to that flow queue.

Figure 4-4 Weighted Fair Queuing
Packet

Compute hash based
on packet header fields.

New Yes
New Queue
Flow?

No

Assign WFQ
packet to Scheduler Hardware Queue
an existing
queue based
on computed

hash.
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134 Chapter 4: Congestion Management and Queuing

Figure 4-4 does not show that, based on how full the interface hold queue is, and based on whether
the packet queue size is beyond a congestive discard threshold value, the packet might end up
being dropped. It is worth mentioning that when a packet arrives, it is assigned a sequence number
for scheduling purposes. The priority of a packet or flow influences its scheduling sequence
number. These concepts and mechanisms are discussed next.

NOTE The sequence number assigned to an arriving packet is computed by adding the
sequence number of the last packet in the flow queue to the modified size of the arriving packet.
The size of the arriving packet is modified by multiplying it by the weight assigned to the packet.
The weight is inversely proportional to the packet priority (from the ToS field). To illustrate this,
consider two packets of the same size but of different priorities arriving at the same time. The
two queues that these packets are mapped to are equally busy. The packet with the higher priority
gets a smaller scheduling sequence number and will most likely be forwarded faster than the
packet with the lower priority.

If all flows have the same priority (weight), WFQ effectively divides the interface bandwidth
among all the existing flows. As a result, low-volume interactive flows are scheduled and
forwarded to the hardware queue and do not end up with packets waiting in their corresponding
queues (or at least not for long). Packets of high-volume flows build up their corresponding queues
and end up waiting and delayed more and possibly dropped.

It is important to note that the number of existing queues in the WFQ system is based on the
number of active flows; in other words, WFQ dynamically builds and deletes queues. The interface
bandwidth is divided among the active flows/queues, and that division is partially dependent on
the priorities of those flows. Therefore, unlike CQ (and indeed CBWFQ, to be discussed in the
next section), WFQ does not offer precise control over bandwidth allocation among the flows.
Also, WFQ does not work with tunneling and encryption, because WFQ needs access to packet
header fields to compute the hash used for assigning packets to flow-based queues.

The number of queues that the WFQ system can build for the active flows is limited. The
maximum number of the queues, also called WFQ dynamic queues, is 256 by default. This number
can be set between 16 and 4096 (inclusive), but the number must be a power of 2. In addition to
the dynamic flows, WFQ allows up to 8 queues for system packets and up to 1000 queues for
RSVP flows. When the number of active flows exceeds the maximum number of dynamic queues,
new flows are assigned to the existing queues. Therefore, multiple flows might end up sharing a
queue. Naturally, in environments that normally have thousands of active flows, WFQ might not
be a desirable queuing discipline.
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Weighted Fair Queuing 135

WFQ Insertion and Drop Policy
WFQ has a hold queue for all the packets of all flows (queues within the WFQ system). The hold
queue is the sum of all the memory taken by the packets present in the WFQ system. If a packet
arrives while the hold queue is full, the packet is dropped. This is called WFQ aggressive dropping.
Aggressive dropping has one exception: if a packet is assigned to an empty queue, it is not
dropped.

Each flow-based queue within WFQ has a congestive discard threshold (CDT). If a packet arrives
and the hold queue is not full but the CDT of that packet flow queue is reached, the packet is
dropped. This is called WFQ early dropping. Early dropping has an exception: if a packet in
another queue has a higher (larger) sequence number than the arriving packet, the packet with the
higher sequence number is dropped instead. The dropped packet is assumed to belong to an
aggressive flow. It can be concluded that the early drop of WFQ punishes packets from aggressive
flows more severely and that packet precedence does not affect WFQ drop decisions.

Benefits and Drawbacks of WFQ
The main benefits of WFQ are as follows:

■ Configuring WFQ is simple and requires no explicit classification.

■ WFQ does not starve flows and guarantees throughput to all flows.

■ WFQ drops packets from the most aggressive flows and provides faster service to
nonaggressive flows.

■ WFQ is a standard and simple queuing mechanism that is supported on most Cisco platforms
and IOS versions.

WFQ has some limitations and drawbacks:

■ WFQ classification and scheduling are not configurable and modifiable.

■ WFQ is supported only on slow links (2.048 Mbps and less).

■ WFQ does not offer guarantees such as bandwidth and delay guarantees to traffic flows.

■ Multiple traffic flows may be assigned to the same queue within the WFQ system.

Configuring and Monitoring WFQ
WFQ is enabled by default on all serial interfaces that are slower than or equal to 2.048 Mbps. If
WFQ is disabled on an interface and you want to enable it or if you want to change its configurable
parameters, you can use the fair-queue command in the interface configuration mode. The
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136 Chapter 4: Congestion Management and Queuing

following shows the optional parameters that can be configured while you enter the fair-queue
command:

Router(config-if)# fair-queue [cdt [dynamic-queues [reservable-queues]]]
Router(config-if)# hold-queue max-limit out

This syntax also shows how the overall size of the WFQ system can be modified: the number of
packets an interface can hold in its outbound software queue can be set using the hold-queue max-
limit out command.

As you can see in this command syntax, configuring WFQ on an interface is simple. The cdt
parameter (congestive discard threshold) sets the number of packets allowed in each queue. The
default is 64, but you can change it to any power of 2 in the range from 16 to 4096. If a queue size
exceeds its CDT limit, new packets that are assigned to this queue are discarded. The dynamic-
queues parameter allows you to set the maximum number of flow queues allowed within the WFQ
system. This number can be between 16 and 4096 (inclusive) and must be a power of 2. (The
default is 256.) The parameter reservable-queues sets the number of allowed reserved
conversations. This number must be between 0 and 1000 (inclusive). (The default is 0.) Reservable
queues are used for interfaces that are configured for features such as Resource Reservation
Protocol (RSVP).

You can check the settings for the WFQ configurable parameters by using the output of the show
interface interface command. Example 4-1 displays sample output of this command. The queuing
strategy is stated to be weighted fair queuing. For the output queue, the current size, maximum
size (hold-queue max-limit), congestive discard threshold (per queue), and number of drops are
stated to be 0, 1000, 64, and 0, respectively. The current number of conversations is stated to be 0,
while it shows that a maximum of 10 conversations has been active during the measurement
interval. The maximum allowed number of concurrent conversations is shown to be 256, which is
the default value.

Example 4-1 Sample Output of the show interface Command
show interfaces serial 1/0
Router#s
Serial1/0 is up, line protocol is up
Hardware is CD2430 in sync mode
MTU 1500 bytes, BW 128000 Kbit, DLY 20000 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation FRAME-RELAY, loopback not set
Keepalive not set
LMI DLCI 1023 LMI type is CISCO frame relay DTE
FR SVC disabled, LAPF state down
Broadcast queue 0/64, broadcasts sent/dropped 105260/0, interface broadcasts 9 2894
Last input 00:00:00, output 00:00:02, output hang never
Last clearing of “show interface” counters 2d20h
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: weighted fair
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Weighted Fair Queuing 137

Example 4-1 Sample Output of the show interface Command (Continued)
Output queue: 0/1000/64/0 (size/max total/threshold/drops)
Conversations 0/10/256 (active/max active/max total)
Reserved Conversations 0/0 (allocated/max allocated)
Available Bandwidth 96000 kilobits/sec
5 minute input rate 2000 bits/sec, 1 packets/sec
5 minute output rate 2000 bits/sec, 0 packets/sec
228008 packets input, 64184886 bytes, 0 no buffer
Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort
218326 packets output, 62389216 bytes, 0 underruns
0 output errors, 0 collisions, 3 interface resets
0 output buffer failures, 0 output buffers swapped out
0 carrier transitions
DCD=up DSR=up DTR=up RTS=up CTS=up
!

You can obtain detailed information about the WFQ system on a particular interface (including a
particular virtual circuit) by using the show queue interface command. Example 4-2 shows
sample output of this command for your review. Observe that the output of this command for each
queue (conversation) displays the IP packet header fields that distinguish one flow from another.
Furthermore, for each conversation (queue), its depth (size), weight (related to distribution of
bandwidth), and other statistics are displayed individually.

Example 4-2 Sample Output of the show queue interface Command
Router# show queue atm2/0.33 vc 33
Interface ATM2/0.33 VC 0/33
Queueing strategy: weighted fair
Total output drops per VC: 18149
Output queue: 57/512/64/18149 (size/max total/threshold/drops)
Conversations 2/2/256 (active/max active/max total)
Reserved Conversations 3/3 (allocated/max allocated)

(depth/weight/discards/tail drops/interleaves) 29/4096/7908/0/0
Conversation 264, linktype: ip, length: 254
source: 10.1.1.1, destination: 10.0.2.20, id: 0x0000, ttl: 59,
TOS: 0 prot: 17, source port 1, destination port 1

(depth/weight/discards/tail drops/interleaves) 28/4096/10369/0/0
Conversation 265, linktype: ip, length: 254
source: 10.1.1.1, destination: 10.0.2.20, id: 0x0000, ttl: 59,
TOS: 32 prot: 17, source port 1, destination port 2
!
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138 Chapter 4: Congestion Management and Queuing

Class-Based Weighted Fair Queuing
CBWFQ addresses some of the limitations of PQ, CQ, and WFQ. CBWFQ allows creation of user-
defined classes, each of which is assigned to its own queue. Each queue receives a user-defined
(minimum) bandwidth guarantee, but it can use more bandwidth if it is available. In contrast to
PQ, no queue in CBWFQ is starved. Unlike PQ and CQ, you do not have to define classes of traffic
to different queues using complex access lists. WFQ does not allow creation of user-defined
classes, but CBWFQ does; moreover, defining the classes for CBWFQ is done with class maps,
which are flexible and user friendly, unlike access lists. Similar to WFQ and CQ, CBWFQ does
not address the low-delay requirements of real-time applications such as VoIP. The next section
discusses LLQ, which through the use of a strict priority queue provides a minimum but policed
bandwidth, plus a low-delay guarantee to real-time applications.

Figure 4-5 shows a CBWFQ with three user-defined classes. As each packet arrives, it is assigned
to one of the queues based on the class to which the packet belongs. Each queue has a reserved
bandwidth, which is a bandwidth guarantee.

Figure 4-5 CBWFQ
Packet 4

Packet 3

Packet 2

Packet 1

Yes Class 1 Queue
Class 1?
BW = 64 kbps
P4

No

P2 P1 P4 P3
Yes Class 2 Queue WFQ
Class 2?
BW = 128 kbps Scheduler
P3

No

Class 3 Queue
BW = 32 kbps
P2 P1
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Class-Based Weighted Fair Queuing 139

CBWFQ can create up to 64 queues, one for each user-defined class. Each queue is a FIFO queue
with a defined bandwidth guarantee and a maximum packet limit. If a queue reaches its maximum
packet limit, it incurs tail drop. To avoid tail drop, you can apply WRED to a queue. WRED is
discussed in the “Congestion Avoidance” section of Chapter 5, “Congestion Avoidance, Policing,
Shaping, and Link Efficiency Mechanisms.” Note that if you apply WRED to one (or more) of the
queues in CBWFQ, you cannot apply WRED directly to the interface, too. In addition to the 64
queues mentioned, a queue called class-default is always present. Packets that do not match any
of the defined classes are assigned to this queue. The 64 queues and the class-default queue are all
FIFO queues, but you can configure the class-default queue (but not the others) to be a WFQ. In
7500 series routers (and maybe others, by the time you read this book), you can configure all
queues to be WFQ. Just as you can apply WRED to any of the queues, you can apply WRED to
the class-default queue. The class-default queue, if you do not specify a reserved bandwidth for it,
uses any remaining bandwidth of the interface.

Classification, Scheduling, and Bandwidth Guarantee
Classification of traffic for the purpose of CBWFQ is done using Cisco IOS modular command-
line interface (MQC), specifically, using class maps. The options available for classification are
based on the IOS version. Furthermore, relevance of certain match criteria depends on the
interface, its encapsulation type, and any other options that might have been implemented on that
interface. For example, you can match the Frame Relay DE (discard eligible) bit only on a Frame
Relay interface. You should match MPLS EXP bits only if MPLS-IP packets are received;
matching CoS bits only makes sense on 802.1Q trunk connections.

Scheduling and the bandwidth guarantee offered to each queue within a CBWFQ system is based
on a weight that is assigned to it. The weight, in turn, is computed by the IOS based on the value
you enter for bandwidth, bandwidth percent, or bandwidth remaining percent on the class that is
assigned to the queue:

■ Bandwidth—Using the bandwidth command, you allocate (reserve) a certain amount of
bandwidth (Kbps) to the queue of a class. This bandwidth amount is subtracted (taken) from
the available/unreserved portion of the maximum reserved bandwidth of the interface. The
maximum reserved bandwidth of an interface is by default equal to 75 percent of the total
bandwidth of that interface, but it is modifiable. Maximum reserved bandwidth is set/modified
using the max-reserved-bandwidth command in the interface configuration mode.

■ Bandwidth percent—Using the bandwidth percent command, you allocate/reserve an
amount of bandwidth equal to a certain percentage of the interface bandwidth, to the queue
of a class. Whatever this amount of bandwidth turns out to be, it is subtracted from the
available/unreserved portion of the maximum reserved bandwidth of the interface. The Cisco
IOS determines the bandwidth of the serial interfaces based on the configured value using the
bandwidth statement.
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140 Chapter 4: Congestion Management and Queuing

■ Bandwidth remaining percent—Using the bandwidth remaining percent command, you
allocate a certain percentage of the remaining available bandwidth of the interface to the
queue of a class. Whatever this amount of bandwidth turns out to be, you subtract it from the
available/unreserved portion of the maximum reserved bandwidth of the interface.

NOTE When you configure the reserved bandwidth for each traffic class in a policy map, you
cannot use the bandwidth command for one class and the bandwidth percent command on
another class. In other words, for all classes within a policy map, you must use either the
bandwidth command or the bandwidth percent command, but not a mix of the two commands.

From the total bandwidth of an interface, a certain percentage is available for reservation; this
percentage is dictated by the value of a parameter called max-reserved-bandwidth on that
interface. The default value of maximum reserved bandwidth is 75, meaning that 75 percent of the
interface bandwidth can be reserved. However, as bandwidth reservation is made for different
queues (and possibly flows or tunnels), the amount of bandwidth remaining for new reservations
naturally diminishes. You can calculate the available bandwidth (available for reservation) based
on this formula:

Available bandwidth = (interface bandwidth x maximum reserved bandwidth) – (sum of
all existing reservations)
Note that the default value of 75 for maximum reserved bandwidth leaves 25 percent of interface
bandwidth for network overhead, including Layer 2 overhead such as CDP. You can modify the
default value for maximum reserved bandwidth, but you are cautioned to do so only if you are
aware of the consequences.

Benefits and Drawbacks of CBWFQ
The main benefits of CBWFQ are as follows:

■ It allows creation of user-defined traffic classes. These classes can be defined conveniently
using MQC class maps.

■ It allows allocation/reservation of bandwidth for each traffic class based on user policies and
preferences.

■ Defining a few (up to 64) fixed classes based on the existing network applications and user
policies, rather than relying on automatic and dynamic creation of flow-based queues (as
WFQ does), provides for finer granularity and scalability.

The drawback of CBWFQ is that it does not offer a queue suitable for real-time applications such
as voice or video over other IP applications. Real-time applications expect low-delay guarantee in
addition to bandwidth guarantee, which CBWFQ does not offer.
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Class-Based Weighted Fair Queuing 141

Configuring and Monitoring CBWFQ
The first step in configuring CBWFQ is defining traffic classes, which is done using class maps.
Example 4-3 shows two traffic classes: transaction-based and business-application. Any packet
that matches access list 100 is classified as transaction-based, and any packet that matches access
list 101 is classified as business-application.

Example 4-3 Class Maps Define Traffic Classes
!
class-map Transaction-Based
match access-group 100
!
class-map Business-Application
match access-group 101
!

Example 4-4 shows a policy map called Enterprise-Policy. This policy creates a queue with a
bandwidth guarantee of 128 Kbps and a maximum packet limit (queue limit) of 50 for the traffic
classified as transaction-based. Enterprise-Policy creates a second queue with a bandwidth
guarantee of 256 Kbps and a maximum packet limit (queue limit) of 90 for the traffic classified as
business-application. The default value for the queue-limit command is 64. Any traffic that does
not belong to transaction-based or business-application classes is assigned to the queue created for
the class-default class. The fair-queue 16 command applied to the class-default class changes its
queue discipline from FIFO to WFQ, and it sets the maximum number of dynamic queues for
WFQ to 16. You can set the number of dynamic queues from 16 to 4096 (inclusive), but the
number has to be a power of 2. Class-default has no bandwidth guarantees in this example.

Example 4-4 Policy Map
!
policy-map Enterprise-Policy
class Transaction-Based
Bandwidth 128
queue-limit 50
class Business-Application
bandwidth 256
queue-limit 90
class class-default
fair-queue 16
!

Example 4-5 shows the three alternative commands to reserve bandwidth for the queues of a
CBWFQ. Remember that within a policy map, one or the other option can be used, but you cannot
mix them within a single policy map.
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142 Chapter 4: Congestion Management and Queuing

Example 4-5 Three Alternative Ways to Reserve Bandwidth for CBWFQ Queues
!
policy-map Example-1
class A
Bandwidth 128
class B
bandwidth 64
!
policy-map Example-2
class C
bandwidth percent 30
class D
bandwidth percent 20
!
policy-map Example-3
class E
bandwidth remaining percent 20
class F
bandwidth remaining percent 20
!

Example 4-6 shows sample output of the show policy-map interface interface command. This
command displays information about the policy map applied to an interface using the service-
policy command. You can see the classes, bandwidth reservations, queuing disciplines, and traffic
statistics for each class, on the output.

Example 4-6 Sample Output of the show policy-map interface Command
Router# show policy-map interface e1/1
Ethernet1/1 output : po1
Weighted Fair Queueing
Class class1
Output Queue: Conversation 264
Bandwidth 937 (kbps) Max Threshold 64 (packets)
(total/discards/tail drops) 11548/0/0
Class class2
Output Queue: Conversation 265
Bandwidth 937 (kbps) Max Threshold 64 (packets)
(total/discards/tail drops) 11546/0/0
Class class3
Output Queue: Conversation 266
Bandwidth 937 (kbps) Max Threshold 64 (packets)
(total/discards/tail drops) 11546/0/0

Low-Latency Queuing
Neither WFQ nor CBWFQ can provide guaranteed bandwidth and low-delay guarantee to selected
applications such as VoIP; that is because those queuing models have no priority queue. Certain
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Low-Latency Queuing 143

applications such as VoIP have a small end-to-end delay budget and little tolerance to jitter (delay
variation among packets of a flow).

LLQ includes a strict-priority queue that is given priority over other queues, which makes it ideal
for delay and jitter-sensitive applications. Unlike the plain old PQ, whereby the higher-priority
queues might not give a chance to the lower-priority queues and effectively starve them, the LLQ
strict-priority queue is policed. This means that the LLQ strict-priority queue is a priority queue
with a minimum bandwidth guarantee, but at the time of congestion, it cannot transmit more data
than its bandwidth permits. If more traffic arrives than the strict-priority queue can transmit (due
to its strict bandwidth limit), it is dropped. Hence, at times of congestion, other queues do not
starve, and get their share of the interface bandwidth to transmit their traffic.

Figure 4-6 shows an LLQ. As you can observe, LLQ is effectively a CBWFQ with one or more
strict-priority queues added. Please note that it is possible to have more than one strict priority
queue. This is usually done so that the traffic assigned to the two queues—voice and video traffic,
for example—can be separately policed. However, after policing is applied, the traffic from the
two classes is not separated; it is sent to the hardware queue based on its arrival order (FIFO).

Figure 4-6 LLQ

BW
Policer No Strict
Drop Priority Queue
Packet?

Yes
Hardware Queue

Bit Bucket
Packet
classifier
No
Packet assigns Tail Drop? Class 1 Queue
packet to
a queue.

No
Tail Drop? Class 2 Queue CBWFQ
Scheduler

No
Tail Drop? Class N Queue

As long as the traffic that is assigned to the strict-priority class does not exceed its bandwidth limit
and is not policed and dropped, it gets through the LLQ with minimal delay. This is the benefit of
LLQ over CBWFQ.
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144 Chapter 4: Congestion Management and Queuing

Benefits of LLQ
LLQ offers all the benefits of CBWFQ, including the ability of the user to define classes and guarantee
each class an appropriate amount of bandwidth and to apply WRED to each of the classes (except to
the strict-priority queue) if needed. In the case of LLQ and CBWFQ, the traffic that is not explicitly
classified is considered to belong to the class-default class. You can make the queue that services the
class-default class a WFQ instead of FIFO, and if needed, you can apply WRED to it.

The benefit of LLQ over CBWFQ is the existence of one or more strict-priority queues with
bandwidth guarantees for delay- and jitter-sensitive traffic. The advantage of LLQ over the
traditional PQ is that the LLQ strict-priority queue is policed. That eliminates the chance of
starvation of other queues, which can happen if PQ is used. As opposed to the old RTP priority
queue, the LLQ strict-priority is not limited to accepting RTP traffic only. You can decide and
assign any traffic you want to the LLQ strict-riority queue using special IOS keywords, using
access lists, or using Network Based Application Recognition (NBAR) options. Finally, like many
other queuing mechanisms, LLQ is not restricted to certain platforms or media types.

Configuring and Monitoring LLQ
Configuring LLQ is almost identical to configuring CBWFQ, except that for the strict-priority
queue(s), instead of using the keyword/command bandwidth, you use the keyword/command
priority within the desired class of the policy map. You can reserve bandwidth for the strict-priority
queue in two ways: you can specify a fixed amount, or you can specify a percentage of the interface
bandwidth. The following command syntax is used to do just that in the appropriate order:

router(config-pmap-c)# priority bandwidth {burst}
router(config-pmap-c)# priority percent percentage {burst}

The burst amount (bytes) is specified as an integer between 32 and 2,000,000; it allows a
temporary burst above the policed bandwidth. Note that if the percent option is used, the
reservable amount of bandwidth is limited by the value of max-reserved-bandwidth on the
interface configuration, which is 75 percent by default.

Example 4-7 shows implementation of LLQ using a policy map called enterprise. The policy map
assigns a class called voice to the strict-priority queue with a bandwidth guarantee of 50 Kbps.
Classes business and class-default form the CBWFQ component of this LLQ.

Example 4-7 A Policy Map to Implement LLQ
router(config)# policy-map enterprise
router(config-pmap)# class voice
router(config-pmap-c)# priority 50
router(config-pmap)# class business
router(config-pmap-c)# bandwidth 200
router(config-pmap)# class class-default
router(config-pmap-c)# fair-queue
!
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Low-Latency Queuing 145

You can use the show policy-map interface interface command to see the packet statistics for all
classes used within a policy map that is applied to an interface using the service-policy command.
Example 4-8 shows (partial) output of this command for the serial 1/0 interface of a router.

Example 4-8 Sample Output of the show policy-map interface Command
router# show policy-map interface serial 1/0
Serial1/0
Service-policy output: AVVID (2022)
Class-map: platinum (match-all) (2035/5)
4253851 packets, 306277272 bytes
1 minute offered rate 499000 bps, drop rate 0 bps
Match: ip dscp 46 (2037)
Strict Priority
Output Queue: Conversation 264
Bandwidth 500 (kbps)
(pkts matched/bytes matched) 4248148/305866656
(total drops/bytes drops) 5/360
Class-map: silver (match-all) (2023/2)
251162 packets, 375236028 bytes
1 minute offered rate 612000 bps, drop rate 0 bps
Match: ip dscp 18 20 22 (2025)
Weighted Fair Queueing
Output Queue: Conversation 265
Bandwidth 25 (%)
(pkts matched/bytes matched) 3/4482
(depth/total drops/no-buffer drops) 0/0/0
mean queue depth: 0
Dscp Random drop Tail drop Minimum Maximum Mark
(Prec) pkts/bytes pkts/bytes threshold threshold probability
0(0) 0/0 0/0 20 40 1/10
1 0/0 0/0 22 40 1/10
2 0/0 0/0 24 40 1/10
3 0/0 0/0 26 40 1/10
4 0/0 0/0 28 40 1/10
(...up to DSCP 63......)
61 0/0 0/0 30 40 1/10
62 0/0 0/0 32 40 1/10
63 0/0 0/0 34 40 1/10
rsvp 0/0 0/0 36 40 1/10
.
<OUTPUT DELETED>
.
Class-map: class-default (match-any) (2039/0)
4719109 packets, 1000522466 bytes
1 minute offered rate 1625000 bps, drop rate 0 bps
Match: any (2041)
4719109 packets, 1000522466 bytes
1 minute rate 1625000 bps
!
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146 Chapter 4: Congestion Management and Queuing

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Congestion happens when the rate of input (incoming traffic switched) to an interface exceeds the
rate of output (outgoing traffic) from an interface. Aggregation, speed mismatch, and confluence
are three common causes of congestion. Queuing is a congestion management technique that
entails creating a few queues, assigning packets to those queues, and scheduling departure of
packets from those queues. Table 4-2 provides a comparative summary for the queuing disciplines
discussed in this chapter.

Table 4-2 Comparison of FIFO, PQ, WRR (CQ), WFQ, CBWFQ, and LLQ

Provides Adequate
a High- for Both
Allows Priority Delay-
User- Queue Sensitive
Default on Allows Definable for and
Some User- Interface Delay- Mission-
Queuing Router Number of Defined Bandwidth Sensitive Critical Configured
Discipline Interfaces Queues Classes Allocation Traffic Traffic Using MQC
FIFO Yes 1 No No No No No
PQ No 4 Yes No Yes No No
WRR (CQ) No User defined Yes Yes No No No
WFQ Yes Number of No No No No No
active flows
CBWFQ No User defined Yes Yes No No Yes
LLQ No User defined Yes Yes Yes Yes Yes
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Q&A 147

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. Why does congestion occur?
2. Define queuing.
3. What are three main tasks that congestion management/queuing mechanisms might perform?
4. What is the default queuing algorithm on Cisco router interfaces?
5. In what situation might FIFO be appropriate?
6. Describe priority queuing.
7. Cisco custom queuing is based on which queuing mechanism?
8. What are the Cisco router queuing components?
9. List the steps that a packet takes when it goes through an interface queuing system.
10. Describe WRR queuing.
11. Describe WFQ and its objectives.
12. How does WFQ define traffic flows?
13. Describe WFQ early dropping and aggressive dropping.
14. What are the benefits and drawbacks of WFQ?
15. What are the default values for CDT, dynamic queues, and reservable queues?
16. How do you adjust the hold queue size?
17. List at least two problems associated with PQ/CQ/WFQ.
18. Describe CBWFQ.
19. What are the three options for bandwidth reservation within CBWFQ?
20. How is available bandwidth calculated?
21. What are the benefits and drawbacks of CBWFQ?
22. How is CBWFQ configured?
23. Describe low-latency queuing.
24. What are the benefits of LLQ?
25. How do you configure LLQ?
1763fm.book Page 148 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:
■ Congestion Avoidance

■ Traffic Shaping and Policing

■ Link Efficiency Mechanisms
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CHAPTER 5
Congestion Avoidance,
Policing, Shaping, and Link
Efficiency Mechanisms
This chapter intends to give you an overview of three main quality of service (QoS) concepts:
congestion avoidance, traffic shaping and policing, and link efficiency mechanisms. Each
concept is presented in its own section. WRED and class-based WRED are the main
mechanisms covered in the “Congestion Avoidance” section. Traffic shaping and policing
concepts are explained in the second section; you will learn the purpose of these mechanisms
and where it is appropriate to use them. Different compression techniques, plus the concept of
link fragmentation and interleaving, are the topics of discussion in the third and final section of
this chapter.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read this entire chapter. The 12-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 5-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 5-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“Congestion Avoidance” 1–4
“Traffic Shaping and Policing” 5–8
“Link Efficiency Mechanisms” 9–12
Total Score (12 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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150 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 8 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 9–10 overall score—Begin with the “Foundation Summary” section and then follow up with
the “Q&A” section at the end of the chapter.

■ 11 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following is not a tail drop flaw?
a. TCP synchronization
b. TCP starvation
c. TCP slow start
d. No differentiated drop
2. Which of the following statements is not true about RED?
a. RED randomly drops packets before the queue becomes full.
b. RED increases the drop rate as the average queue size increases.
c. RED has no per-flow intelligence.
d. RED is always useful, without dependency on flow (traffic) types.
3. Which of the following is not a main parameter of a RED profile?
a. Mark probability denominator
b. Average transmission rate
c. Maximum threshold
d. Minimum threshold
4. Which of the following is not true about WRED?
a. You cannot apply WRED to the same interface as CQ, PQ, and WFQ.
b. WRED treats non-IP traffic as precedence 0.
c. You normally use WRED in the core routers of a network.
d. You should apply WRED to the voice queue.
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“Do I Know This Already?” Quiz 151

5. Which of the following is not true about traffic shaping?
a. It is applied in the outgoing direction only.
b. Shaping can re-mark excess packets.
c. Shaping buffers excess packets.
d. It supports interaction with Frame Relay congestion indication.
6. Which of the following is not true about traffic policing?
a. You apply it in the outgoing direction only.
b. It can re-mark excess traffic.
c. It can drop excess traffic.
d. You can apply it in the incoming direction.
7. Which command is used for traffic policing in a class within a policy map?
a. police
b. drop
c. remark
d. maximum-rate
8. Which of the following does not apply to class-based shaping?
a. It does not support FRF.12.
b. It classifies per DLCI or subinterface.
c. It understands FECN and BECN.
d. It is supported via MQC.
9. Which of the following is not a valid statement about compression?
a. Many compression techniques remove as much redundancy in data as possible.
b. A single algorithm might yield different compression ratios for different data types.
c. If available, compression is always recommended.
d. Compression can be hardware based, hardware assisted, or software based.
10. Which of the following is not true about Layer 2 payload compression?
a. It reduces the size of the frame payload.
b. It reduces serialization delay.
c. Software-based compression might yield better throughput than hardware-based com-
pression.
d. Layer 2 payload compression is recommended on all WAN links.
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152 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

11. Which of the following is the only true statement about header compression?
a. RTP header compression is not a type of header compression.
b. Header compression compresses the header and payload.
c. Header compression may be class based.
d. Header compression is performed on a session-by-session (end-to-end) basis.
12. Which of the following is not true about fragmentation and interleaving?
a. Fragmentation and interleaving is recommended when small delay-sensitive packets are
present.
b. Fragmentation result is not dependent on interleaving.
c. Fragmentation and interleaving might be necessary, even if LLQ is configured on the
interface.
d. Fragmentation and interleaving is recommended on slow WAN links.
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Congestion Avoidance 153

Foundation Topics

Congestion Avoidance
Congestion avoidance is used to avoid tail drop, which has several drawbacks. RED and its
variations, namely WRED and CBWRED, are commonly used congestion-avoidance techniques
used on Cisco router interfaces. Congestion avoidance is one of the main pieces of a QoS solution.

Tail Drop and Its Limitations
When the hardware queue (transmit queue, TxQ) is full, outgoing packets are queued in the
interface software queue. If the software queue becomes full, new arriving packets are tail-
dropped by default. The packets that are tail-dropped have high or low priorities and belong to
different conversations (flows). Tail drop continues until the software queue has room. Tail drop
has some limitations and drawbacks, including TCP global synchronization, TCP starvation, and
lack of differentiated (or preferential) dropping.

When tail drop happens, TCP-based traffic flows simultaneously slow down (go into slow start)
by reducing their TCP send window size. At this point, the bandwidth utilization drops
significantly (assuming that there are many active TCP flows), interface queues become less
congested, and TCP flows start to increase their window sizes. Eventually, interfaces become
congested again, tail drops happen, and the cycle repeats. This situation is called TCP global
synchronization. Figure 5-1 shows a diagram that is often used to display the effect of TCP global
synchronization.

Figure 5-1 TCP Global Synchronization
Link
Utilization

Peak

Average
Link
Utilization

Time
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154 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

The symptom of TCP global synchronization, as shown in Figure 5-1, is waves of congestion
followed by troughs during which time links are underutilized. Both overutilization, causing
packet drops, and underutilization are undesirable. Applications suffer, and resources are wasted.

Queues become full when traffic is excessive and has no remedy, tail drop happens, and aggressive
flows are not selectively punished. After tail drops begin, TCP flows slow down simultaneously,
but other flows (non-TCP), such as User Datagram Protocol (UDP) and non-IP traffic, do not.
Consequently, non-TCP traffic starts filling up the queues and leaves little or no room for TCP
packets. This situation is called TCP starvation. In addition to global synchronization and TCP
starvation, tail drop has one more flaw: it does not take packet priority or loss sensitivity into
account. All arriving packets are dropped when the queue is full. This lack of differentiated
dropping makes tail drop more devastating for loss-sensitive applications such as VoIP.

Random Early Detection
RED was invented as a mechanism to prevent tail drop. RED drops randomly selected packets
before the queue becomes full. The rate of drops increases as the size of queue grows; better said,
as the size of the queue grows, so does the probability of dropping incoming packets. RED does
not differentiate among flows; it is not flow oriented. Basically, because RED selects the packets
to be dropped randomly, it is (statistically) expected that packets belonging to aggressive (high
volume) flows are dropped more than packets from the less aggressive flows.

Because RED ends up dropping packets from some but not all flows (expectedly more aggressive
ones), all flows do not slow down and speed up at the same time, causing global synchronization.
This means that during busy moments, link utilization does not constantly go too high and too low
(as is the case with tail drop), causing inefficient use of bandwidth. In addition, average queue size
stays smaller. You must recognize that RED is primarily effective when the bulk of flows are TCP
flows; non-TCP flows do not slow down in response to RED drops. To demonstrate the effect of
RED on link utilization, Figure 5-2 shows two graphs. The first graph in Figure 5-2 shows how,
without RED, average link utilization fluctuates and is below link capacity. The second graph in
Figure 5-2 shows that, with RED, because some flows slow down only, link utilization does not
fluctuate as much; therefore, average link utilization is higher.

RED has a traffic profile that determines when packet drops begin, how the rate of drops change,
and when packet drops maximize. The size of a queue and the configuration parameters of RED
guide its dropping behavior at any given point in time. RED has three configuration parameters:
minimum threshold, maximum threshold, and mark probability denominator (MPD). When the
size of the queue is smaller than the minimum threshold, RED does not drop packets. As the size
of queue grows above the minimum threshold and continues to grow, so does the rate of packet
drops. When the size of queue becomes larger than the maximum threshold, all arriving packets
are dropped (tail drop behavior). MPD is an integer that dictates to RED to drop 1 of MPD (as
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Congestion Avoidance 155

many packets as the value of mark probability denominator), while the size of queue is between
the values of minimum and maximum thresholds.

Figure 5-2 Comparison of Link Utilization with and Without RED
Link
Utilization

TCP Flows Behavior Without RED

Peak

Average
Link
Utilization

Time

Link
Utilization

TCP Flows Behavior After RED

Peak
Average
Link
Utilization

Time

For example, if the MPD value is set to 10 and the queue size is between minimum and maximum
threshold, RED drops one out of ten packets. This means that the probability of an arriving packet
being dropped is 10 percent. Figure 5-3 is a graph that shows a packet drop probability of 0 when
the queue size is below the minimum threshold. It also shows that the drop probability increases
as the queue size grows. When the queue size reaches and exceeds the value of the maximum
threshold, the probability of packet drop equals 1, which means that packet drop will happen with
100 percent certainty.
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156 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Figure 5-3 RED Profile Demonstration
Drop
Probability

100%

Queue
Size

Min-Threshold Max-Threshold

The minimum threshold of RED should not be too low; otherwise, RED starts dropping packets
too early, unnecessarily. Also, the difference between minimum and maximum thresholds should
not be too small so that RED has the chance to prevent global synchronization. RED essentially
has three modes: no-drop, random-drop, and full-drop (tail drop). When the queue size is below
the minimum threshold value, RED is in the no-drop mode. When the queue size is between the
minimum and maximum thresholds, RED drops packets randomly, and the rate increases linearly
as the queue size grows. While in random-drop mode, the RED drop rate remains proportional to
the queue size and the value of the mark probability denominator. When the queue size grows
beyond the maximum threshold value, RED goes into full-drop (tail drop) mode and drops all
arriving packets.

Weighted Random Early Detection
WRED has the added capability of differentiating between high- and low-priority traffic,
compared to RED. With WRED, you can set up a different profile (with a minimum threshold,
maximum threshold, and mark probability denominator) for each traffic priority. Traffic priority is
based on IP precedence or DSCP values. Figure 5-4 shows an example in which the minimum
threshold for traffic with IP precedence values 0, 1, and 2 is set to 20; the minimum threshold for
traffic with IP precedence values 3, 4, and 5 is set to 26; and the minimum threshold for traffic with
IP precedence values 6 and 7 is set to 32. In Figure 5-4, the minimum threshold for RSVP traffic
is set to 38, the maximum threshold for all traffic is 40, and the MPD is set to 10 for all traffic.
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Congestion Avoidance 157

Figure 5-4 Weighted RED Profiles
Drop
Probability

100%

Average
Queue Size
20 26 30 32 34 36 38 40

Minimum Maximum-Threshold
Threshold for Minimum
Minimum
IP Precedence 0,1,2 Threshold Minimum
Threshold for
for IP Threshold
RSVP
Precedence 3,4,5 for IP
Precedence 6,7

WRED considers RSVP traffic as drop sensitive, so traffic from non-RSVP flows are dropped
before RSVP flows. On the other hand, non-IP traffic flows are considered least important and are
dropped earlier than traffic from other flows. WRED is a complementary technique to congestion
management, and it is expected to be applied to core devices. However, you should not apply
WRED to voice queues. Voice traffic is extremely drop sensitive and is UDP based. Therefore, you
must classify VoIP as the highest priority traffic class so that the probability of dropping VoIP
becomes very low.

WRED is constantly calculating the current average queue length based on the current queue
length and the last average queue length. When a packet arrives, first based on its IP precedence
or DSCP value, its profile is recognized. Next, based on the packet profile and the current average
queue length, the packet can become subject to random drop. If the packet is not random-dropped,
it might still be tail-dropped. An undropped packet is queued (FIFO), and the current queue size
is updated. Figure 5-5 shows this process.
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158 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Figure 5-5 WRED Operation

Arrived IP Packet Calculate
Average
Queue Size

Select WRED
Queue No
profile based on WRED FIFO
Full?
IP Prec or DSCP.
Queue
Random
Tail
Drop Yes
Drop

Bit Bit
Bucket Bucket

Class-Based Weighted Random Early Detection
When class-based weighted fair queueing (CBWFQ) is the deployed queuing discipline, each
queue performs tail drop by default. Applying WRED inside a CBWFQ system yields CBWRED;
within each queue, packet profiles are based on IP precedence or DSCP value. Currently, the only
way to enforce assured forwarding (AF) per-hop behavior (PHB) on a Cisco router is by applying
WRED to the queues within a CBWFQ system. Note that low-latency queuing (LLQ) is composed
of a strict-priority queue (policed) and a CBWFQ system. Therefore, applying WRED to the
CBWFQ component of the LLQ yields AF behavior, too. The strict-priority queue of an LLQ
enforces expedited forwarding (EF) PHB.

Configuring CBWRED
WRED is enabled on an interface by entering the random-detect command in the IOS interface
configuration mode. By default, WRED is based on IP precedence; therefore, eight profiles exist,
one for each IP precedence value. If WRED is DSCP based, there are 64 possible profiles. Non-
IP traffic is treated equivalent to IP traffic with IP precedence equal to 0. WRED cannot be
configured on an interface simultaneously with custom queuing (CQ), priority queuing (PQ), or
weighted fair queuing (WFQ). Because WRED is usually applied to network core routers, this
does not impose a problem. WRED has little performance impact on core routers.

To perform CBWRED, you must enter the random-detect command for each class within the
policy map. WRED is precedence-based by default, but you can configure it to be DSCP-based if
desired. Each traffic profile (IP precedence or DSCP based) has default values, but you can modify
those values based on administrative needs. For each IP precedence value or DSCP value, you can
set a profile by specifying a min-threshold, max-threshold, and a mark-probability-denominator.
With the default hold-queue size within the range 0 to 4096, the min-threshold minimum value is
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Congestion Avoidance 159

1 and the max-threshold maximum value is 4096. The default value for the mark probability
denominator is 10. The commands for enabling DSCP-based WRED and for configuring the
minimum-threshold, maximum-threshold, and mark-probability denominator for each DSCP
value and for each IP precedence value within a policy-map class are as follows:

Router(router-policy-c)# random-detect dscp-based
Router(router-policy-c)# random-detect dscp dscp-value min-threshold
max-threshold mark-prob-denominator
Router(router-policy-c)# random-detect precedence precedence-value
min-threshold max-threshold mark-prob-denominator

Applying WRED to each queue within a CBWFQ system changes the default tail-dropping
behavior of that queue. Furthermore, within each queue, WRED can have a different profile for
each precedence or DSCP value. Example 5-1 shows two class maps (Business and Bulk) and a
policy map called Enterprise that references those class maps. Class Business is composed of
packets with IP precedence values 3 and 4, and 30 percent of the interface bandwidth is dedicated
to it. The packets with precedence 4 are considered low drop packets compared to the packets with
a precedence value of 3 (high drop packets). Class Bulk is composed of packets with IP
precedence values of 1 and 2 and is given 20 percent of the interface bandwidth. The packets with
a precedence value of 2 are considered low drop packets compared to the packets with a
precedence value of 1 (high drop packets). The policy map shown in Example 5-1 applies fair-
queue and random-detect to the class class-default and provides it with the remainder of interface
bandwidth. Note that you can apply fair-queue and random-detect simultaneously to the class-
default only.

Example 5-1 CBWRED: IP Precedence Based
class-map Business
match ip precedence 3 4
class-map Bulk
match ip precedence 1 2
!
policy-map Enterprise
class Business
bandwidth percent 30
random-detect
random-detect precedence 3 26 40 10
random-detect precedence 4 28 40 10
class Bulk
bandwidth percent 20
random-detect
random-detect precedence 1 22 36 10
random-detect precedence 2 24 36 10
class class-default
fair-queue
random-detect
!
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160 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Please remember that you cannot simultaneously apply WRED and WFQ to a class policy.
(WRED random-detect and WFQ queue-limit commands are mutually exclusive.) Whereas
WRED has its own method of dropping packets based on max-threshold values per profile, WFQ
effectively tail-drops packets based on the queue-limit value. Recall that you cannot apply WRED
and PQ, CQ, or WFQ to an interface simultaneously either.

Example 5-2 shows a CBWRED case that is similar to the one given in Example 5-1; however,
Example 5-2 is DSCP based. All AF2s plus CS2 form the class Business, and all AF1s plus CS1
form the class Bulk. Within the policy map Enterprise, class Business is given 30 percent of the
interface bandwidth, and DSCP-based random-detect is applied to its queue. For each AF and CS
value, the min-threshold, max-threshold, and mark-probability-denominator are configured to
form four profiles within the queue. Class Bulk, on the other hand, is given 20 percent of the
interface bandwidth, and DSCP-based random-detect is applied to its queue. For each AF and CS
value, the min-threshold, max-threshold, and mark-probability-denominator are configured to
form four profiles within that queue.

Example 5-2 CBWRED: DSCP Based
class-map Business
match ip dscp af21 af22 af23 cs2
class-map Bulk
match ip dscp af11 af12 af13 cs1
!
policy-map Enterprise
class Business
bandwidth percent 30
random-detect dscp-based
random-detect dscp af21 32 40 10
random-detect dscp af22 28 40 10
random-detect dscp af23 24 40 10
random-detect dscp cs2 22 40 10
class Bulk
bandwidth percent 20
random-detect dscp-based
random-detect dscp af11 32 36 10
random-detect dscp af12 28 36 10
random-detect dscp af13 24 36 10
random-detect dscp cs1 22 36 10
class class-default
fair-queue
random-detect dscp-based
!

In Example 5-2, similar to Example 5-1, the fair-queue and random-detect commands are
applied to the class-default class. In Example 5-2 however, random-detect is DSCP-based as
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Congestion Avoidance 161

opposed to Example 5-1, where the class-default random-detect is based on IP precedence by
default.

Use the show policy-map interface interface command to see the packet statistics for classes on
the specified interface or PVC. (Service policy must be attached to the interface or the PVC.) The
counters that are displayed in the output of this command are updated only if congestion is present
on the interface. Note that the show policy-map interface command displays policy information
about Frame Relay PVCs only if Frame Relay traffic shaping (FRTS) is enabled on the interface;
moreover, ECN marking information is displayed only if ECN is enabled on the interface.
Example 5-3 shows a policy map called sample-policy that forms an LLQ. This policy assigns
voice class to the strict priority queue with 128 kbps reserved bandwidth, assigns gold class to a
queue from the CBWFQ system with 100 kbps bandwidth reserved, and assigns the silver class to
another queue from the CBWFQ system with 80 kbps bandwidth reserved and RED applied to it.

Example 5-3 A Sample Policy Map Implementing LLQ
policy-map sample-policy
class voice
priority 128
class gold
bandwidth 100
class silver
bandwidth 80
random-detect

Example 5-4 shows sample output of the show policy-map interface command for the serial 3/1
interface with the sample-policy (shown in Example 5-3) applied to it.

Example 5-4 Monitoring CBWFQ
Router# show policy-map interface serial3/1

Serial3/1

Service-policy output: sample-policy

Class-map: voice (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: ip precedence 5
Weighted Fair Queueing
Strict Priority
Output Queue: Conversation 264
Bandwidth 128 (kbps) Burst 3200 (Bytes)
(pkts matched/bytes matched) 0/0
(total drops/bytes drops) 0/0

continues
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162 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Example 5-4 Monitoring CBWFQ (Continued)

Class-map: gold (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: ip precedence 2
Weighted Fair Queueing
Output Queue: Conversation 265
Bandwidth 100 (kbps) Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: silver (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: ip precedence 1
Weighted Fair Queueing
Output Queue: Conversation 266
Bandwidth 80 (kbps)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0
exponential weight: 9
mean queue depth: 0

class Transmitted Random drop Tail drop Minimum Maximum Mark
pkts/bytes pkts/bytes pkts/bytes thresh thresh prob
0 0/0 0/0 0/0 20 40 1/10
1 0/0 0/0 0/0 22 40 1/10
2 0/0 0/0 0/0 24 40 1/10
3 0/0 0/0 0/0 26 40 1/10
4 0/0 0/0 0/0 28 40 1/10
5 0/0 0/0 0/0 30 40 1/10
6 0/0 0/0 0/0 32 40 1/10
7 0/0 0/0 0/0 34 40 1/10
rsvp 0/0 0/0 0/0 36 40 1/10

Class-map: class-default (match-any)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: any
!

In Example 5-4, WRED is applied only to the silver class, and WRED is IP precedence-based by
default. On the output of the show command for the silver class, you can observe the statistics for
nine profiles: eight precedence levels plus RSVP flow.
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Traffic Shaping and Policing 163

Traffic Shaping and Policing
Traffic shaping and policing are two different mechanisms for traffic conditioning. Both
mechanisms measure the rate of different traffic classes against a policy or SLA. SLA stands for
service level agreement, and it is usually set up between an enterprise and a service provider with
regard to bandwidth, traffic rates, reliability, availability, QoS, and billing matters. Traffic shaping
usually buffers the traffic that is in excess of the policy/agreement. Policing either drops the excess
traffic or changes its marking to a lower level (re-marking). Therefore, traffic shaping is applied
on an interface in the outbound direction, but traffic policing can be applied in either the inbound
or outbound direction. These traffic-conditioning mechanisms are often deployed at network edge.

Because traffic policing merely drops or re-marks excess traffic, it does not impose delay to the
conforming (non-excess) traffic. If the excess traffic is dropped, it has to be retransmitted. Traffic
policing can re-mark and transmit excess traffic instead of dropping it. Traffic shaping buffers
excess traffic and releases it steadily based on the policy specifications. Traffic shaping comes in
many variations, including class-based traffic shaping, FRTS, generic traffic shaping (GTS). Cisco
IOS traffic-shaping tools do not provide the means for re-marking traffic.

The main purposes for traffic policing are as follows:

■ To limit the traffic rate to a value less than the physical access rate—This is called
enforcing subrate access. When the customer pays for an access rate (for example, 1.544
Mbps) that is less than the physical access rate (for example, 155.52 Mbps) between customer
and service provider facilities, the provider uses rate limiting (policing) to enforce the subrate
value.

■ To limit the traffic rate for each traffic class—When an enterprise and service provider
have an SLA that states the maximum rate for each traffic class (or marking), the provider uses
traffic policing to enforce that SLA (at the edge).

■ To re-mark traffic—Traffic is usually re-marked if it exceeds the rate specified in the SLA.
Cisco IOS traffic policing allows you to mark and re-mark Layer 2 and Layer 3 protocol data
units (PDU) such as IP precedence, IP DiffServ Codepoint (DSCP), Ethernet 802.1Q/p class
of service (CoS), Frame Relay DE, and so on.

Following are the main purposes for traffic shaping:

■ To slow down the rate of traffic being sent to another site through a WAN service such
as Frame Relay or ATM—If the remote site or the carrier network becomes congested, the
sending device is usually notified (such as by Frame Relay backward explicit congestion
notification, or BECN), and it can buffer the traffic and drop its sending rate until the network
condition improves. Different access rates at two sites connected via a wide area service is a
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164 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

common situation called asymmetric circuit end access bandwidth. However, sometimes the
receiving end becomes congested and the senders have to slow down through shaping not due
to asymmetric bandwidths, but because many sites send traffic to a single site at once; this is
called aggregation.

■ To comply with the subscribed rate—A customer must apply traffic shaping to the traffic
being sent to the service provider WAN (FR or ATM) or Metro Ethernet networks.

■ To send different traffic classes at different rates—If an SLA specifies a particular
maximum rate for each traffic class (with specific markings), the sender must perform class-
based traffic shaping to prevent traffic from being dropped or re-marked.

Figure 5-6 shows an enterprise that has a central site and three remote sites connected using Frame
Relay virtual circuits. All the access rates at these sites are different. If the central site sends traffic
at 1.544 Mbps (T1) to site A, then site A, which has a 512-Kbps access rate, will have congestion
and possibly drops. To avoid that, the central site can shape the traffic being sent to site A. The rate
mismatch between the central site and site A is an example of an asymmetric bandwidth situation.
What will happen if sites A, B, and C simultaneously send traffic to the central site? In that case,
if all remote sites send traffic at their maximum access rates, congestion will result, this time at the
central site; that is because the aggregate traffic from the remote sites exceeds the access rate at
the central site.

Figure 5-6 Speed Mismatch and Aggregation Require Traffic Shaping

Central Site

1.544 Mbps
T1

WAN
512 Kbps 767 Kbps

1.544 Mbps

Site A Site C

Site B
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Traffic Shaping and Policing 165

Where does traffic shaping and traffic policing usually take place? The CE devices can perform
policing on the interfaces facing inside their site and enforce traffic rates. For instance, bulk traffic
such as file-transfer over the WAN can be limited to a specific rate. Service providers usually
perform policing on the edge device of their network on the interface receiving or sending traffic
to the customer devices. Traffic shaping is often performed on the customer edge (CE) device,
outbound on the interface sending traffic to remote sites over the provider backbone.

Traffic shaping and policing similarities and differences are as follows:

■ Both traffic shaping and traffic policing measure traffic; sometimes, different traffic classes
are measured separately.

■ Policing can be applied to the inbound and outbound traffic (with respect to an interface), but
traffic shaping applies only to outbound traffic.

■ Shaping buffers excess traffic and sends it according to a preconfigured rate, whereas policing
drops or re-marks excess traffic.

■ Shaping requires memory for buffering excess traffic, which creates variable delay and jitter.
Policing does not require extra memory, and it does not impose variable delay.

■ If policing drops packets, certain flow types such as TCP-based flows will resend dropped
traffic. Non-TCP traffic might resend a lot more traffic than just the dropped ones.

■ Policing can re-mark traffic, but traffic shaping does not re-mark traffic.

■ Traffic shaping can be configured based on network conditions and signals, but policing does
not respond to network conditions and signals.

Measuring Traffic Rates
The operating systems on Cisco devices measure traffic rates using a bucket and token scheme.
The token and bucket scheme has a few variations: single bucket with single rate, dual bucket with
single rate, and dual bucket with dual rates. The Cisco ONT course covers only the single bucket
with single rate model. To transmit one byte of data, the bucket must have one token. Tokens are
put into the bucket at the rate equivalent to the SLA rate; for example, for a Frame Relay virtual
circuit, the committed information rate (CIR) is used as the guide to replenish tokens in the bucket.
If the size of data to be transmitted (in bytes) is smaller than the number of tokens, the traffic is
called conforming; when traffic conforms, as many tokens as the size of data are removed from the
bucket, and the conform action, which is usually forward data, is performed. If the size of data to
be transmitted (in bytes) is larger than the number of tokens, the traffic is called exceeding. In the
exceed situation, tokens are not removed from the bucket, but the action performed (exceed action)
is either buffer and send data later (in the case of shaping) or drop or mark data (in the case of
policing).
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166 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Figure 5-7 shows that tokens are dropped into the bucket based on an SLA rate; it also shows
that if the bucket becomes full, excess tokens spill and are wasted in the single bucket model.
Furthermore, Figure 5-7 shows that when traffic is forwarded one token is needed for each byte
of data.

Figure 5-7 Single Bucket, Single Rate Token Bucket Scheme

T
Tokens are
T
replenished.
T When the bucket gets full
T T excess tokens are wasted
in the single bucket scheme.
T
Token
Bucket T T T
T T

As traffic is forwarded, one
token is used for each
byte of forwarded traffic.

It is important to know the definitions and relationships between the parameters within the token
bucket scheme. CIR stands for committed information rate, Bc stands for committed burst, and Tc
stands for committed time interval. The relationship between these parameters is as follows:

CIR (bits per second) = Bc (bits) / Tc (seconds)
Instead of continuously dropping tokens into the bucket, and when the bucket is full, discarding
the just-added tokens, the operating system adds tokens to the buckets only when there is traffic
activity. Every time a packet arrives, the operating system computes the time difference between
the arrival time of the new packet and the arrival time of the last packet, and for the time difference
computed, it adds the appropriate number of tokens according to this formula:

Number of tokens added = time difference (sec) × CIR / 8
The time difference between the current packet arrival time and the previous packet arrival time is
computed in seconds and then multiplied by CIR (which is expressed in bits per seconds) to
compute the number of bits. Then the result is divided by eight to compute the number of bytes.
The number of bytes computed indicates the number of tokens that should have been added to the
token bucket during the time between the arrival of the last packet and the arrival of the current
packet. The computed number of tokens is added to the bucket right away. The total number of
tokens (bytes) in the bucket cannot exceed the Bc value. Any extra tokens are discarded and
therefore wasted. Administrators usually specify the CIR and Bc values and let the system
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Link Efficiency Mechanisms 167

compute the Tc value automatically. The larger the Bc value, the larger burst of data that is
possible; with a large Bc value, the bucket saves more tokens.

Cisco IOS Policing and Shaping Mechanisms
Cisco IOS offers class-based traffic policing. Using modular QoS command-line interface (MQC),
class-based traffic policing is applied to a class within a policy map with the police command. As
stated in the previous section, Cisco IOS offers different Token Bucket schemes for policing:
single bucket/single rate, dual bucket/single rate, and dual bucket/dual rate. Furthermore,
multiaction policing—meaning taking multiple actions when traffic conforms, exceeds, or
violates—is also supported by class-based traffic policing. In addition to Cisco routers, class-
based traffic policing is available on some Cisco Catalyst switches.

Cisco IOS also offers class-based traffic shaping. Using MQC, class-based traffic shaping is
applied to a class within a policy map. When used in combination with CBWFQ, class-based
traffic shaping controls the upper limit of the outgoing traffic rate for a class, while the bandwidth
statement guarantees the minimum bandwidth or rate for that class.

Frame Relay traffic shaping controls Frame Relay traffic only and can be applied to a Frame Relay
subinterface or Frame Relay DLCI. Whereas Frame relay traffic shaping supports Frame Relay
fragmentation and interleaving (FRF.12), class-based traffic shaping does not. On the other hand,
both class-based traffic shaping and Frame Relay traffic shaping interact with and support Frame
Relay network congestion signals such as BECN and forward explicit congestion notification
(FECN). A router that is receiving BECNs shapes its outgoing Frame Relay traffic to a lower rate.
If it receives FECNs, even if it has no traffic for the other end, it sends test frames with the BECN
bit set to inform the other end to slow down.

Enterprises apply traffic policing at the access and distribution layers to control traffic entering the
core or leaving the campus toward the WAN circuits. Most enterprises apply traffic shaping on the
interfaces of the edge devices connected to WAN service. Traffic shaping is useful when speed
mismatch or aggregation occurs and you want to avoid congestion and drops. Service providers
apply traffic policing on the interfaces of the edge devices receiving traffic from customers; this
helps them meter different traffic class rates against the SLA rates. Service providers also apply
traffic shaping on the edge devices, sending traffic to customer sites.

Link Efficiency Mechanisms
The main link efficiency mechanisms deployed today are compression- and fragmentation-based.
There are several types of compression: link compression, layer 2 payload compression, RTP
header compression, and TCP header compression. Fragmentation is usually combined with
interleaving. Compression makes link utilization more efficient, and it is a QoS technique that
actually makes more bandwidth available. Fragmentation aims at reducing the expected delay of
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packets by reducing the maximum packet size over a circuit or connection. Compression is a
technique used in many of the link efficiency mechanisms. Compression reduces the size of data
to be transferred; therefore, it increases throughput and reduces overall delay. Many compression
algorithms have been developed over time. An example for a compression algorithm is Lempel-
Ziv (LZ) used by Stacker compression. Most compression algorithms take advantage of and
remove the repeated patterns and redundancy in data. One main difference between compression
algorithms is the type of data the algorithm has been optimized for. For example, MPEG has been
developed for and works well for compressing video, whereas the Huffman algorithm compresses
text-based data well.

The success of compression algorithms is measured and expressed by the ratio of raw data to
compressed data; a ratio of 2:1 is common. According to Shannon’s theorem, compression has a
theoretical limit; it is believed that algorithms of today that run on high-end CPUs can reach the
highest possible compression levels. If compression is hardware based, the main CPU cycles are
not used; on the other hand, if compression is software based, the main CPU is interrupted and its
cycles are used for performing compression. For that reason, when possible, hardware
compression is recommended over software compression. Some compression options are Layer 2
payload compression and upper layer (Layer 3 and 4) header compression.

Layer 2 Payload Compression
Layer 2 payload compression, as the name implies, compresses the entire payload of a Layer 2
frame. For example, if a Layer 2 frame encapsulates an IP packet, the entire IP packet is
compressed. Layer 2 payload compression is performed on a link-by-link basis; it can be
performed on WAN connections such as PPP, Frame Relay, high-level data link control (HDLC),
X.25, and Link Access Procedure, Balanced (LAPB). Cisco IOS supports Stacker, Predictor, and
Microsoft Point-to-Point Compression (MPPC) as Layer 2 compression methods. The primary
difference between these methods is their overhead and utilization of CPU and memory.

Because Layer 2 payload compression reduces the size of the frame, serialization delay is reduced.
Increase in available bandwidth (hence throughput) depends on the algorithm efficiency.
Depending on the complexity of the compression algorithm and whether the compression is
software based or hardware based (or hardware assisted), compression introduces some amount
of delay. However, overall delay (latency) is reduced, especially on low-speed links, whenever
Layer 2 compression is used. Layer 2 payload compression is useful over circuits or connections
that require the Layer 2 headers to remain in tact. For example, over a Frame Relay or ATM circuit
you can use Layer 2 payload compression. Link compression, on the other hand, compresses the
entire Layer 2 data unit including its header, which won’t work over Frame Relay and ATM, but
would work well over PPP or HDLC connections.

Figure 5-8 shows three cases, the first of which makes no use of payload compression. The second
and third scenarios in Figure 5-8 use software-based and hardware-based Layer 2 payload
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Link Efficiency Mechanisms 169

compression, respectively. Hardware compression and hardware-assisted compression are
recommended, because they are more CPU efficient than software-based compression. The
throughput gain that a Layer 2 payload compression algorithm yields depends on the algorithm
itself and has no dependency on whether it is software or hardware based. In Figure 5-8, hardware
compression resulted in the least overall delay, but its software compression counterpart yielded
better throughput results.

Figure 5-8 Layer 2 Payload Compression Options and Results
No Payload Throughput:
Compression 512 Kbps
BW = 512 Kbps

Delay = 8 ms
Delay = 1 ms Total Delay = 9 ms

Software-Based Throughput:
Payload Compression 1000 Kbps
BW = 512 Kbps

Delay = 8 ms
Delay = 10 ms Total Delay = 18 ms

Hardware-Assisted Throughput:
Payload Compression 716 Kbps
BW = 512 Kbps

Delay = 4 ms
Delay = 2 ms Total Delay = 6 ms

Throughput is dictated
by the effectiveness
of the compression
algorithm.

Header Compression
Header compression reduces serialization delay and results in less bandwidth usage, yielding
more throughput and more available bandwidth. As the name implies, header compression
compresses headers only; for example, RTP header compression compresses Real-time Transport
Protocol (RTP), User Datagram Protocol (UDP), and IP headers, but it does not compress the
application data. This makes header compression especially useful for cases in which application
payload size is small. Without header compression, the header (overhead)-to-payload (data) ratio
is large, but with header compression, the overhead-to-data ratio reduces significantly.

Common header compression options such as TCP header compression and RTP header
compression use a simple yet effective technique. Because the headers of the packets in a single
flow are identical (some exceptions may apply), instead of sending the identical (and relatively
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170 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

large) header with every packet, a number or index that refers to that entire header is sent instead.
This technique is based on a dictionary style of compression algorithms, in which phrases or
blocks of data are replaced with a short reference to that phrase or block of data. The receiving
end, based on the reference number or index, places the real header back on the packet and
forwards it.

When you enable TCP or RTP header compression on a link, all TCP or RTP flows are header-
compressed as a result. First, note that this is done on a link-by-link basis. Second, note that you
cannot enable the feature on a subset of sessions or flows. If you plan to perform header
compression on specific packet types or applications, what you need to do is class-based header
compression. Class-based header compression is performed by applying appropriate IOS
commands to the desired classes within a policy map using MQC.

Header compression is not CPU-intensive; therefore, the extra delay introduced due to header
compression is negligible. Assume that a 512-Kbps link exists between two routers, similar to the
one shown in Figure 5-9. In the first case shown in Figure 5-9, header compression is not used,
and the forwarding delay of 1 ms and data propagation delay of 8 ms yield a total delay of 9 ms
between the two routers shown. With no compression performed, the link throughput is the same
as the link bandwidth, which is 512 Kbps. In Figure 5-9, the link where header compression is
performed shows a processing delay of 2 ms but a data propagation delay of only 4 ms, yielding
a total delay of 6 ms. Furthermore, the link where header compression is performed provides more
throughput, in this case a total throughput of 716 Kbps.

Figure 5-9 Header Compression Results
No Header Throughput:
Compression 512 Kbps
BW = 512 Kbps

Delay = 8 ms
Delay = 1 ms Total Delay = 9 ms

Header Throughput:
Compression 716 Kbps
BW = 512 Kbps

Delay = 4 ms
Delay = 2 ms Total Delay = 6 ms

If the links shown in Figure 5-9 are configured as PPP links and RTP packets carry 20-byte voice
payloads through them, the header (overhead) to payload (data) ratio can be reduced significantly
with RTP header compression. Since a PPP header is 6 bytes long and RTP/UDP/IP headers add
up to 40 bytes, the header to payload ratio is (40 + 6) / 20, which equals 230 percent without RTP
header compression. On the other hand, with RTP header compression, if the no checksum option
is used, the RTP/UDP/IP header is reduced to 2 bytes only; the header (overhead)-to-payload
(data) ratio in this case reduces to (2 + 6) / 20, which equals 40 percent.
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Link Efficiency Mechanisms 171

Link Fragmentation and Interleaving
When an interface is congested, packets first go through the software queue and then are
forwarded to the hardware queue; when the interface has no congestion, packets skip the software
queue and go straight to the hardware queue. You can use advanced queuing methods such as LLQ
to minimize the software queuing delay that delay-sensitive packets such as VoIP experience.

Packets must always go through the hardware queue, which is FIFO based. If a VoIP packet ends
up behind one or more large packets in the hardware queue, it might experience too much delay
in that area and end up going over its total end-to-end delay budget. The goal for the end-to-end
delay budget of a VoIP packet (one-way) is 150 ms to 200 ms.

Imagine that a VoIP packet ends up in a Tx (HW) queue behind a 1500-byte frame that has to be
transmitted by the interface hardware out of a 256-Kbps link. The amount of time that the VoIP
packet has to wait for transmission of the 1500-byte frame ahead of it is 47 ms (1500 (bytes) × 8
(bits/byte) / 256000 (bits/sec)). Typically, during the design phase, a 10- to 15-ms delay budget is
allocated to serialization on slow links. This example clearly demonstrates that with the presence
of large data units, the delay will go much beyond the normally expected value.

It is possible to mitigate the delay imposed by the large data units ahead of VoIP (or other delay-
sensitive packets) in the hardware (Tx) queue. The solution is fragmentation and interleaving
(LFI). You must enable fragmentation on a link and specify the maximum data unit size (called
fragment size). Fragmentation must be accompanied by interleaving; otherwise, fragmentation
will have no effect. Interleaving allows packets of different flows to get between fragments of large
data units in the queue.

Applying Link Efficiency Mechanisms
Link efficiency mechanisms discussed in this section might not be necessary on all interfaces and
links. It is important that you identify network bottlenecks and work on the problem spots. On fast
links, many link efficiency mechanisms are not supported, and if they are, they might have
negative results. On slow links and where bottlenecks are recognized, you must calculate the
overhead-to-data ratios and consider all compression options. On some links, you can perform full
link compression. On some, you can perform Layer 2 payload compression, and on others, you
will probably perform header compression such as RTP or TCP header compression only. Link
fragmentation and interleaving is always a good option to consider on slow links. It is noteworthy
that compounding compression methods usually has an adverse affect and slows down throughput.

At the WAN edge on WAN links with equal or less bandwidth than T1/E1, it is recommended to
enable both TCP/RTP header compression and LFI. These improve WAN link utilization and
reduce the serialization delay. Because Layer 2 payload compression is CPU-intensive, it is
recommended only if it can be hardware-based or hardware-assisted.
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172 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Tail drop is the default queuing response to congestion. It has three significant drawbacks:

■ TCP synchronization—Packet drops from many sessions at the same time cause TCP
sessions to slow down (decrease send windows size) and speed up (increase send window) at
the same time, causing inefficient link utilization.

■ TCP starvation—Aggressive and non-TCP flows might fill up the queues, leaving little or
no room for other less aggressive applications and TCP packets (specially after slowdown).

■ No differentiated drop—Tail drop does not punish aggressive flows in particular, and it does
not differentiate between high- and low-priority packets.

RED avoids tail drop by randomly dropping packets when the queue size is above a min-threshold
value, and it increases drop rate as the average queue size increases. RED has the following
benefits:

■ Only the TCP sessions whose packets are dropped slow down.

■ The average queue size is kept small, reducing the chances of tail drop.

■ Link utilization becomes higher and more efficient.

RED, or RED profile, is configured using three parameters: minimum threshold, maximum
threshold, and mark probability denominator. When the average queue size is below the minimum
threshold, RED is in the no-drop mode. When the average queue size is between the minimum
threshold and the maximum threshold, RED is in random-drop mode. When the average queue
size is above the maximum threshold, RED is in full-drop mode.

WRED can use multiple profiles based on IP precedence (up to 8 profiles) or DSCP (up to 64
profiles). Using profiles, WRED can drop less important packets more aggressively than more
important packets. You can apply WRED to an interface, a virtual circuit (VC), or a class within a
policy map. The last case is called WRED, or CBWRED. CBWRED is configured in combination
with CBWFQ.
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Foundation Summary 173

Traffic-shaping and policing are traffic-conditioning tools. These mechanisms classify packets and
measure traffic rates. Shaping queues excess packets to stay within a desired rate, whereas policing
either re-marks or drops excess packets to keep them within a rate limit.

Policing is used to do the following:

■ Limit access to resources when high-speed access is used but not desired (subrate access)

■ Limit the traffic rate of certain applications or traffic classes

■ Mark down (recolor) exceeding traffic at Layer 2 or Layer 3

Shaping is used to do the following:

■ Prevent and manage congestion in ATM, Frame Relay, and Metro Ethernet networks, where
asymmetric bandwidths are used along the traffic path.

■ Regulate the sending traffic rate to match the subscribed (committed) rate in ATM, Frame
Relay, or Metro Ethernet networks.

Following are the similarities and differences between policing and shaping:

■ Both traffic shaping and traffic policing measure traffic. (Sometimes, different traffic classes
are measured separately.)

■ Policing can be applied to the inbound and outbound traffic (with respect to an interface), but
traffic shaping applies only to outbound traffic.

■ Shaping buffers excess traffic and sends it according to a preconfigured rate, whereas policing
drops or re-marks excess traffic.

■ Shaping requires memory for buffering excess traffic, which creates variable delay and jitter;
policing does not require extra memory, and it does not impose variable delay.

■ Policing can re-mark traffic, but traffic shaping does not re-mark traffic.

■ Traffic shaping can be configured to shape traffic based on network conditions and signals,
but policing does not respond to network conditions and signals.

The operating systems on Cisco devices measure traffic rates using a bucket and token scheme.
The important points to remember about a token bucket scheme are these:

■ To be able to transmit 1 byte of data, the bucket must have one token.

■ If the size of data to be transmitted (in bytes) is smaller than the number of tokens, the traffic
is called conforming. When traffic conforms, as many tokens as the size of data are removed
from the bucket, and the conform action, which is usually forward data, is performed.
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174 Chapter 5: Congestion Avoidance, Policing, Shaping, and Link Efficiency Mechanisms

■ If the size of data to be transmitted (in bytes) is larger than the number of tokens, the traffic
is called exceeding. In the exceed situation, tokens are not removed from the bucket, but the
action performed (exceed action) is either buffer and send data later (in case of shaping), or it
is drop or mark data (in the case of policing).

Similarities and differences between class-based shaping and FRTS are as follows:

■ FRTS controls Frame Relay traffic only and can be applied to a Frame Relay subinterface or
Frame Relay DLCI.

■ Whereas Frame Relay traffic shaping supports Frame Relay fragmentation and interleaving
(FRF.12), class-based traffic shaping does not.

■ Both class-based traffic shaping and FRTS interact with and support Frame Relay network
congestion signals such as BECN and FECN.

■ A router that is receiving BECNs shapes its outgoing Frame Relay traffic to a lower rate, and
if it receives FECNs, even if it has no traffic for the other end, it sends test frames with the
BECN bit set to inform the other end to slow down.

Compression identifies patterns in data and removes redundancy as much as possible. It increases
throughput and decreases latency. Many compression algorithms exist for different types of data.
Hardware compression (or hardware-assisted) is preferred over software compression because it
does not use main CPU cycles. Payload compression reduces the size of the payload. Header
compression reduces the header overhead.

Link efficiency mechanisms are often deployed on WAN links to increase the throughput and
decrease the delay and jitter. Cisco IOS link efficiency mechanisms include the following:

■ Layer 2 payload compression (Stacker, Predictor, MPPC)

■ Header compression (TCP, RTP, and class-based)

■ Link Fragmentation and Interleaving (LFI)
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Q&A 175

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. Name two of the limitations and drawbacks of tail drop.
2. Explain TCP global synchronization.
3. Explain TCP starvation.
4. Explain why RED does not cause TCP global synchronization.
5. What are the three configuration parameters for RED?
6. Briefly explain how WRED is different from RED.
7. Explain how class-based weighted random early detection is implemented.
8. Explain how assured forwarding per-hop behavior is implemented on Cisco routers.
9. List at least two of the main purposes of traffic policing.
10. List at least two of the main purposes of traffic shaping.
11. List at least four of the similarities and differences between traffic shaping and policing.
12. In the token bucket scheme, how many tokens are needed for each byte of data to be
transmitted?
13. Explain in the single bucket, single rate model when conform action and exceed action take
place.
14. What is the formula showing the relationship between CIR, Bc, and Tc?
15. Compare and contrast Frame Relay traffic shaping and class-based traffic shaping.
16. Briefly explain compression.
17. Briefly explain Layer 2 payload compression.
18. Provide a brief explanation for header compression.
19. Is it possible to mitigate the delay imposed by the large data units ahead of delay-sensitive
packets in the hardware (Tx) queue?
20. Where should link efficiency mechanisms be applied?
1763fm.book Page 176 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:

■ Implementing QoS Pre-Classify

■ Deploying End-to-End QoS
1763fm.book Page 177 Monday, April 23, 2007 8:58 AM

CHAPTER 6
Implementing QoS Pre-Classify
and Deploying End-to-End QoS

This brief chapter is composed of two sections. The first section is focused on the concept of
QoS pre-classify and how it is used to ensure that IOS QoS features work in conjunction with
tunneling and encryption. The second section deals with the topics related to deploying end-to-
end QoS. The concept of control plane policing is discussed last.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 10-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 6-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics.

Table 6-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score

“Implementing QoS Pre-Classify” 1–5

“Deploying End-to-End QoS” 6–10

Total Score (10 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to
the “Do I Know This Already?” Quizzes and Q&A Sections.” The suggested choices for your
next step are as follows:
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178 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

■ 6 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 7–8 overall score—Begin with the “Foundation Summary” section and then follow up with
the “Q&A” section at the end of the chapter.

■ 9 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following is not a critical function of VPNs?
a. Confidentiality
b. Data integrity
c. Authentication
d. Accounting
2. Which of the following is not a VPN type?
a. Client-initiated remote access
b. Internet
c. NAS-initiated remote access
d. Intranet
3. Which type of interface is QoS pre-classify for?
a. Tunnel interface
b. Loopback interface
c. VRF interface
d. Logical interface
4. Which of these QoS pre-classify deployment options is invalid?
a. Apply the policy to the tunnel interface without QoS pre-classify when you want to
classify packets based on the pre-tunnel header.
b. Apply the policy to the physical interface without QoS pre-classify when you want to
classify packets based on the post-tunnel header.
c. Apply the policy to the tunnel interface without QoS pre-classify when you want to
classify packets based on the post-tunnel header.
d. Apply the policy to the physical interface and enable QoS pre-classify when you want to
classify packets based on the pre-tunnel header.
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“Do I Know This Already?” Quiz 179

5. Which of the following Cisco IOS commands enables QoS pre-classify on an interface?
a. qos classify
b. preclassify
c. preclassify qos
d. qos pre-classify
6. Which of the following is not a typical IP QoS SLA parameter?
a. Delay
b. Jitter
c. CLP
d. Loss
7. Which of the following is an invalid campus QoS guideline?
a. Mark traffic at the distribution layer device.
b. Use multiple queues on the transmit interfaces.
c. Perform QoS in hardware when possible.
d. Police unwanted traffic as close to the source as possible.
8. Which of the following is not a Cisco router functional plane?
a. Data plane
b. Process plane
c. Management plane
d. Control plane
9. Which of the following is not a CoPP deployment step?
a. Define a packet classification criteria.
b. Define a service policy.
c. Enter global configuration mode and apply a QoS policy.
d. Enter control plane configuration mode and apply QoS policy.
10. Which of the following is not a typical customer edge-to-provider edge WAN link required
QoS feature?
a. Queuing
b. Shaping
c. LFI or cRTP
d. CoPP
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180 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

Foundation Topics

Implementing QoS Pre-Classify
QoS pre-classify was designed so that tunneled interfaces could classify packets on the output
interface before data was encrypted and tunneled. Considering the growth of VPN popularity, the
ability to classify traffic within a tunnel for QoS purposes is increasingly in demand. QoS pre-
classify allows Cisco IOS QoS features and services to remain effective even on tunnel interfaces
and when encryption is used. Therefore, service providers and customers can continue to provide
appropriate service levels to voice, video, and mission-critical traffic while they use VPN for
secure transport.

Virtual Private Networks (VPN)
Virtual private network (VPN) is a private connectivity path between two end points, built on a
public or shared infrastructure. Traditional ATM and Frame Relay circuits are referred to as Layer
2 VPNs, whereas IPsec tunnels over the Internet are called Layer 3 VPNs. A Layer 3 VPN can use
tunneling, encryption, or both. The three main functions that VPNs can provide are as follows:

■ Confidentiality—This is usually accomplished by encryption, using methods such as DES or
3DES. The intention is that eavesdroppers should not be able to decrypt/decipher and read the
encrypted data (within a reasonable period).

■ Authentication—This provides proof of origin to the receiver. Through authentication,
origin of information is certified and guaranteed by the receiver. Certificates are often
exchanged to facilitate the authentication process.

■ Data integrity—The receiver of packets and data is often interested in making sure that the
data has not been altered or corrupted during transit. A data integrity check using hashing
algorithms such as SHA and MD5 helps do just that.

You can implement confidentiality using encryption at different layers of the OSI model: at the
application layer, transport layer, Internet layer, or network interface (data link) layer. Secure Shell
(SSH) and Secure Multipurpose Internet Mail Extensions (S/MIME) are examples of protocols
that provide encryption or authentication services to applications. Secure Socket Layer (SSL) can
provide authenticity, privacy, and integrity to TCP-based applications. When these services are
offered at the application or transport layer, they only work for one or a few concurrent
applications; therefore, they are not considered application-independent and flexible. On the other
hand, security services at the data link layer are nonscalable and expensive because they must be
configured on a link/circuit-by-link/circuit basis. As a result, providing security services at Layer 3
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Implementing QoS Pre-Classify 181

(Internet layer), which means providing protection for as many applications as desired without
having to do multiple configurations, has become the most popular.

The end points of a VPN are either end systems or networks; based on that, VPNs are divided into
two categories:

■ Remote access VPNs

■ Site-to-site VPNs

The first category of VPN, remote access VPN, is either client-initiated or network access server
(NAS)-initiated. When a person uses a VPN client application to establish a secure tunnel across
an Internet service provider (ISP) (shared) network directly to an enterprise network, the VPN is
referred to as client-initiated. In the network access server (NAS)-initiated case, however, the user
dials in to the ISP, and the ISP NAS in turn establishes a secure tunnel to the enterprise private
network.

The second category of VPN, site-to-site VPN, has two main types: intranet VPN and extranet
VPN. The intranet VPN connects offices of an enterprise, such as the headquarters, branch offices,
and remote offices, over a public network. The extranet VPN, on the other hand, provides private
connectivity between offices of different companies and enterprises over a public infrastructure.
These companies and enterprises, due to their business relationship, have connectivity
requirements; suppliers, partners, or communities of interest often have such requirements.

QoS Pre-Classify Applications
Two commonly used tunneling protocols that are relevant to VPNs, discussed in the ONT course,
are GRE and IPsec. Because these tunneling protocols, at the tunnel end points, encapsulate the
original IP packet and use a new IP header, the original IP header is no longer available to the QoS
mechanisms on the outbound (egress) interface. The good news is that the original ToS byte of an
IP packet is copied to a ToS byte of the new IP header. Therefore, if the QoS mechanisms on the
egress interface only consider the ToS byte (DSCP and ECN, or IP Precedence), it is unnecessary
to perform any extra configurations, such as using the qos pre-classify command (on the router
where the tunnel emanates). However, if other fields—such as the source or destination IP address,
protocol number, or source port or destination port numbers—need to be processed, QoS pre-
classify configuration is necessary on the tunnel head router. When packets from different flows
and applications are transported through a tunnel interface, they are encapsulated with the same
new IP header with identical source and destination IP addresses (tunnel ends) and protocol
numbers (tunnel protocol number). The only difference between those packets might be the ToS
byte, which is directly copied from the ToS byte of the original IP packet header.

GRE can encapsulate different protocol packet types inside IP tunnels, creating a virtual point-to-
point link to remote Cisco routers over an IP internetwork. GRE, however, does not provide data
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182 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

confidentiality using encryption. The main strength of GRE tunnel is that, in addition to
transporting IP unicast packets, it can transport packets of IP routing protocols, multicast packets,
and non-IP traffic.

Secure VPNs use IPsec because it can provide data confidentiality, data integrity, and data
authentication between the tunnel end points/peers. Two mechanisms protect data sent over an
IPsec tunnel:

■ Authentication Header (AH)

■ Encapsulating Security Payload (ESP)

Using Secure Hash Algorithm (SHA) or Message Digest 5 (MD5), IPsec AH provides partial
integrity and authentication for IP datagrams. IP protocol number assigned to AH is 51. IPsec AH
can operate in either transport mode or tunnel mode. In transport mode the AH header is inserted
after the original IP packet’s header. In tunnel mode however, the original IP packet is entirely
encapsulated in another IP packet (new/outer) and the AH header in inserted after the
encapsulating/outer IP packet’s header. Figure 6-1 illustrates this. Tunnel mode is often used to
provide connectivity between networks that use private addressing; the outer IP packet’s address
is routable and allows delivery of the inner IP packet from one private site to another. Figure 6-1
shows two IPsec packets with AH headers: one in transport mode, and the other in tunnel mode.
IPsec AH alone does not provide data confidentiality through encryption.

Figure 6-1 IPsec AH in Tunnel Mode and in Transport Mode
IPsec AH in Tunnel Mode:
New IP Header
Payload IP Header AH Header Protocol: 51 (AH)
ToS = Inner ToS

IPsec AH in Transport Mode:
Original IP Header
Payload AH Header Protocol: 51 (AH)
Inner ToS

IPsec ESP provides data confidentiality (through encryption) and data authentication. If only the
payload of the IP packet needs to be protected, the ESP header is inserted between the IP header
and the IP payload, and only the IP payload is encrypted. This is ESP in transport mode. The IP
protocol number 50 identifies ESP. If the entire IP packet including its header needs to be
protected, the original IP packet is encrypted and encapsulated in another IP packet, with the ESP
header between the new IP header and the encapsulated and encrypted (original) IP header. This
is called ESP tunnel mode. Figure 6-2 shows two IPsec packets with ESP headers: one in transport
mode, and the other in tunnel mode.
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Implementing QoS Pre-Classify 183

Figure 6-2 IPsec ESP in Transport Mode and in Tunnel Mode
IPsec ESP in Tunnel Mode:
New IP Header
ESP Auth ESP Trailer Payload IP Header ESP Header Protocol = 50 (ESP)
ToS = Inner ToS

IPsec ESP in Transport Mode:
Original IP Header
ESP Auth ESP Trailer Payload ESP Header Protocol = 50 (ESP)
Inner ToS

Please note that in tunnel mode, with both IPsec AH and ESP, the original packet header ToS byte
is copied to the encapsulating IP packet header ToS byte; therefore, it is available for QoS
purposes. In transport mode, the entire original IP header is available for QoS processing.

QoS Pre-Classification Deployment Options
Many QoS features that are supported on physical interfaces are also supported on, and are often
required on, tunnel interfaces. A QoS service policy that is normally applied to a physical interface
can also be applied to a tunnel interface. In that situation, you must answer two questions:

1. Does the QoS policy classify an IP packet merely based on the ToS byte?
2. If the QoS policy classifies traffic based on fields other than or in addition to the ToS byte,
should the classification be done based on the values of those fields in the pre-tunnel IP packet
header or based on the values of those fields in the post-tunnel IP packet header?
With GRE tunnel, IPsec AH (transport and tunnel mode), and IPsec ESP (transport and tunnel
mode), if packet classification is ToS based only, no extra configuration is necessary. That is
because the IOS by default copies the ToS byte from the inner IP packet to the ToS byte of the
encapsulating IP packet when tunneling. Of course, when IPsec AH and IPsec ESP are in transport
mode, the original ToS byte is already present and available for examination. Therefore, the
challenge is presented when packet classification is based on fields other than or in addition to the
ToS byte on the pre-tunnel IP packet. A pre-tunnel IP packet means that, in addition to being
encapsulated, the inner IP packet of a tunnel may be encrypted.

The qos pre-classify command configures the IOS to make a temporary copy of the IP packet
before it is encapsulated or encrypted so that the service policy on the (egress) interface can do its
classification based on the original (inner) IP packet fields rather than the encapsulating (outer) IP
packet header. If the classification is merely based on ToS byte, though, qos pre-classify is not
necessary. A QoS service policy can be applied to the physical interface or to the tunnel interface.
Applying a service policy to a physical interface causes that policy to affect all tunnel interfaces
on that physical interface. Applying a service policy to a tunnel interface affects that particular
tunnel only and does not affect other tunnel interfaces on the same physical interface.
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184 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

When you apply a QoS service policy to a physical interface where one or more tunnels emanate,
the service policy classifies IP packets based on the post-tunnel IP header fields. However, when
you apply a QoS service policy to a tunnel interface, the service policy performs classification on
the pre-tunnel IP packet (inner packet). If you want to apply a QoS service policy to the physical
interface, but you want classification to be performed based on the pre-tunnel IP packet, you must
use the qos pre-classify command.

The qos pre-classify command is an interface configuration mode command that is not enabled
by default. This command is restricted to tunnel interfaces, virtual templates, and crypto maps, and
it is not available on any other interface type. Example 6-1 shows a QoS service policy called to-
remote-branch is applied to the serial1/1 interface of a router. A GRE tunnel with IPsec emanates
from this serial interface. Because in this example it is required that the QoS service policy
classifies pre-tunnel IP packets, the qos pre-classify command is applied to the tunnel1 interface
and to the crypto map named vpn.

Example 6-1 QoS Pre-Classification Example
interface serial1/1
ip address 10.1.1.1 255.255.255.252
service-policy output to-remote-branch
crypto map vpn

interface tunnel1
ip address 192.168.1.1 255.255.255.252
tunnel source serial1/1
tunnel destination 10.1.1.2
crypto map vpn
qos pre-classify

crypto map vpn 10 ipsec-isakmp
set peer 10.1.1.2
set transform-set remote-branch-vpn
match ip address 100
qos pre-classify

You might wonder why the service policy applied to the serial1/1 interface in Example 6-1 was
not applied to the tunnel1 interface instead. It is because, this way the service policy applies not
to just one, but to all the tunnels that emanate from that physical interface. Also, please notice that
the qos pre-classify command, in Example 6-1, is applied to both the tunnel1 and to the crypto
map called vpn. If the qos pre-classify command were not applied to the crypto map, the router
would see only one flow: the GRE tunnel (protocol 47).
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Deploying End-to-End QoS 185

Deploying End-to-End QoS
End-to-end QoS means that all the network components between the end points of a network
communication dialogue need to implement appropriate QoS mechanisms consistently. If, for
example, an enterprise (customer) uses the services and facilities of a service provider for
connectivity between its headquarters and branch offices, both the enterprise and the service
provider must implement the proper IP QoS mechanisms. This ensures end-to-end QoS for the
packets going from one enterprise location to the other, traversing through the core network of the
service provider.

At each customer location where traffic originates, traffic classification and marking need to be
performed. The connection between the customer premises equipment and the provider equipment
must have proper QoS mechanisms in place, respecting the packet markings. The service provider
might trust customer marking, re-mark customer traffic, or encapsulate/tag customer traffic with
other markings such as the EXP bits on the MPLS label. In any case, over the provider core, the
QoS levels promised in the SLA must be delivered. SLA is defined and described in the next
section. In general, customer traffic must arrive at the destination site with the same markings that
were set at the site of origin. The QoS mechanisms at the customer destination site, all the way to
the destination device, complete the requirements for end-to-end QoS.

Figure 6-3 shows several key locations within customer and provider premises where various QoS
mechanisms must be deployed. As the figure points out, end-to-end QoS is accomplished by
deploying proper QoS mechanisms and policies on both the customer (enterprise) devices and the
service provider (core) devices.

Figure 6-3 End-to-End QoS: Features and Related Implementation Points
End-to-End QoS = Enterprise QoS + Service Provider QoS
QoS at the WAN QoS in the Service
QoS in Campus Edge CE/PE Provider IP Cloud
Branch
Campus

Customer Customer
Edge Edge
Si
Si Service Provider IP
Cloud

IP

QoS—Campus Access QoS—Campus Distribution QoS—WAN Edge QoS—Service Provider Cloud

• Speed and Duplex Settings • Layer 3 Policing, Marking • Define SLA • Capacity Planning
• Classification and Trust on IP • Multiple Queues on • Classification, Marking • DiffServ Backbone
• Phone and Access Switch Switch Ports • Low-Latency Queuing • Low-Latency Queuing
• Multiple Queues on • Priority Queuing for VoIP • Link Fragmentation and or MDRR
Switch Ports • WRED within Data Queue Interleaving • WRED
• Priority Queuing for VoIP for Congestion Avoidance • WRED and Shaping
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186 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

Correct end-to-end per-hop behavior (PHB) for each traffic class requires proper implementation
of QoS mechanisms in both the enterprise and the service provider networks. In the past, IP QoS
was not given much attention in enterprise campus networks because of abundant available
bandwidth. Today, however, with the emergence of applications such as IP Telephony,
videoconferencing, e-learning, and mission-critical data applications in the customer campus
networks, many factors such as packet drop management, buffer management, queue
management, and so on, in addition to bandwidth management, are required within the campus to
minimize loss, delay, and jitter.

Figure 6-3 displays some of the main requirements within the campus and provider building
blocks constituting the end-to-end QoS. Proper hardware, software, and configurations such as
buffer settings and queuing are the focus of the access layer QoS. Policing, marking, and
congestion avoidance are implemented at the campus distribution layer. At the WAN edge, it is
often required to make some complex QoS configurations related to the subscribed WAN service.
Finally, in the service provider IP core, congestion-management and congestion-avoidance are the
main mechanisms in operation; the key QoS mechanisms used within the service provider core IP
network are low-latency queuing (LLQ) and weighted random early detection (WRED).

QoS Service Level Agreements (SLAs)
An SLA is a contractual agreement between an enterprise (customer) and a service provider
regarding data, voice, and other service or a group of services. Internet access, leased line, Frame
Relay, and ATM are examples of such services. After the SLA is negotiated, it is important that it
is monitored for compliance of the parties involved with the terms of the agreement. The service
provider must deliver services as per the qualities assured in the SLA, and the customer must
submit traffic at the rates agreed upon, to receive the QoS level assured by the SLA. Some of the
QoS parameters that are often explicitly negotiated and measured are delay, jitter, packet loss,
throughput, and service availability. The vast popularity of IP Telephony, IP conferencing, e-
learning, and other real-time applications has made QoS and SLA negotiation and compliance
more important than ever.

Traditionally, enterprises obtained Layer 2 service from service providers. Virtual circuits (VC),
such as permanent VCs, switched VCs, and soft PVCs, provided connectivity between remote
customer sites, offering a variety of possible topologies such as hub and spoke, partial mesh, and
full mesh. Point-to-point VCs have been the most popular type of circuits with point-to-point SLA
assurances from the service provider. With Layer 2 services, the provider does not offer IP QoS
guarantees; the SLA is focused on Layer 1 and Layer 2 measured parameters such as availability,
committed information rate (CIR), committed burst (Bc), excess burst (Be), and peak information
rate. WAN links sometimes become congested; therefore, to provide the required IP QoS for voice,
video, and data applications, the enterprise (customer) must configure its equipment (WAN
routers) with proper QoS mechanisms. Examples of such configurations include Frame Relay
traffic shaping, Frame Relay fragmentation and interleaving, TCP/RTP header compression, LLQ,
and class-based policing.
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Deploying End-to-End QoS 187

In recent years, especially due to the invention of technologies such as IPsec VPN and MPLS
VPN, most providers have been offering Layer 3 services instead of, or at least in addition to, the
traditional Layer 2 services (circuits). In summary, the advantages of Layer 3 services are
scalability, ease of provisioning, and service flexibility. Unlike Layer 2 services where each circuit
has a single QoS specification and assurance, Layer 3 services offer a variety of QoS service levels
on a common circuit/connection based on type or class of data (marking). For example, the
provider and the enterprise customer might have an SLA based on three traffic classes called
controlled latency, controlled load, and best effort. For each class (except best effort) the SLA
states that if it is submitted at or below a certain rate, the amount of data drop/loss, delay, and jitter
will be within a certain limit; if the traffic exceeds the rate, it will be either dropped or re-marked
(lower).

It is important that the SLA offered by the service provider is understood. Typical service provider
IP QoS SLAs include three to five traffic classes: one class for real-time, one class for mission-
critical, one or two data traffic classes, and a best-effort traffic class. The real-time traffic is treated
as a high-priority class with a minimum, but policed, bandwidth guarantee. Other data classes are
also provided a minimum bandwidth guarantee. The bandwidth guarantee is typically specified as
a percentage of the link bandwidth (at the local loop). Other parameters specified by the SLA for
each traffic class are average delay, jitter, and packet loss. If the interface on the PE device serves
a single customer only, it is usually a high-speed interface, but a subrate configuration offers the
customer only the bandwidth (peak rate) that is subscribed to. If the interface on a PE device serves
multiple customers, multiple VLANs or VCs are configured, each serving a different customer. In
that case, the VC or subinterface that is dedicated to each customer is provisioned with the subrate
configuration based on the SLA.

Figure 6-4 displays a service provider core network in the middle offering Layer 3 services with
IP QoS SLA to its customer with an enterprise campus headquarters and an enterprise remote
branch. The customer in this example runs IP Telephony applications such as VoIP calls between
those sites.

To meet the QoS requirements of the typical telephony (VoIP) applications, the enterprise must
not only negotiate an adequate SLA with the provider, but it also must make proper QoS
configurations on its own premise devices so that the end-to-end QoS becomes appropriate for the
applications. In the example depicted in Figure 6-4, the SLA guaranteed a delay (latency) <= 60
ms, jitter <= 20 ms, and packet loss <= 0.5 percent. Because the typical end-to-end objectives for
delay, jitter, and packet loss for VoIP are <= 150 ms, <= 30 ms, and <= 1 percent, respectively, the
enterprise must make sure that the delay, jitter, and loss within its premises will not exceed 90 ms,
10 ms, and 0.5 percent, respectively.
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188 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

Figure 6-4 Example for IP QoS SLA for VoIP
Maximum One-Way End-to-End QoS Budget for Delay (Latency), Jitter, and Packet Loss:
Latency <= 150 ms/Jitter <= 30 ms/Loss <= 1%

Enterprise
Campus Enterprise
Headquarters Service Provider Remote Branch
Provider Provider
Edge Edge
Provider

Customer Customer
Edge Edge
Si
Si IP

IP

Maximum One-Way
Service Provider Service Levels
Latency <= 60 ms
Jitter <= 20 ms
Loss <= 0.5%

Enterprise Campus QoS Implementations
Provisioning QoS functions and features within campus networks on access, distribution, and core
switches is a part of end-to-end QoS. This is in large part due to the growth of IP Telephony, video
conferencing, e-learning, and mission-critical applications within enterprise networks. Certain
efforts are spent on access devices, whereas others are spent on distribution and core equipment
to minimize packet loss, delay, and jitter. Figure 6-5 shows the series of devices—all the way from
end-user workstation (PC) to the core campus LAN switch and WAN edge router—that exist in a
typical campus LAN environment.

Figure 6-5 Typical Campus LAN Devices

Campus LAN WAN Edge

IP
Si Si WAN

Access Router
Switch Distribution Core
Switch Switch
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Deploying End-to-End QoS 189

Important guidelines for implementing QoS in campus networks are as follows:

■ Classify and mark traffic as close to the source as possible—Classifying and marking
packets as close to the source as possible, and not necessarily by the source, eliminates the
need for classification and marking efforts to be repeated and duplicated at various network
locations. However, marking of all end devices cannot be trusted either, because it opens the
door to abuse.
■ Police traffic as close to the source as possible—Policing unwanted traffic as close to the
source as possible is the most efficient way of handling excessive and possibly invasive and
malicious traffic. Unwanted traffic, such as denial of service (DoS) attacks and worm attacks,
can cause network outage by overwhelming network resources, device CPUs, and memories.
■ Establish proper trust boundaries—Trust Cisco IP phones marking, but not the markings
of user workstations (PCs).
■ Classify and mark real-time voice and video as high-priority traffic— Higher priority
marking for voice and video traffic gives them queue assignment, delay, jitter, and drop
advantage over other types of traffic.
■ Use multiple queues on transmit interfaces—This minimizes transmit queue congestion
and packet drops and delays due to transmit buffer congestion.
■ When possible, perform hardware-based rather than software-based QoS—Contrary to
Cisco IOS routers, Cisco Catalyst switches perform QoS functions in special hardware
(application-specific integrated circuits, or ASICs). Use of ASICs rather than software-based
QoS is not taxing on the main processor and allows complex QoS functions to be performed
at high speeds.
Congestion is a rare event within campus networks; if it happens, it is usually instantaneous and
does not sustain. However, critical and real-time applications (such as Telephony) still need the
protection and service guarantees for those rare moments. QoS features such as policing, queuing,
and congestion avoidance (WRED) must be enabled on all campus network devices where
possible. Within campus networks, link aggregation, oversubscription on uplinks, and speed
mismatches are common causes of congestion. Enabling QoS within the campus is even more
critical under abnormal network conditions such as during DoS and worm attacks. During such
attacks, network traffic increases and links become overutilized. Enabling QoS features within the
campus network not only provides service-level guarantees for specific application types, but it
also provides network availability assurance, especially during network attacks.

In campus networks, access switches require these QoS policies:

■ Appropriate trust, classification, and marking policies

■ Policing and markdown policies

■ Queuing policies
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190 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

The distribution switches, on the other hand, need the following:

■ DSCP trust policies

■ Queuing policies

■ Optional per-user micro-flow policies (if supported)

WAN Edge QoS Implementations
WAN edge QoS configurations are performed on CE and PE devices that terminate WAN circuits.
Commonly used WAN technologies are Frame Relay and ATM. Important QoS features
implemented on the CE and PE devices are LLQ, compression, fragmentation and interleaving,
policing, and shaping. Figure 6-6 shows a customer site connected to a provider IP network
through a Frame Relay connection between a CE device and a PE device. Note that a similar
connection between the CE and the PE devices exists at the remote site.

Figure 6-6 WAN Edge QoS Implementation Points

Service Provider
IP Backbone

Provider Edge

Frame
WAN Edge
Relay

Customer Edge

Si
Si

IP
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Deploying End-to-End QoS 191

For the traffic that is leaving a local customer site through a CE device going toward the provider
network and entering it through a PE device, output QoS mechanisms on the CE device and input
QoS mechanisms on the PE device must be implemented. The implementation will vary
somewhat, depending on whether the CE device is provider managed—in other words, if it is
under the control of the provider. Table 6-2 shows the QoS requirements on the CE and the PE
devices in both cases: when the CE device is provider managed, and when the CE device is not
provider managed. When the CE device is provider managed, the service provider manages and
performs the QoS policies and configurations. Otherwise, the customer enterprise controls the
QoS policies and configures the CE device.

Table 6-2 QoS Mechanisms Necessary for Traffic Leaving a Customer Site

Managed CE Unmanaged CE

The service provider controls the output QoS The service provider does not control the output
policy on the CE. QoS policy on the CE.

The service provider enforces SLA using the The service provider enforces SLA using the input
output QoS policy on the CE. QoS policy on the PE.

The output policy uses queuing, dropping, and The input policy uses policing and marking.
shaping.

Elaborate traffic classification or mapping of Elaborate traffic classification or mapping of
existing markings takes place on the CE. existing markings takes place on the PE.

LFI* and compressed RTP might be necessary.

* LFI = link fragmentation and interleaving

When the CE device is provider managed, the provider can enforce the SLA by applying an
outbound QoS service policy on the CE device. For example, an LLQ or class-based weighted fair
queueing (CBWFQ) can be applied to the egress interface of the CE device to provide a policed
bandwidth guarantee to voice and video applications and a minimum bandwidth guarantee to
mission-critical applications. You can use class-based shaping to rate-limit data applications. You
can apply congestion avoidance (WRED), shaping, compression (cRTP), and LFI outbound on the
managed CE device. When the CE device is not provider managed, the provider enforces the SLA
on the ingress interface of the PE device using tools such as class-based policing. Policed traffic
can be dropped, re-marked, or mapped (for example, DSCP to MPLS EXP).

At the remote site, where the traffic leaves the provider network through the PE router and enters
the enterprise customer network through the CE device, most of the QoS configurations are
configured on the PE device (outbound), regardless of whether the CE device is managed. The
service provider enforces the SLA using output QoS policy on the PE device. Output policy
performs congestion management (queuing), dropping, and possibly shaping. Other QoS
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192 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

techniques such as LFI or cRTP can also be performed in the PE device. CE input policy is usually
not necessary; of course, the configuration of the CE device that is not provider managed is at the
discretion of the enterprise customer.

Control Plane Policing (CoPP)
Control plane attacks are growing, so protecting the network infrastructure against these types of
attacks is imminently required. Control plane policing (CoPP), a Cisco IOS feature that has been
available since IOS release 12.2(18)S, allows you to configure a QoS filter that manages the traffic
flow of control plane packets. Using CoPP, you can protect the control plane of Cisco IOS routers
and switches against DoS and reconnaissance attacks and ensure network stability (router/switch
stability in particular) during an attack. Deploying CoPP is a recommended best practice and a key
protection mechanism.

The route processor routes and forwards the majority of the traffic that enters a router; the
destination of this type of traffic, called data plane traffic, is elsewhere other than the router itself.
On the other hand, some traffic—such as routing updates, management traffic, keepalives, and so
on—is indeed for the router; this type of traffic is called control and management plane traffic.
Formally, the Cisco router functional planes are enlisted as data plane, management plane, control
plane, and service plane. Excessive and malicious traffic in the form of control and management
traffic aimed at the route processor can have the following devastating results:

■ High or close to 100 percent utilization of CPU or other resources such as memory and buffers
■ Loss of routing updates and keepalives, resulting in route flaps and erroneous NLRI (network
layer reachability information) withdrawals and updates
■ Slow response times and interactive sessions, including command-line interface (CLI)
through virtual terminal lines
■ Queue buildups, resulting in excessive delays and tail drops, or drops due to lack of buffer space
CoPP mitigates control plane attacks and ensures stability and availability of the routers and
switches. CoPP is configured by applying a policy map to the control plane from the control plane
configuration mode. In other words, CoPP is applied using modular QoS command-line interface
(MQC), providing filtering and rate-limiting for control plane packets. Those devices with route
processors on line card modules can be protected by distributed CoPP or control plane
configuration mode on the particular slot number. The four steps to configure CoPP are as follows:

Step 1 Define a packet classification criteria. (Use MQC class-map.)
Step 2 Define a service policy. (Use MQC policy-map.)
Step 3 Enter control plane configuration mode.
Step 4 Apply a QoS policy.
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Deploying End-to-End QoS 193

Example 6-2 shows a configuration that allows two trusted hosts with source addresses 10.1.1.1
and 10.1.1.2 to forward Telnet packets to the control plane without constraint, while policing all
remaining Telnet packets to the control plane at the specified rate. The access list matches all
Telnet traffic except that from hosts 10.1.1.1 and 10.1.1.2. The class map telnet-class is defined for
all traffic matching access list 100. The policy map telnet-policy applies the police command to
the traffic matching class telnet-class. Finally, the telnet-policy is applied to the control plane using
the service-policy command. The QoS policy shown in Example 6-2 is applied for aggregate CP
services to all packets that are entering the control plane from all line cards in the router.

Example 6-2 CoPP Example: QoS Policy Applied for Aggregate CP Services
!
class-map telnet-class
match access-group 100
!
policy-map telnet-policy
class telnet-class
police 80000 conform transmit exceed drop
!
control-plane
service-policy input telnet-policy
!
access-list 100 deny tcp host 10.1.1.1 any eq telnet
access-list 100 deny tcp host 10.1.1.2 any eq telnet
access-list 100 permit tcp any any eq telnet
!

Example 6-3 shows a similar example, but for distributed CP services, allowing two trusted hosts
with source addresses 10.1.1.1 and 10.1.1.2 to forward Telnet packets to the control plane without
constraint, while policing all remaining Telnet packets that enter through slot 1 at the specified rate.

Example 6-3 CoPP Example: QoS Policy Applied for Distributed CP Services
!
class-map telnet-class
match access-group 100
!
policy-map telnet-policy
class telnet-class
police 80000 conform transmit exceed drop
!
control-plane slot 1
service-policy input telnet-policy
!
access-list 100 deny tcp host 10.1.1.1 any eq telnet
access-list 100 deny tcp host 10.1.1.2 any eq telnet
access-list 100 permit tcp any any eq telnet
!
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194 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

A virtual private network (VPN) carries private traffic over a public network. It is established
between end systems or between networks. VPNs provide three functions:

■ Confidentiality

■ Data integrity

■ Origin authentication

Two general types of VPN exist, each with its own variations:

■ Remote access

— Client initiated
— Network access server (NAS) initiated
■ Site-to-site

— Intranet
— Extranet
The most common Layer 3 tunneling protocols are as follows:

■ Generic routing encapsulation (GRE)

■ Internet Protocol security (IPsec)

QoS pre-classify is a Cisco IOS feature that allows packets to be classified before tunneling and
encryption occur. The need to classify traffic within a traffic tunnel is growing side by side with
the growth in VPN popularity.
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Foundation Summary 195

The QoS pre-classify feature provides access to the original (encapsulated) IP packet header fields.
If IP QoS classification needs access to those fields, using the QoS pre-classify feature will be
necessary. You can use one of three approaches to apply the QoS policy and the QoS pre-classify
features:

■ To classify packets based on the pre-tunnel header, apply the QoS policy to the tunnel
interface without QoS pre-classify.

■ To classify packets based on the post-tunnel header, apply the QoS policy to the physical
interface without QoS pre-classify.

■ To classify packets based on the pre-tunnel header, apply the QoS policy to the physical
interface and enable QoS pre-classify.

With the growth of multimedia and real-time applications such as IP Telephony, conferencing, and
e-learning, QoS service level agreements (SLAs) have become more important than before. The
QoS SLAs provide contractual assurance for parameters such as availability, throughput, delay,
jitter, and packet loss.

To meet the QoS requirements of customer applications, the service provider and customer both
must implement proper QoS mechanisms. This means that you must implement QoS in the
customer campus, at the customer WAN edge device outbound toward the provider network, on
the PE device inbound from the customer edge, on the provider network, on the PE device at the
remote site outbound toward the remote customer site, and on the remote customer site (remote
campus). Generally, deploying end-to-end QoS is specified to be necessary at three locations:

■ On campus

■ On the WAN edge (CE/PE)

■ On the service provider cloud

Table 6-3 provides a short list of important QoS-related tasks that might be necessary at different
locations on the customer and provider premises. Implementing these and possibly other tasks on
both the customer and provider devices supports the effort to provide end-to-end QoS.
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Table 6-3 Necessary QoS Tasks (at Different Spots) for End-to-End QoS

Campus Service
Campus Access Distribution WAN Edge Provider Cloud

Speed and duplex settings Layer 3 policing and SLA definitions Capacity planning
marking

Classification and trust Multiple queues on Classification and DiffServ
settings switch ports marking implementation (PHB)

Phone and access switch Priority queuing for Low-latency queuing Low-latency queuing
configurations VoIP

Multiple queues on switch WRED within data Link fragmentation Modified deficit round
ports queues for congestion and interleaving robin
avoidance

Priority queuing for VoIP - WRED and traffic WRED
shaping

Following are the general guidelines for enterprise campus QoS implementations:

■ Implement multiple queues on all interfaces to prevent transmit queue congestion and packet
drops.

■ Assign voice (and video) traffic to the highest priority queue.

■ Establish proper trust boundaries. (Trust the Cisco IP phone CoS setting, not the PCs.)

■ Classify and mark traffic as close to the source as possible.

■ Use class-based policing to rate-limit certain unwanted excess traffic.

■ Try to perform QoS in hardware rather than software when possible.

WAN edge QoS implementation has two facets: features applied to the traffic leaving the
enterprise network (toward the provider network), and features applied to the traffic leaving the
provider network toward the customer network. In both cases, note whether the CE device is
provider managed.

For the traffic leaving the enterprise network (via CE) moving toward the provider network, if the
CE device is provider managed, these are the QoS requirements:

■ The service provider controls output of the QoS policy on the CE.

■ The service provider should enforce SLA using the output QoS policy on the CE.
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Foundation Summary 197

■ The output policy uses queuing, dropping, and shaping.

■ Elaborate traffic classification or mapping of existing markings takes place on the CE.

■ Link fragmentation and interleaving and compressed RTP might be necessary.

If the CE device is not provider managed, these are the QoS requirements:

■ The service provider controls output of the QoS policy on the CE.

■ The service provider should enforce SLA using the input QoS policy on the PE.

■ The input policy should use policing and marking.

■ Elaborate traffic classification or mapping of existing markings takes place on the PE.

For traffic leaving the service provider network (via PE) moving toward the enterprise customer
network, these are the QoS requirements (for both managed and unmanaged-CE cases):

■ The service provider should enforce SLA using the output QoS policy on the PE.

■ The output policy should use queuing, dropping, and optionally shaping.

■ LFI or cRTP might be required.

■ The input QoS policy on the CE is not implemented if the CE is provider managed, but the
customer might implement certain QoS features on the customer-managed CE to meet the
end-to-end QoS budgets.

The functional planes of Cisco routers are data plane, management plane, control plane, and
service plane. Control plane policing (CoPP) is a Cisco IOS feature that allows you to build QoS
filters to manage the flow of control plane packets to protect the control plane against DoS attacks.
Because infrastructure attacks are becoming increasingly common, protecting the control plane of
the routers and switches using CoPP against reconnaissance and DoS attacks is crucial. The four
steps required to deploy CoPP (using MQC) are as follows:

Step 1 Define the packet classification criteria.
Step 2 Define a service policy.
Step 3 Enter the control plane configuration mode.
Step 4 Apply a QoS policy.
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198 Chapter 6: Implementing QoS Pre-Classify and Deploying End-to-End QoS

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. Provide a definition for VPN.
2. What types of interfaces is QoS pre-classify designed for?
3. What Cisco IOS command enables QoS pre-classify on an interface?
4. What are the QoS pre-classification deployment options?
5. Provide a definition for QoS SLA.
6. What are the typical maximum end-to-end (one-way) QoS SLA requirements (delay, jitter,
loss) for voice?
7. Provide at least two of the guidelines for implementing QoS in campus networks.
8. Provide at least two QoS policies to be implemented on campus network access or distributed
switches.
9. Provide a definition for CoPP.
10. How is CoPP deployed?
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This chapter covers the
following subjects:

■ Introducing AutoQoS

■ Implementing and Verifying AutoQoS

■ AutoQoS Shortcomings and Remedies
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CHAPTER 7
Implementing AutoQoS

This chapter is focused on Cisco AutoQoS. AutoQoS is a QoS deployment automation tool that
is suitable for midsize enterprise networks. It has evolved from its limited Voice over IP (VoIP)-
focused version to an enterprise version (for Cisco routers) with protocol discovery and more
general and sophisticated configuration results. This chapter provides a description for AutoQoS
and its benefits followed by a lesson on implementing AutoQoS enterprise on routers and
AutoQoS VoIP on Cisco LAN switches. Next, the chapter discusses verifying and monitoring
results of enabling AutoQoS on Cisco devices. Finally, it presents the shortcomings of AutoQoS
along with recommendations to address and resolve those issues.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 10-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 7-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 7-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering
These Questions Questions Score
“Introducing AutoQoS” 1–4
“Implementing and Verifying AutoQoS” 5–7
“AutoQoS Shortcomings and Remedies” 8–10
Total Score (10 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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202 Chapter 7: Implementing AutoQoS

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 6 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 7–8 overall score—Begin with the “Foundation Summary” section and then follow up with
the “Q&A” section at the end of the chapter.

■ 9 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Which of the following is not a key benefit of Cisco AutoQoS?
a. It automates and simplifies QoS deployment and provisioning.
b. AutoQoS results are maintenance free and can’t be tuned.
c. It reduces configuration errors.
d. It allows customers to retain complete control over their QoS configuration.
2. Which of the following statements is true about the evolution of Cisco AutoQoS?
a. Cisco AutoQoS has evolved from AutoQoS VoIP to AutoQoS for Enterprise. AutoQoS
for Enterprise extends the AutoQoS capabilities beyond VoIP, but it is only supported on
Catalyst Switches.
b. Cisco AutoQoS has evolved from AutoQoS VoIP to AutoQoS for Enterprise. AutoQoS
for Enterprise extends the AutoQoS capabilities beyond VoIP, and it has an autodiscov-
ery step.
c. Cisco AutoQoS has evolved from basic AutoQoS to AutoQoS VoIP. AutoQoS VoIP
extends the AutoQoS capabilities to support Voice over IP.
d. Cisco AutoQoS has evolved from basic AutoQoS to AutoQoS VoIP. AutoQoS VoIP
extends the AutoQoS capabilities to support Voice over IP, but it is supported only on
Cisco routers.
3. Which of the following is not one of the five key elements of QoS deployment?
a. Intrusion detection
b. Application classification and policy generation
c. Configuration and monitoring (reporting)
d. Consistency
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“Do I Know This Already?” Quiz 203

4. Which of the following is not true about NBAR protocol discovery?
a. NBAR protocol discovery is able to identify and classify static port applications.
b. NBAR protocol discovery is able to identify and classify dynamic port applications.
c. NBAR protocol discovery is able to identify and classify HTTP applications based on
URL, MIME type, or host name.
d. NBAR protocol discovery is able to identify and classify IP applications only.
5. Which of the following is a Cisco AutoQoS router configuration prerequisite?
a. No QoS policy (service) policy can be applied to the interface.
b. CEF must be enabled on the interface.
c. Correct bandwidth must be configured on the interface.
d. All of the above.
6. In deploying Cisco AutoQoS for Enterprise on routers, what is Step 1 (or Phase 1) of the two-
step (2-phase) approach?
a. Profiling the data using autodiscovery
b. Assigning the appropriate bandwidth and scheduling parameters
c. Mapping applications to their corresponding DiffServ classes
d. Enabling CEF
7. Which of the following is not a Cisco LAN switch AutoQoS verification command?
a. show auto qos
b. show auto qos interface interface
c. show auto discovery qos
d. show mls qos maps [cos-dscp | dscp-cos]
8. Which of the following is not one of the three most common Cisco AutoQoS issues that can
arise?
a. Too many traffic classes are generated; classification is overengineered.
b. The configuration that AutoQoS generates does not adapt to changing network traffic
conditions automatically.
c. The configuration that AutoQoS generates is not modifiable.
d. The configuration that AutoQoS generates fits common network scenarios but does not
fit some circumstances, even after extensive autodiscovery.
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204 Chapter 7: Implementing AutoQoS

9. Which of the following is a possible way to tune and modify the class maps or policy maps
that AutoQoS generates?
a. Do it directly at the router command-line interface (CLI) using MQC
b. Use Cisco QoS Policy Manager (QPM)
c. Copy the class maps or policy maps into a text editor, and modify the configuration
offline
d. All of the above
10. Besides NBAR and ACLs, which of the MQC classification options can you use to tune an
AutoQoS-generated configuration?
a. match input interface, match ip dscp, match ip precedence, match ip cos
b. match input interface, match ip dscp, match ip precedence, match ip rtp
c. match ip dscp, match ip precedence, match ip rtp, match ip cos
d. match input interface, match ip dscp, match ip cos, match ip rtp
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Introducing AutoQoS 205

Foundation Topics

Introducing AutoQoS
With the growth of bandwidth requirements by today’s applications and convergence of voice,
video, and data applications over common IP infrastructures (networks), deploying QoS tech-
nologies and services is a necessity within modern networks. Although you must manage delay,
jitter, available bandwidth, and packet loss, the solution must remain scalable and manageable
with respect to both simplicity and cost. Following are some of the challenges that enterprises
face:

■ The voice quality of IP Telephony applications must be high.

■ The required bandwidth for mission-critical applications must be guaranteed.

■ QoS must be simple enough to reduce errors, the deployment period, and costs.

Cisco AutoQoS is a QoS deployment automation tool that is suitable for midsize enterprises and
branches. Following are the main benefits of Cisco AutoQoS:

■ The built-in intelligence of AutoQoS makes its auto-generated configuration code suitable for
most common enterprise QoS requirements.

■ AutoQoS protects mission-critical applications against otherwise less-important applications,
providing guaranteed resources and preferential treatments.

■ Using AutoQoS does not require in-depth knowledge of QoS, Cisco IOS commands, or the
varied networking technologies involved.

■ AutoQoS-generated configurations are based on modular QoS command-line interface
(MQC) and follow the Cisco recommendations for best practices and the DiffServ model.

■ You can examine the results of AutoQoS-generated commands, and modify them if necessary,
to suit each particular need.

The first phase or release of AutoQoS, referred to as AutoQoS VoIP, was developed to automate
generation of QoS configurations for those who had or planned to deploy IP Telephony in their
enterprise but lacked the expertise to do so properly. AutoQoS VoIP operates both on Cisco routers
and Catalyst switches. It generates the required access lists, class maps, policy maps, interface
configurations, and so on to provide adequate configuration supporting IP Telephony applications.
AutoQoS VoIP uses Network Based Application Recognition (NBAR) for classification and
marking of packet DiffServ Codepoint (DSCP) fields. It can also trust markings of the packets and
not re-mark them.
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The second phase or release of AutoQoS, referred to as AutoQoS for Enterprise (or AutoQoS
Enterprise for brevity), is available only for routers. AutoQoS Enterprise has added capabilities for
voice, video, and data, plus another feature called protocol discovery. AutoQoS Enterprise has two
deployment stages:

1. Discovering types and volumes of traffic types using NBAR protocol discovery and
generating appropriate policies accordingly
2. Implementing the generated policies
You can review the application types discovered during the auto-discovery stage and the QoS
policies generated (suggested) by AutoQoS Enterprise first. After that review, you can implement
the AutoQoS-generated policies completely, modify them, or not implement them at all. However,
it is noteworthy that AutoQoS Enterprise addresses all of the following five key elements of QoS
deployment:

■ Application classification—Utilizing NBAR, AutoQoS Enterprise can perform intelligent
classification based on deep packet inspection; using CDP (version 2), an IP phone is
recognized as an attached device whose packets will be classified accordingly.

■ Policy generation—AutoQoS Enterprise generates policies based on device and interface
settings and the traffic observed in the discovery stage. These policies can be tuned further if
desired. For example, on WAN interfaces, auto-generated policies take into account the need
for techniques such as fragmentation and compression.

■ Configuration—AutoQoS Enterprise is easily enabled on router interfaces. It automates
detection of connected IP phones, which in turn affects the QoS configuration of the interface.

■ Monitoring and reporting—AutoQoS can automate generation of alerts, SNMP traps,
system loggings, and summary reports. You can use QPM to monitor, view, and evaluate the
statistics and the information (QoS feedback) gathered.

■ Consistency—AutoQoS generates consistent policies and configurations on the Cisco
devices on which it is deployed. A user can inspect generated policies, filters, and so on, plus
the gathered statistics from the discovery stage.

The discovery stage of AutoQoS Enterprise uses NBAR protocol discovery. NBAR protocol
discovery first collects and analyzes packets that are going through the interface of a router; then
it generates statistics on the types and numbers of the packets processed. All traffic types that
NBAR supports (close to 100 applications and protocols) that go through an interface in either
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Implementing and Verifying AutoQoS 207

direction (input or output) are discovered and analyzed in real-time. The statistics reported per-
interface and per-protocol include 5-minute bit rates (bps), packet counts, and byte counts. NBAR
protocol discovery can identify and classify all of the following application types:

■ Applications that target a session to a well-known (UDP/TCP) destination port number,
referred to as static port applications

■ Applications that start a control session using a well-known port number but negotiate another
port number for the session, referred to as dynamic port applications

■ Some non-IP applications

■ HTTP applications based on URL, MIME type, or host name

Implementing and Verifying AutoQoS
Before you implement AutoQoS and enable it on router interfaces, it is useful to know the router
AutoQoS deployment restrictions. Some design considerations are also worth learning with regard
to deploying AutoQoS on routers. Finally, you must know the prerequisites for configuring
AutoQoS on Cisco routers.

You can enable Cisco AutoQoS Enterprise on certain types of interfaces and permanent virtual
circuits (PVCs) only. These are the interface and PVC types on which you can enable AutoQoS
enterprise for a Cisco router:

■ Serial interfaces with PPP or high-level data link control (HDLC) encapsulation.

■ Frame Relay point-to-point subinterfaces. (Multipoint is not supported.)

■ ATM point-to-point subinterfaces (PVCs) on both slow (<=768 kbps) and fast serial (>768
kbps) interfaces.

■ Frame Relay-to-ATM interworking links.

On low-speed serial links, you must enable AutoQoS Enterprise on both ends, and the configured
bandwidths must be consistent. For PPP encapsulations, Multilink PPP (MLP) is enabled auto-
matically, and the IP address of the serial interface is removed and put on the virtual template
(MLP bundle). Frame Relay data-link connection identifier (DLCI) and ATM PVCs have some
similar restrictions with respect to enabling AutoQoS. Table 7-2 shows those restrictions side by
side.
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Table 7-2 AutoQoS Restrictions on Frame Relay DLCIs and ATM PVCs

Frame Relay DLCI Restrictions ATM PVC Restrictions
You cannot configure AutoQoS on a Frame —
Relay DLCI if a map class is attached to the
DLCI.
You cannot configure AutoQoS on a low-speed You cannot configure AutoQoS on a low-speed
Frame Relay DLCI if a virtual template is PVC if a virtual template is already configured
already configured for the DLCI. for the ATM PVC.
For low-speed Frame Relay DLCI configured For low-speed ATM PVCs, MLP over ATM is
with Frame Relay-to-ATM interworking, MLP configured automatically; the subinterface
over Frame Relay is configured automatically; must have an IP address.
the subinterface must have an IP address.
When MLP over Frame Relay is configured, the When MLP over ATM is configured, the IP
IP address is removed and placed on the MLP address is removed and put on the MLP bundle.
bundle.

Based on the interface type, interface bandwidth, and encapsulation, AutoQoS might enable
different features on the router interfaces. The bandwidth that is configured on a router interface
at the time AutoQoS is enabled on that interface plays a significant role toward the configuration
that AutoQoS generates for that interface. AutoQoS will not respond and alter the generated
configuration if the interface bandwidth is changed afterward. Following are examples of the
features that AutoQoS can enable on router interfaces:

■ LLQ—Low-latency queuing reserves a priority queue for VoIP (RTP) traffic, providing a
guaranteed but policed bandwidth. Other queues (class-based weighted fair queueing, or
CBWFQ) serve traffic such as mission-critical, transactional, and signaling traffic and give
them bandwidth guarantees.

■ cRTP—Compressed RTP reduces the IP/UDP/RTP header from 40 bytes to 2/4 bytes
(without/with CRC). This feature is enabled on low-speed serial interfaces. The reduced
header overhead significantly improves link efficiency.

■ LFI—The link fragmentation and interleaving feature fragments large data packets on slow
interfaces. Therefore, on shared outbound queues (such as the hardware queues) of slow
interfaces, VoIP packets are not stuck behind large packets, experiencing jitter and long
delays.

On Frame Relay interfaces, fragmentation is configured based on the assumption that the G.729
codec is used and that a voice packet should not experience more than 10-ms delay due to being
stuck behind another packet on the hardware queue. If G.729 is the assumed codec, an IP packet
that is encapsulating two 10-ms digitized voice samples ends up being 60 bytes long ((8000 bps ×
20/1000 sec / 8)+ 40 bytes). Because a VoIP packet should not be fragmented, the minimum
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Implementing and Verifying AutoQoS 209

fragment size is set to 60 bytes. The maximum fragment size, on the other hand, is calculated
based on the 10-ms maximum delay and the configured bandwidth. For example, if the bandwidth
is configured as 64 kbps, the maximum fragment size is calculated as 80 bytes (64,000 bps × 10/
1000 sec / 8 bits/byte). If a codec such as G.711 is used instead of G.729, or if the bandwidth of
an interface changes, you might have to modify the fragmentation size or disable and enable
AutoQoS so that the changes affect the generated configuration.

Some prerequisites must be satisfied before AutoQoS is enabled on a router. Those prerequisites
are as follows:

■ You must enable Cisco Express Forwarding (CEF) on the interface where AutoQoS is
intended to be enabled, because AutoQoS relies on NBAR for discovery, and NBAR needs
CEF.

■ You cannot apply a QoS policy (service policy) to the interface prior to enabling AutoQoS on
that interface.

■ On all interfaces and subinterfaces, you must properly configure the bandwidth. On slow
interfaces or subinterfaces, you must configure an IP address. AutoQoS enables MLP on slow
interfaces and moves the IP address to the MLP virtual template.

■ For AutoQoS SNMP traps to work, you must enable SNMP on the router and specify the
server address for SNMP traps destination. This address must be reachable from the router.
The SNMP community string “AutoQoS” must have write permission.

Two-Step Deployment of AutoQoS Enterprise on Routers
Deploying AutoQoS for the Enterprise on Cisco routers is a two-step (or two-phase) process. Step
1 is the auto-discovery step. Step 2 is generation and deployment of MQC-based QoS policies
based on the discovery step.

AutoQoS discovery uses NBAR protocol discovery. The type and volume of traffic on the network
is discovered and analyzed in real-time to be able to generate realistic policies in Step 2. Generally
speaking, the longer the auto-discovery runs, the more accurate the results will be. The default
period is 3 days, but the administrator can certainly decide if 3 days is sufficient, or if it is too long
or too short. Depending on the variety of applications and how often they run (once a day, week,
or month), the length of time for running auto-discovery must be determined. The auto-discovery
step is/was missing in AutoQoS VoIP; it goes straight to policy generation.

In Step 2, AutoQoS for Enterprise uses the results from the auto-discovery step to generate
templates and install them on router interface(s). Templates are the basis for generation of MQC
class maps and policy maps. After the policy maps are generated, AutoQoS applies them to the
intended interfaces (using service-policy).
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In Step 1, auto-discovery is enabled from the interface configuration mode by entering the auto
discovery qos command:

Router(config-if)# auto discovery qos [t
trust ]

The optional keyword trust allows packets to be classified and receive appropriate QoS treatments
based on preset QoS (DSCP) markings. Without the optional trust keyword, packets are classified
based on the NBAR classification scheme, not based on the preset packet markings. The interface
in which you enable auto-discovery must have CEF enabled and have its bandwidth configured
(serial interface). Also, if it is a slow interface (<=768 kbps), it must have an IP address. The auto
discovery qos command is not supported on a subinterface, and it is not supported on an interface
that has a policy attached to it already. The configured bandwidth of the interface should not be
changed after the command has been applied. Typing no auto discovery qos will stop auto-
discovery (data collection), and it will remove any reports that have been generated. If you want
to view the auto-discovery results, even before they are completed, you can do so by typing this
command:

Router# show auto discovery qos

The second step of implementing AutoQoS for Enterprise is enabling AutoQoS on the interface
upon completion of the discovery step. The auto qos command causes generation of QoS templates
that are used to create MQC class maps and policy maps and application of the policies to the
interface. This command is also entered from the interface configuration mode:

Router(config-if)# auto qos [v
voip [t
trust] [f
fr-atm]]

You use the keyword voip if you are enabling AutoQoS VoIP rather than AutoQoS Enterprise.
On an earlier Cisco IOS release, this might be your only option. AutoQos for Enterprise was
introduced in IOS release 12.3(7)T. Remember that AutoQoS VoIP does not require the discovery
step. The trust keyword, again, is about respecting the preset marking of the entering packets
rather than ignoring their marking and classifying them using NBAR. The keyword fr-atm
enables AutoQoS VoIP for Frame Relay-to-ATM interworking.

Deploying AutoQoS VoIP on IOS-Based Catalyst Switches
Cisco Catalyst (LAN) switches support only AutoQoS VoIP. Catalyst switches with Cisco IOS are
configured differently from Catalyst switches with Cisco Catalyst operating systems. The ONT
course and exam focus on the Cisco IOS-based configuration and commands. More specifically,
the emphasis is put on 2950(EI), 3550, and 4500 switches, with a few references made to 6500
switches. Please note that the 2950 switches require the Enhanced Image (EI) software and not the
Standard Image (SI) for AutoQoS VoIP.

To enable AutoQoS VoIP on Catalyst switches, you need to be aware of two commands. The first
one is meant for an access port where either a workstation or an IP phone is connected. The second
command you need is meant for ports that are connected to other trusted devices such as routers
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Implementing and Verifying AutoQoS 211

and switches. A trusted device is a device whose marked traffic (QoS marking) is honored by the
local device. Please note that AutoQoS VoIP support for IP softphone is only available on Catalyst
6500 switches (up to the time the Cisco ONT course was first released).

NOTE An IP phone has a built-in 3-port Ethernet switch. One port is connected to the IP
phone and is not visible from outside the IP phone case. The two other Ethernet ports are
accessible and located under the IP phone case. One port is meant to be connected to a LAN
switch and the other is for a user workstation to plug into it and get connectivity to the LAN
switch.

From the following two AutoQoS VoIP commands, the first command is for the IP phone
connections, and the other is for trusted connections to other network devices:

switch(config-if)# auto qos voip cisco-phone
switch(config-if)# auto qos voip trust

The command auto qos voip trust is applied to an interface that is assumed to be connected to a
trusted device; the interface is usually an uplink trunk connection to another switch or router. This
command also reconfigures the egress queues on the interface where it is applied.

On the interface where you apply the auto qos voip cisco-phone command, you must have CDP
version 2 enabled. Using CDP version 2, the switch determines whether a Cisco IP phone is attached
to the port. If CDP is not enabled or if it is CDP version 1, a syslog warning message is displayed.
When a Cisco IP phone is detected on a port where the auto qos voip cisco-phone command is
entered, the port classification trusts the QOS marking of the incoming packets. The command is
said to extend the trust boundary to the Cisco IP phone when it is detected. The command also
reconfigures the egress queues on the interface. The mls qos global configuration command is
automatically enabled when you enter the auto qos voip command on the first interface. The mls
qos command enables the QoS feature on the Catalyst switch.

The AutoQoS VoIP commands should not be applied to an interface where QoS commands have
previously been configured. However, after you enable AutoQoS on an interface, you can fine-tune
and modify the AutoQoS-generated configuration commands if necessary. The QoS markings of
the incoming traffic are honored (trusted) on an interface in two cases. The first case is when the
auto qos voip trust command is applied to an interface. The second case is when a Cisco IP phone
is attached to the switch port, and the auto qos voip cisco-phone is applied to the interface. If a
Cisco IP phone is disconnected from such a port and a workstation is connected to the port directly,
the switch discovers the departure of the Cisco IP phone (using CDP version 2), and it changes its
behavior to no trust on that port. The egress queuing and buffer allocation on a port are determined
automatically based on the interface type; AutoQoS VoIP generates optimal priority queuing (PQ)
and weighted round-robin (WRR) configurations for all port types, including static, dynamic
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212 Chapter 7: Implementing AutoQoS

access, voice VLAN (VVLAN), and trunk ports. As for mapping of the QoS markings, AutoQoS
VoIP maps class of service (CoS) to DSCP as it forwards traffic toward the egress interface queue.

Verifying AutoQoS on Cisco Routers and IOS-Based Catalyst Switches
Monitoring and verifying AutoQoS on routers and switches have similarities and differences.
Recall that AutoQoS Enterprise, which includes an initial protocol discovery phase, is not
supported on Catalyst switches yet. On the other hand, Cisco Catalyst switches have a unique
behavior of mapping the CoS setting of the incoming frames to DSCP, using a CoS-to-DSCP
mapping scheme; this is useful for egress interface queuing purposes.

On both Cisco routers and Cisco IOS-based Catalyst switches, the show auto qos command
displays the AutoQoS templates and initial configuration, whereas the show policy-map interface
command displays the autogenerated policies and QoS parameters for each interface. The show
auto discovery qos command, which relates to the discovery phase of the AutoQoS for enterprise
and is therefore applicable only to routers, displays autodiscovery results for you to review. On
Cisco IOS-based catalyst switches, the CoS-to-DSCP mapping can be displayed using the show
mls qos maps command. Table 7-3 shows some important QoS verification commands for routers
and Cisco IOS-based Catalyst switches.

Table 7-3 AutoQoS Verification Commands

Command Type Command Syntax
Router show auto discovery qos [interface interface]
Router show auto qos [interface interface]
Router show policy-map interface interface
Switch show auto qos [interface interface]
Switch show mls qos interface [interface | vlan vlan-id | buffers | policers |
queuing | statistics]
Switch show mls qos maps [cos-dscp | dscp-cos]

Sample (and partial) output of the router commands included in Table 7-3 is shown in Example 7-1.
As displayed, the output of the show auto discovery qos interface command on a router displays
the results of the collected data during the autodiscovery phase on the specified interface. It is
recommended that you run the discovery phase for at least three days for more accurate results.

Example 7-1 Sample Output of AutoQoS Monitoring Commands
router# show auto discovery qos interface serial 3/1.1
Serial3/1.1
AutoQoS Discovery enabled for applications
Discovery up time: 3 hours, 53 minutes
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Implementing and Verifying AutoQoS 213

Example 7-1 Sample Output of AutoQoS Monitoring Commands (Continued)
AutoQoS Class information:
Class Voice:
Recommended Minimum Bandwidth: 512 Kbps/50% (PeakRate).
Detected applications and data:
Application/ AverageRate PeakRate Total
Protocol (kbps/%) (kbps/%) (bytes)
----------- ----------- -------- --------
rtp audio 3/<1 512/50 841512

router# show auto qos
!
policy-map AutoQoS-Policy-Se3/1.1
class AutoQoS-Voice-Se3/1.1
priority percent 70
set dscp ef
class AutoQoS-Inter-Video-Se3/1.1
bandwidth remaining percent 10
set dscp af41
class AutoQoS-Stream-Video-Se3/1.1
bandwidth remaining percent 5
set dscp cs4
class AutoQoS-Transactional-Se3/1.1
bandwidth remaining percent 5

router# show policy-map interface FastEthernet0/0.1
FastEthernet0/0.1
Service-policy output: voice_traffic
Class-map: dscp46 (match-any)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: ip dscp 46
0 packets, 0 bytes
5 minute rate 0 bps
Traffic Shaping
Target Byte Sustain Excess Interval Increment Adapt
Rate Limit bits/int bits/int (ms) (bytes) Active
2500 10000 10000 333 1250 -
...

As shown on the sample output display, the show auto qos interface command on a router
displays the MQC-based policy maps, class maps, and access lists generated by AutoQoS for the
specified interface. The show policy-map interface command displays the packet statistics for all
classes that are configured and referenced by the policy that is applied to the specified interface.
Please note that for brevity, only small portions of the outputs are displayed.
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214 Chapter 7: Implementing AutoQoS

In Example 7-2, you can see sample (and partial) output of the switch commands included in Table
7-3. The show auto qos command on a Catalyst switch displays the commands that the AutoQoS
VoIP has initially generated for the switch (prior to any modifications that might have been
applied). The sample output shows that 20 percent of the bandwidth is allocated to queue 1, 1
percent to queue 2, and 80 percent to queue 3. Because a value of 0 percent is assigned to queue
number 4, this queue is the designated priority queue. CoS values of 0, 1, 2, and 4 are directed to
queue 1, whereas CoS values 3, 6, and 7 are mapped to queue 3. CoS value 5 is mapped to queue
4. Queue 2 is not used at all. Finally, the CoS-to-DSCP mappings are shown (CoS 0 to DSCP 0,
CoS 1 to DSCP 8, and so on).

Example 7-2 Sample (and Partial) Output of the Switch Commands Included in Table 7-3
switch# show auto qos
Initial configuration applied by AutoQoS:
wrr-queue bandwidth 20 1 80 0
no wrr-queue cos-map
wrr-queue cos 1 0 1 2 4
wrr-queue cos 3 3 6 7
wrr-queue cos 4 5
mls qos map cos-dscp 0 8 16 26 32 46 48 56
!
interface FastEthernet0/3
mls qos trust device cisco-phone
mls qos trust cos

switch# show mls qos interface gigabitethernet0/1 statistics
Ingress
dscp: incoming no_change classified policed dropped (in bytes)
1 : 0 0 0 0 0
Others: 203216935 24234242 178982693 0 0

Egress
dscp: incoming no_change classified policed dropped (in bytes)
1 : 0 n/a n/a 0 0

WRED drop counts:
qid thresh1 thresh2 FreeQ
1 : 0 0 1024
2 : 0 0 1024

switch# show mls qos maps dscp-cos
Dscp-cos map:
dscp: 0 8 10 16 18 24 26 32 34 40 46 48 56
-----------------------------------------------
cos: 0 1 1 2 2 3 7 4 4 5 5 7 7
...
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AutoQoS Shortcomings and Remedies 215

The output of the show mls qos interface interface command has various optional keywords
available. A sample output in which the statistics keyword is used is shown in Example 7-2. The
output of the show mls qos maps dscp-cos is shown last; it is obvious that the output displays the
way DSCP is mapped to the CoS value for the egress packets. Please note that you can modify the
default CoS-to-DSCP and DSCP-to-CoS mappings using the global configuration mode mls qos
map command.

AutoQoS Shortcomings and Remedies
The policy maps and class maps that AutoQoS generates do not always suit the needs of a network
completely. In that case, you can modify the policy maps and class maps to meet the specific
network requirements. Therefore, it is important to know how to fine-tune the configuration that
Cisco AutoQoS generates. Some Cisco IOS show commands are specifically helpful for
determining which parts of the configuration need modification.

Automation with Cisco AutoQoS
Cisco AutoQoS is capable of performing the following tasks and might generate appropriate
configurations to accomplish them:

■ Defining the trust boundaries (or extended trust boundaries) and re-marking incoming traffic
on trusted and untrusted links

■ Defining traffic classes based on the applications and protocols discovered in the network

■ Creating queuing mechanisms with proper configurations such as bandwidth guarantee for
each traffic type, based on the DiffServ model

■ Enabling interface-specific transport features, such as LFI, Multilink PPP (MLP), cRTP, TCP
Header compression, traffic shaping, and Frame Relay traffic shaping (FRTS), when
necessary based on link bandwidth and encapsulation

■ Defining alarms and event logging settings for monitoring purposes

■ Defining CoS-to-DSCP mappings (or other required mappings), DSCP-to-egress queue
mappings, and the proper queue sizes and WRR weights on Cisco Catalyst LAN switches

Based on Cisco best-practices recommendations and the discovered application and protocol
types, AutoQoS can enable six QoS mechanisms using DiffServ technology. Table 7-4 shows the
six DiffServ functions and the corresponding Cisco IOS features that AutoQoS can enable for that
function.
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216 Chapter 7: Implementing AutoQoS

Table 7-4 DiffServ Functions and Cisco IOS Features That AutoQoS Enables

DiffServ Function Cisco IOS QoS Feature That AutoQoS Uses
Classification Using NBAR (on untrusted links)

Using IP precedence, DSCP, or CoS (trusted)
Marking Class-based marking
Congestion LLQ (Strict PQ + CBWFQ) using percentage BW
management
WRR (on Catalyst LAN switches)
Shaping CBTS1

FRTS2
Congestion avoidance WRED3
Link efficiency LFI

MLP

cRTP

1 CBTS = class-based traffic shaping
2 FRTS = Frame Relay traffic shaping
3 WRED = weighted random early detection

Using MQC, AutoQoS defines up to 10 traffic classes based on packet marking on trusted links or
using NBAR on untrusted links. Classified packets are marked at trust boundary spots (as close to
the traffic source as possible), preferably in the wiring closet switches and IP phones. Table 7-5
shows the ten classes of traffic that AutoQoS can define along with the DSCP and CoS values that
AutoQoS assigns to them. The number of traffic classes defined depends on the results of the
discovery phase.

Table 7-5 Traffic Classes That AutoQoS Defines

CoS
Class Name Traffic Type DSCP Value Value
IP Routing Network control traffic such as routing CS6 6
protocols
Interactive Voice Interactive voice bearer traffic EF 5
Interactive Video Interactive video data traffic AF41 4
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AutoQoS Shortcomings and Remedies 217

Table 7-5 Traffic Classes That AutoQoS Defines (Continued)

CoS
Class Name Traffic Type DSCP Value Value
Streaming Video Streaming media traffic CS4 4
Telephony Signaling Telephony signaling and control traffic CS3 3
Transactional and Database applications that are AF21 2
Interactive transactional in nature
Network Network management traffic CS2 2
Management
Bulk Data Bulk data transfers, web traffic, general AF11 1
data service
Scavenger Entertainment, rogue traffic, and less CS1 1
than best-effort traffic
Best Effort All noncritical and miscellaneous traffic BE 0

To ensure predictable network behavior and good voice (and video) quality while providing the
appropriate amount of bandwidth to Enterprise applications, especially during congestion,
AutoQoS enables the most modern queuing mechanisms—LLQ and WRR—where they are
needed. Voice traffic is treated as DiffServ EF with highest priority and placed in a strict priority
queue with a guaranteed but policed bandwidth. Signaling and enterprise data traffic are treated as
DiffServ AF classes, and CBWFQ is utilized for those classes, giving each class a separate queue
with minimum bandwidth guarantees. Unclassified traffic is treated as DiffServ BE and is assigned
to the default class. The bandwidth allocations are done using a percentage of the link bandwidth
for better scalability and manageability reasons. On LAN switches, WRR is utilized with a priority
queue for real-time traffic. Also, AutoQoS uses modifiable CoS-to-DSCP and DSCP-to-CoS
mappings within Cisco LAN switches.

AutoQoS enables FRTS where it is needed. FRTS is especially important for two reasons:

■ The interface clock rate (physical speed) is usually higher than the committed information
rate (CIR). As stated before, correct bandwidth configuration on serial interfaces and sub-
interfaces is necessary before activation of AutoQos on those interfaces.

■ Enterprise sites are usually connected in a hub-and-spoke topology, and traffic flows from one
or many sites to another site can cause congestion and data loss at the destination site.

WRED is the congestion avoidance technique that AutoQoS deploys to avoid tail drop and
congestion at network bottleneck areas. Global synchronization and dropping of high-priority
packets are the mitigation targets of congestion avoidance using WRED. AutoQoS deploys link-
efficiency mechanisms to address insufficient bandwidth and long delays on slow links. The link-
efficiency mechanisms that AutoQoS deploys include LFI, MLP, Frame Relay fragmentation, and
cRTP.
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218 Chapter 7: Implementing AutoQoS

Common AutoQoS Problems
AutoQoS was developed to automate QoS configuration for common enterprise network
scenarios. Therefore, the configuration that AutoQoS yields does not necessarily suit and satisfy
the requirements of every network. Following are the three most common Cisco AutoQoS issues
that might arise:

■ Too many traffic classes are generated; classification is overengineered.

■ The configuration that AutoQoS generates does not adapt automatically to changing network
traffic conditions.

■ The configuration that AutoQoS generates fits common network scenarios but does not fit
some circumstances, even after extensive autodiscovery.

Based on the traffic and protocol types discovered during the autodiscovery phase, AutoQoS
can generate up to ten traffic classes. Most enterprises, to keep the configurations simple and
manageable, deploy only three to six traffic classes. Currently, AutoQoS does not have a knob to
let you configure the maximum number of classes to be generated. However, it is recommended
that if the number of generated traffic classes is too many for your needs, you should modify the
AutoQoS-generated configuration and reduce the number of traffic classes. You can consolidate
two or more similar traffic classes into a common class.

AutoQoS generates QoS templates and policies based on the device configuration at the time
AutoQoS was enabled and based on the network applications and protocols detected at the time
autodiscovery was run. Therefore, it is recommended that configurations such as interface band-
width be done carefully, and before the AutoQoS discovery is allowed to run for as long as possible
(preferably several days). If the device configuration changes, or if network traffic conditions
change, AutoQoS-generated configuration will not adapt to the changes. However, if you disable
AutoQoS, rerun the AutoQoS discovery, and enable AutoQoS again, the AutoQoS will generate
its templates and policies based on the new network conditions.

If AutoQoS-generated configuration does not suit your network needs and circumstances, you
might have to give the autodiscovery phase more time for a more thorough discovery and
classification. However, letting the autodiscovery run for a long time does not always solve this
problem. This is because the AutoQoS was developed for most common Enterprise networks and
based on Cisco best-practice recommendations, but it does not necessarily meet the special
requirements of all networks. To solve this problem, you can modify the configuration that
AutoQoS generates. The AutoQoS-generated configuration is MQC compliant, and you can use
MQC to enhance the configuration to meet your specific needs.
1763fm.book Page 219 Monday, April 23, 2007 8:58 AM

AutoQoS Shortcomings and Remedies 219

Interpreting and Modifying AutoQoS Configurations
The show auto qos command displays all the QoS mechanisms (and the corresponding
configurations) that Cisco AutoQoS has enabled on a router, with or without autodiscovery.
Therefore, you can inspect all the QoS templates that were generated as a result of applying Cisco
AutoQoS. You can gather several particular facts from the output of the show auto qos command,
the most important of which are these:

■ The number of traffic classes.

■ The classification options used.

■ The traffic markings performed.

■ The queuing mechanisms generated and the options used.

■ Other QoS mechanisms, such as traffic shaping, applied per traffic class.

■ Other traffic parameters, such as CIR, suggested for a Frame Relay connection.

■ The interface, subinterface, or virtual circuit where the policies are applied.

The number of traffic classes that AutoQoS identifies is recognized based on the number of class
maps that have been generated. The match and set statements within each class map reveal the
classification options used and the class-based markings performed. From within the policy maps,
you can observe the queue types generated and the corresponding parameters; the priority and
bandwidth commands reveal the queue type and the amount of bandwidth guarantee for each
queue. From within the policy maps, you can also observe other QoS mechanisms, such as class-
based shaping, congestion avoidance (WRED), or link efficiency mechanisms (LFI or cRTP)
applied to each traffic class. You can discover traffic parameters such as the CIR or committed
burst applied to a Frame Relay map class—in other words, suggested by AutoQoS—by inspecting
the show auto qos command output. The output of this command also shows the actual interface,
subinterface, or virtual circuit where the policies that AutoQoS generates are applied. Finally, the
Remote Monitoring (RMON) traps that are logged for voice packet drops are displayed in the
output of the show auto qos command.

Using Cisco IOS command-line interface (CLI), you can modify the class maps, policy maps, and
traffic parameters that AutoQoS generates. You might have to do this for two major reasons:

■ The AutoQoS-generated commands do not completely satisfy the specific requirements of the
Enterprise network.

■ The network condition, policies, traffic volume and patterns, and so on might change over
time, rendering the AutoQoS-generated configuration dissatisfying.
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220 Chapter 7: Implementing AutoQoS

If the network engineers (or administrators) have the ability and the expertise to modify and adapt
the AutoQoS-generated configuration, they will not need to redeploy the whole AutoQoS procedure
again. You can modify and tune the AutoQoS-generated class maps and policy maps by doing the
following:

■ Using Cisco QoS Policy Manager (QPM).

■ Directly entering the commands one at the time at the router CLI using MQC.

■ Copying the existing configuration, a class map for example, into a text editor and modifying
the configuration using the text editor, offline. Next, using CLI, remove the old undesirable
configuration and then add the new configuration by copying and pasting the text from the
text editor. This is probably the easiest way to modify and tune the AutoQoS-generated class
maps and policy maps.

For classification purposes, in addition to using NBAR and ACLs, MQC offers more classification
options that you can use for tuning. Some of those classification options and their corresponding
match statements are as follows:

■ Based on the specific ingress interface where the traffic comes from:
match input interface interface

■ Based on the Layer 2 CoS value of the traffic:
match cos cos-value [cos-value ...]

■ Based on the Layer 3 IP precedence value:
match ip precedence ip-prec-value [ip-prec-value ...]

■ Based on the Layer 3 IP DSCP value:
match ip dscp ip-dscp-value [ip-dscp-value ...]

■ Based on the RTP port value range:
match ip rtp starting-port-number port-range

The modifying and tuning of the configuration that AutoQoS generates will probably take a few
rounds of modification and testing before it fully satisfies your requirements. Figure 7-1 shows a
flowchart about using AutoQoS, verifying its auto-generated commands, and modifying the auto-
generated commands if necessary.
1763fm.book Page 221 Monday, April 23, 2007 8:58 AM

AutoQoS Shortcomings and Remedies 221

Figure 7-1 Verifying and Modifying AutoQoS-Generated Configurations

Start AutoQoS
Discovery Let Autodiscovery Run
for Longer Period

Autodiscovery Results
Examine Autodiscovery No Do Not Meet Expectations
OK?
Results While in Progress

Modify the AutoQos-
Enable AutoQoS Yes Generated Configuration
(Generate Templates) (Manually or Using QPM)

Modify

View the Generated Modify or
No
Class Maps and OK?
Autodiscovery Results
Start Over? Start
Policy Maps Over
Do Not Meet Expectations
Yes

No
Keep the Configuration,
But Monitor Network and Traffic or
Yes
Traffic Condition Changes Network Conditions
Changed?

The procedure for modifying an existing, active classification or policy that AutoQoS generates
can be summarized into a three-step process:

Step 1 Review the existing QoS policy, identify the new requirements, and outline
the configuration modifications necessary.
Step 2 Modify the AutoQoS-generated configuration according to the new
requirements.
Step 3 Review the new (modified) configuration.
Please note that if you modify the AutoQoS-generated configuration, the AutoQoS generated
commands will not be removed properly when you enter the no auto qos command. The no auto
qos command only removes the original (unmodified) commands that AutoQoS generated.
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222 Chapter 7: Implementing AutoQoS

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should help
solidify some key facts. If you are doing your final preparation before the exam, the information in
this section is a convenient way to review the day before the exam.

Cisco AutoQoS is an automation tool for deploying QoS policies. Following are the key benefits
of Cisco AutoQoS:

■ Uses Cisco IOS built-in intelligence to automate generation of QoS configurations for most
common business scenarios

■ Protects business-critical data applications in the Enterprise to maximize their availability

■ Simplifies QoS deployment

■ Reduces configuration errors

■ Makes QoS deployment cheaper, faster, and simpler

■ Follows the DiffServ model

■ Allows customers to have complete control over their QoS configuration

■ Enables customers to modify and tune the configurations that Cisco AutoQoS automatically
generates to meet their specific needs or changes in the network conditions

The two phases of Cisco AutoQoS evolution are as follows:

1. AutoQoS VoIP
This was the first phase of AutoQoS.
One command provisions the basic required QoS commands.
It is supported across a broad range of router and switch platforms.
2. AutoQoS for Enterprise
This is the second phase of AutoQoS.
It extends the AutoQoS capabilities for data, voice, and video.
It is, however, supported only on routers.
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Foundation Summary 223

It is deployed in a two-step process. In the first step, called autodiscovery, it discovers
the traffic types and loads using NBAR protocol discovery. In the second step, it
generates and implements QoS policies.
Cisco AutoQoS addresses five key elements of QoS deployment:

■ Application classification

■ Policy generation

■ Configuration

■ Monitoring and reporting

■ Consistency

AutoQoS for Enterprise uses NBAR protocol discovery. NBAR protocol discovery analyzes traffic
in real-time, identifies approximately 100 Layer 4 through 7 applications and protocols using
stateful and deep packet inspection, and provides bidirectional, per-interface, and per-protocol
statistics. NBAR protocol discovery is able to identify and classify all of the following application
types:

■ Applications that target a session to a well-known (UDP/TCP) destination port number,
referred to as static port applications

■ Applications that start a control session using a well-known port number but negotiate another
port number for the session, referred to as dynamic port applications

■ Some non-IP applications

■ HTTP applications based on URL, MIME type, or host name

You can enable Cisco AutoQoS Enterprise on certain types of interfaces and permanent virtual
circuits (PVCs) only. These are the interface and PVC types that you can enable AutoQoS
Enterprise for on a Cisco router:

■ Serial interfaces with PPP or HDLC encapsulation.

■ Frame Relay point-to-point subinterfaces. (Multipoint is not supported.)

■ ATM point-to-point subinterfaces (PVCs) on both slow (<=768 kbps) and fast serial (>768
kbps) interfaces.

■ Frame Relay-to-ATM interworking links.
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224 Chapter 7: Implementing AutoQoS

Following are the router prerequisites for configuring Cisco AutoQoS:

■ The router cannot have a QoS policy attached to the interface.

■ You must enable CEF on the router interface (or PVC).

■ You must specify the correct bandwidth on the interface or subinterface.

■ You must configure a low-speed interface (<= 768 Kbps) and an IP address.

You deploy AutoQoS for Enterprise on Cisco routers in two steps (or two phases):

Step 1 Traffic is profiled using autodiscovery.
You do this by entering the auto qos discovery command in the interface
configuration mode.
Step 2 MQC-based QoS policies are generated and deployed.
You do this by entering the auto qos command in interface configuration
mode.
On Cisco LAN switches, AutoQoS VoIP is enabled on each interface using the auto qos voip
[trust | cisco-phone] command. The trust keyword is used for trusted connections such as an
uplink to a trusted switch or router so that the ingress VoIP packet marking is trusted. You use the
cisco-phone keyword for Cisco IP phone connections and to enable the trusted boundary feature.
You use CDP to detect the presence or absence of a Cisco IP phone.

The commands for verifying Cisco AutoQoS on routers are as follows:

■ show auto discovery qos

Allows you to examine autodiscovery results
■ show auto qos

Allows you to examine Cisco AutoQoS templates and initial configuration
■ show policy-map interface

Allows you to explore interface statistics for autogenerated policy
The commands for verifying Cisco AutoQoS on Cisco LAN switches are as follows:

■ show auto qos

Allows you to examine Cisco AutoQoS templates and initial configuration
1763fm.book Page 225 Monday, April 23, 2007 8:58 AM

Foundation Summary 225

■ show policy-map interface

Allows you to explore interface statistics for autogenerated policy
■ show mls qos maps

Allows you to examine CoS-to-DSCP maps
The three most common Cisco AutoQoS issues that might arise, and their corresponding solutions,
are as follows:

■ Too many traffic classes are generated; classification is overengineered.

Solution: Manually consolidate similar classes to produce the number of classes
needed.
■ The configuration that AutoQoS generates does not automatically adapt to changing network
traffic conditions.

Solution: Run Cisco AutoQoS discovery on a periodic basis, followed by re-
enabling of Cisco AutoQoS.
■ The configuration that AutoQoS generates fits common network scenarios but does not fit
some circumstances, even after extensive autodiscovery

Solution: Manually fine-tune the AutoQoS-generated configuration.
You examine the AutoQoS-generated configuration using the show auto qos command, which
provides the following information:

■ Number of traffic classes identified (class maps)

■ Traffic classification options selected (within class maps)

■ Traffic marking options selected (within policy maps)

■ Queuing mechanisms deployed and their corresponding parameters (within policy maps)

■ Other QoS mechanisms deployed (within policy maps)

■ Where the autogenerated policies are applied: on the interface, subinterface, or PVC

You might have to modify the configuration that AutoQoS generates for two reasons:

■ The AutoQoS-generated commands do not completely satisfy the specific requirements of the
Enterprise network.

■ The network condition, policies, traffic volume and patterns, and so on might change over
time, rendering the AutoQoS-generated configuration dissatisfying.
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226 Chapter 7: Implementing AutoQoS

You can modify and tune the AutoQoS-generated class maps and policy maps by doing the
following:

■ Using Cisco QoS Policy Manager (QPM).

■ Directly entering the commands one at a time at the router command-line interface using
MQC.

■ Copying the existing configuration, a class map for example, into a text editor and modifying
the configuration using the text editor offline. Next, using CLI, remove the old undesirable
configuration and then add the new configuration by copying and pasting the text from the
text editor. This is probably the easiest way.

For classification purposes, in addition to using NBAR and ACLs, you can use the following
classification options that MQC offers for tuning:

■ Based on the specific ingress interface where the traffic comes from:
match input interface interface

■ Based on the Layer 2 CoS value of the traffic:
match cos cos-value [cos-value ...]

■ Based on the Layer 3 IP precedence value:
match ip precedence ip-prec-value [ip-prec-value ...]

■ Based on the Layer 3 IP DSCP value:
match ip dscp ip-dscp-value [ip-dscp-value ...]

■ Based on the RTP port value range:
match ip rtp starting-port-number port-range
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Q&A 227

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. List at least three key benefits of Cisco AutoQoS.
2. What are the two phases of AutoQoS evolution?
3. What are the five key elements of QoS deployment that Cisco AutoQoS addresses?
4. Which application types is NBAR protocol discovery able to identify and classify?
5. On what types of router interfaces or PVCs can you enable Cisco AutoQoS?
6. What are the router prerequisites for configuring AutoQoS?
7. What are the two steps (or phases) of AutoQoS for Enterprise?
8. List at least two commands for verifying AutoQoS on Cisco routers.
9. List at least two commands for verifying AutoQoS on Cisco LAN switches.
10. What are the three most common Cisco AutoQoS issues that can arise, and their
corresponding solutions?
11. List at least three pieces of information that can be obtained from the output of the show auto
qos command.
12. What are the two major reasons for modifying the configuration that AutoQoS generates?
13. Specify two methods for modifying and tuning the AutoQoS-generated class maps and policy
maps.
14. In addition to using NBAR and ACLs, what classification options does MQC offer?
1763fm.book Page 228 Monday, April 23, 2007 8:58 AM

This part covers the following ONT exam topics. (To view the ONT exam
overview, visit http://www.cisco.com/web/learning/le3/current_exams/
642-845.html.)

■ Describe and configure WLAN QoS.
■ Describe and configure wireless security on Cisco Clients and APs (e.g.,
SSID, WEP, LEAP, etc.).
■ Describe basic wireless management (e.g., WLSE and WCS). Configure and
verify basic WCS configuration (i.e., login, add/review controller/AP status,
security, and import/review maps).
1763fm.book Page 229 Monday, April 23, 2007 8:58 AM

Part III: Wireless LAN

Chapter 8 Wireless LAN QoS Implementation

Chapter 9 Introducing 802.1x and Configuring Encryption and
Authentication on Lightweight Access Points

Chapter 10 WLAN Management
1763fm.book Page 230 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:

■ The Need for Wireless LAN QoS

■ Current Wireless LAN QoS
Implementation

■ Configuring Wireless LAN QoS
1763fm.book Page 231 Monday, April 23, 2007 8:58 AM

CHAPTER 8
Wireless LAN QoS
Implementation

This chapter first describes WLAN QoS and why it is needed. Next, it covers the current
implementation of WLAN QoS. The last section describes how to configure WLAN QoS for
different QoS profiles using the Cisco WCS web user interface.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 10-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 8-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics. You can keep track of your score here, too.

Table 8-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“The Need for Wireless LAN QoS” 1–6
“Current Wireless LAN QoS Implementation” 7–8
“Configuring Wireless LAN QoS” 9–10
Total Score (10 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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232 Chapter 8: Wireless LAN QoS Implementation

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 6 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 7–8 overall score—Begin with the “Foundation Summary” section and then follow up with
the “Q&A” section at the end of the chapter.

■ 9 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. Select the correct statement about wireless LANs.
a. WLANs are mostly implemented as extensions to wired LANS.
b. WLANs are occasionally implemented as overlays to wired LANs.
c. WLANs are sometimes implemented as substitutes for wired LANs.
d. All of the above.
2. Which statement is true about 802.11 wireless media access control?
a. It uses CSMA/CD.
b. It uses token passing.
c. It uses CSMA/CA.
d. All of the above.
3. Distributed coordinated function (DCF) performs collision avoidance using which of these?
a. Radio frequency (RF) carrier sense
b. Interframe spacing (IFS)
c. Random back-off/contention windows (CW)
d. All of the above
4. IEEE provides QoS extensions to wireless LANs by which of the following drafts/standards?
a. 802.11g
b. 802.11e
c. 802.11d
d. 802.11a
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“Do I Know This Already?” Quiz 233

5. Select the item that is not a real-time function performed by the access point in the Split-MAC
architecture.
a. Key management
b. Beacon generation
c. Probe transmission
d. Encryption/decryption
6. Which of the following shows the correct mapping of 802.11e priority levels to WMM access
categories?
a. Voice(Platinum)=6/7, Video(Gold)=4/5, Best-Effort(Silver)=1/2, Back-
ground(Bronze)=0/3
b. Voice(Platinum)=6/7, Video(Gold)=4/5, Best-Effort(Silver)=3/2, Back-
ground(Bronze)=0/1
c. Voice(Platinum)=6/7, Video(Gold)=4/5, Best-Effort(Silver)=0/3, Back-
ground(Bronze)=1/2
d. None of the above
7. Select the correct statement about how wireless LAN controller copies/maps QoS fields in the
Split-MAC architecture.
a. Wireless LAN controller copies the IP DSCP field (inner) to the DSCP field (outer) of
the LWAPP data unit.
b. Wireless LAN controller maps the IP DSCP field (inner) to the 802.1p field (outer) of
the LWAPP data unit.
c. Wireless LAN controller maps the DSCP field from the LWAPP data unit to the 802.1p
field on the 802.1Q frame.
d. All of the above.
8. Which priority (802.1p) value is used/reserved for LWAPP control messages?
a. 6
b. 7
c. 0
d. 1
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234 Chapter 8: Wireless LAN QoS Implementation

9. Which of the following is not a parameter set in the Edit QoS Profile page of the Web User
Interface for Wireless LAN Controller, as a part of Per-User Bandwidth Contract?
a. Average voice rate
b. Average data rate
c. Burst data rate
d. Burst real-time rate
10. On the WLANs > Edit page of the web user interface of the wireless LAN controller, what
does it mean if the general WMM or 802.11e policy for the interaction between the wireless
client and the access point is set to Required?
a. This setting means that WMM or 802.11e QoS requests are ignored.
b. This setting means that QoS is offered to WMM or 802.11e-capable clients.
c. This setting means that all clients must be WMM/802.11e compliant to use this
WLAN ID.
d. None of the above is correct.
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The Need for Wireless LAN QoS 235

Foundation Topics

The Need for Wireless LAN QoS
WLANs are growing in popularity. They are mostly implemented as extensions to, but are occasionally
deployed as overlays to, wired LANs, or replacements for wired LANs. The difference between
wired and wireless LANs is in the physical layer and in the MAC layer. Please note that Logical
Link Control (LLC) and MAC are considered upper and lower sublayers of the OSI Layer 2 Data
Link Control (DLC) layer, respectively. Upper-layer protocols and applications such as IP, TCP,
and FTP run identically on both wired and wireless platforms. Figure 8-1 shows two access
switches (Layer 2) connected to a distribution (multilayer) switch. The access switch on the right
side has wired devices plugged into its access ports. The access switch on the left side, however,
is connected to a wireless LAN controller (WLC), which in turn is connected to and controls two
wireless LAN APs. Each wireless client on the left side of Figure 8-1 communicates with an AP
by sending and receiving frames over the radio frequency (RF) link, and it gains access to the
network.
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236 Chapter 8: Wireless LAN QoS Implementation

Figure 8-1 Wireless LAN Extending the Wired LAN

Wireless LAN
Controller (WLC)

Layer 2/3
Network

Wireless
Access Points
(AP)

Wireless
Clients

Wired Ethernet uses carrier sense multiple access with collision detection (CSMA/CD) as its
MAC mechanism. Wireless LAN (802.11), on the other hand, lacks the ability to read and send
data at the same time; therefore, it cannot detect collision like its wired counterpart can. Hence,
WLAN uses carrier sense multiple access with collision avoidance (CSMA/CA) as the MAC
mechanism. Collision avoidance is accomplished by distributed coordinated function (DCF). DCF
uses RF carrier sense, inter-frame spacing (IFS), and random back-off/CWs. Please note that
random back-off/CWs are sometimes casually referred to as random wait timers.
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The Need for Wireless LAN QoS 237

To be able to offer end-to-end QoS, the wireless portion and components of a network must
comply with and satisfy the QoS needs of the applications. Following are some of the main QoS
needs of applications, such as voice and video:

■ Dedicated bandwidth

■ Controlled jitter and delay (latency)

■ Managed congestion

■ Shaped traffic (rate limited)

■ Prioritized traffic (with drop preference)

WLAN QoS Description
IEEE has provided the QoS extensions to WLANs in the 802.11e specifications. The ONT
courseware refers to 802.11e as a draft for standardization, but at the time of this writing, 802.11e
is already approved and is considered a new standard. IEEE defines 802.11e as the first wireless
standard, adding QoS features to the existing IEEE 802.11b and IEEE 802.11a (and other)
wireless standards, while maintaining full backward compatibility with them. While 802.11e was
in the standardization process, Wi-Fi Alliance released a specification called the Wi-Fi Multimedia
(WMM) for the interim period.

WMM is a subset of 802.11e; for instance, WMM reduces the eight priority levels of 802.11e to
four access categories. Note that access category has the same meaning as priority level. Using the
basic CSMA/CA-based DCF, each client generates a random back-off number between 0 and a
minimum contention window (CWmin) and waits until the RF channel is free for an interval called
distributed coordinated function inter-frame space (DCF IFS or DIFS). From that moment on, the
channel is continuously checked; if it is free, the random back-off number is decremented by 1
until it becomes 0. At that time, the client sends the frame. If the channel becomes busy, the client
has to wait until the channel is free, wait for a DIFS interval, and start decrementing the random
back-off interval all over again.

The CSMA/CA-based DCF gives all devices the same priority, so it is considered a best-effort
mechanism. WMM, on the other hand, provides traffic prioritization (or RF prioritization) by
using four access categories: Platinum (or voice), Gold (or video), Silver (best-effort), and Bronze
(background), in descending priority order. The four access categories are in effect four queues,
each of which gets a higher probability of transmitting than the access priority (or queue) below
it. If a specific type of traffic is not assigned to an access category, it is categorized as best-effort
(Silver). The eight 802.11e priority levels are mapped to four WMM access categories, as shown
in Table 8-2.
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238 Chapter 8: Wireless LAN QoS Implementation

Table 8-2 Mapping of 802.11e Priority Levels to WMM Access Categories

WMM Access Category 802.11e Priority Level
Voice (Platinum) 6 or 7
Video (Gold) 4 or 5
Best-Effort (Silver) 0 or 3
Background (Bronze) 1 or 2

802.11e (and its subset WMM) uses Enhanced Distributed Coordination Function (EDCF) by
employing different CW/back-off timer values for different priorities (access categories). If a
client finds the RF channel available, it waits for a DIFS period, and then it has to wait for a
random back-off period based on the CWmin associated with the priority of the traffic being
submitted (more accurately, the queue that the traffic is submitted from). If the traffic is high
priority, its CWmin is smaller, giving it a shorter back-off timer value; if the traffic is lower priority,
its CWmin is larger, giving it a longer back-off timer value. Note that with EDCF, even though
high-priority traffic such as voice is statistically expected to be transmitted before lower-priority
traffic, it is not guaranteed to do so at all times; therefore, technically EDCF cannot be equated to
a strict priority system. With EDCF, IFS (Inter Frame Space) is referred to as AIFS (Arbitrated
IFS).

NOTE In the original ONT student course material, on the page titled “WLAN QoS RF Back-
Off Timing,” SIFS is mistakenly used instead of DIFS. Short inter-frame space (SIFS) is used
only before transmitting important frames such as acknowledgements, and it has no random
back-off. SIFS is not used to transmit regular data frames. Data frames, on the other hand, must
wait for a DIFS and then begin the random back-off procedure.

Split MAC Architecture and Light Weight Access Point
To centralize the security, deployment, management, and control aspects of WLANs, Split MAC
Architecture (a part of Cisco Unified Wireless Network Architecture) shifts some of the functions
traditionally performed on the autonomous AP to a central location (device). The main functions
performed on legacy autonomous APs are shown in Table 8-3 categorized under two columns:
real-time 802.11/MAC functionality, and non-real-time 802.11/MAC functionality.

Table 8-3 Real-Time and Non-Real-Time 802.11 MAC Functions

802.11/MAC Real-Time Functions 802.11/MAC Non-Real-Time Functions
Beacon generation Association/disassociation/reassociation
Probe transmission and response 802.11e/WMM resource reservation
Power management 802.1x EAP
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Current Wireless LAN QoS Implementation 239

Table 8-3 Real-Time and Non-Real-Time 802.11 MAC Functions (Continued)

802.11/MAC Real-Time Functions 802.11/MAC Non-Real-Time Functions
802.11e/WMM scheduling and queuing Key management
MAC layer data encryption/decryption Authentication
Control frame/message processing Fragmentation
Packet buffering Bridging between Ethernet and WLAN

To address the centralized RF management needs of the enterprises, Cisco designed a centralized
lightweight AP (LAP or LWAP) wireless architecture with Split-MAC architecture as its core.
Split-MAC architecture divides the 802.11 data and management protocols and AP capabilities
between a lightweight AP and a centralized WLAN controller. The real-time MAC functions, such
as those listed in the left column of Table 8-3, including handshake with wireless clients, MAC
layer encryption, and beacon handling, are assigned to the LWAP. The non-real-time functions
such as those listed in the right column of Table 8-3, including frame translation and bridging, plus
user mobility, security, QoS, and RF management, are assigned to the wireless LAN controller.

Current Wireless LAN QoS Implementation
Wireless RF is an OSI Layer 2 technology, and its QoS is currently based on 802.11e or WMM
specifications. With the addition of WLANs, to maintain end-to-end QoS in a network, it is
necessary to perform mapping between Layer 2 (802.1p) priority or Layer 3 DSCP (or IP
precedence) and 802.11e priority (or WMM access category). If a wireless AP connects to an
access port (non-trunk port, lacking 801.1p marking) of a LAN switch, the Layer 2 802.11e (or
WMM) marking of data coming from the wireless client is lost because the data is forwarded from
the AP to the LAN switch. This is also true for the traffic arriving at the LAN switch (with Layer
2 801.1p marking), which then has to forward the traffic through an access port to the wireless AP.
In the absence of Layer 2 QoS information, such as on a non-trunk connection, it will be necessary
to utilize the Layer 3 QoS information such as the DSCP marking.

In the Cisco centralized LWAP wireless architecture (with Split-MAC architecture as its core),
WLAN controller ensures that traffic traversing between it and the LWAP maintains its QoS
information. The WLAN data coming from the wireless clients to the LWAP is tunneled to the
WLAN controller using Lightweight Access Point Protocol (LWAPP). In the opposite direction,
the traffic coming from the wired LAN to the WLAN controller is also tunneled to the LWAP using
LWAPP.

Figure 8-2 shows a WLAN controller with a wired 802.1Q trunk connection to a multilayer LAN
switch. The WLAN controller has two LWAPs associated to it. The WLAN controller has an
LWAPP tunnel set up with each of the LWAPs. The LWAPP tunnel can be set up over a Layer 2
or a Layer 3 network. In Layer 2 mode, the LWAPP data unit is in an Ethernet frame. Furthermore,
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240 Chapter 8: Wireless LAN QoS Implementation

the WLAN controller and the AP must be in the same broadcast domain and IP subnet. In Layer 3
mode, however, the 3 LWAPP data unit is in a User Datagram Protocol (UDP/IP frame). Moreover,
the WLAN controller and AP can be in the same or different broadcast domains and IP subnets.
Examples for the supported wireless LAN controllers are Cisco 2000, 4100, 4400 Series wireless
LAN controllers, Cisco WiSM, Cisco WLCM for integrated services routers, Airespace 3500,
4000, and 4100 Series wireless LAN controllers. Examples for the supported APs are Cisco
Aironet 1000, 1130, 1230, 1240, and 1500 series LWAPs.

Figure 8-2 LWAPP Tunnel in the Split-MAC Architecture

Multilayer
LAN Switch

Layer 2 (802.1p) QoS
802.1Q
+
Trunk
Layer 3 (DSCP) QoS

Wireless LAN
Controller (WLC)

LWAPP Layer 2/3 LWAPP
Tunnel Network Tunnel

Lightweight
Access Points

802.11e or
RF WMM QoS RF

Wireless
Clients

To maintain continuous (end-to-end) QoS, the WLAN controller on one end of the LWAPP tunnel
and the LWAP at the other end of the LWAPP tunnel must do some mapping between the QoS
marking of the received data units and the QoS markings/fields of the data unit they send forward.
The wireless LAN controller sends and receives 802.1Q frames to and from the wired multilayer
LAN switch. 802.1Q frames have the CoS (priority/802.1p) field for QoS marking purposes. The
wireless LAN controller and the LWAP send and receive LWAPP data units to each other. The
LWAPP data unit has an 802.1p (CoS) equivalent field and a DSCP equivalent field. (The LWAP
does not understand the 802.1p field of the LWAPP data unit.) The LWAP and the wireless clients
exchange RF, with 802.11e (or WMM) providing the QoS marking field. Table 8-4 shows the
mapping between IP DSCP value and 802.1p and 802.11e values.
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Current Wireless LAN QoS Implementation 241

Table 8-4 QoS Markings Mapping Table

802.1p
Cisco 802.1Q/P Priority-Based Traffic Type IP DSCP Priority 802.11e Priority
Network control/reserved 56–62 7 7
Inter-network control/IP routing 48 6 7
Voice 46 (EF) 5 6
Video 34 4 5
(AF41)
Voice control 26 3 4
(AF31)
Background (Gold) 18 2 2
(AF21)
Background (Silver) 10 1 1
(AF11)
Best effort 0 (BE) 0 0 or 3

Figure 8-3 is a comprehensive depiction of data moving from a multilayer LAN switch to a
wireless client through a wireless LAN controller and an LWAP, and data moving in the opposite
direction from the wireless client to the multilayer LAN switch through an LWAP and a wireless
LAN controller. This process involves four steps, accordingly marked in Figure 8-3, which will be
addressed next.

Step 1 in Figure 8-3 shows that when the WLAN controller receives an 802.1Q frame (encapsulating
an IP packet) from the multilayer LAN switch, it forwards the IP packet toward the LWAP,
encapsulating it in an LWAPP data unit. For QoS continuity purposes, the WLAN controller
copies the IP DSCP field (inner) to the LWAPP data unit DSCP field (outer). The WLAN
controller also maps the IP DSCP field (inner) to the LWAPP data unit’s 802.1p field (outer). The
mapping of DSCP to 802.1p is done according to Table 8-4. Please note that the LWAPP control
packets exchanged between the WLAN controller and the LWAP are always tagged with the
802.1p value of 7. Indeed, LWAPP reserves the 802.1p value of 7 for the LWAPP control packets.

Step 2 in Figure 8-3 shows that when the LWAP receives a LWAPP data unit (encapsulating an IP
packet) from the WLAN controller, it forwards the IP packet toward the wireless client using RF,
and it uses 802.11e/WMM for Layer 2 QoS marking. The LAWP maps the DSCP field from the
LWAPP data unit to the 802.11e/WMM field based on Table 8-4.
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242 Chapter 8: Wireless LAN QoS Implementation

Figure 8-3 Mapping of Inner QoS Fields to LWAPP Tunnel QoS Fields
Multilayer
(Wired)
LAN Switch
802.1Q
Trunk
IP Packet

802.1p 802.1p
DSCP ... DSCP ...
(CoS) (CoS)

Wireless LAN Mapped Copied 1
Mapped 4 Controller

DSCP ... DSCP ... 802.1p DSCP ... DSCP ...
(Outer) (Inner) (Outer) (Outer) (Inner)

LWAPP Data Unit LWAPP Data Unit
Encapsulating an IP Packet LWAPP Encapsulating an IP Packet
Tunnel
Mapped 3 Mapped 2

LAP

802.11e DSCP 802.11e DSCP
... ...
(or WMM) (Inner) (or WMM) (Inner)

Wireless
Client

In Step 3, shown in Figure 8-3, the LWAP receives RF from the wireless client transporting an IP
packet with the Layer 2 QoS marking in the 802.11e/WMM field. The LWAP forwards the IP
packet toward the WLAN controller, encapsulating it in an LWAPP data unit. For QoS continuity
purposes, the LWAP maps the 802.11e value to the DSCP field on the LWAPP data unit based on
Table 8-4. Note that because the LWAP does not understand 802.1p, it does not mark that field on
the LWAPP data unit.

Finally, in Step 4, shown in Figure 8-3, the WLAN controller receives an LWAPP data unit from
the LWAP encapsulating an IP packet. The WLAN controller forwards the IP packet toward the
multilayer LAN switch, encapsulating it in an 801.1Q frame (over the trunk connection). For QoS
continuity purposes, the WLAN controller maps the DSCP field from the LWAPP data unit to the
802.1p field on the 802.1Q frame based on Table 8-4. Please note that packets with no QoS
markings received from the WLAN will be categorized as best-effort (default Silver) when the
WLAN controller transmits them toward the LAN.
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Configuring Wireless LAN QoS 243

Configuring Wireless LAN QoS
Cisco WLAN controllers have a built-in web user interface. Figure 8-4 shows a typical web user
interface.

Figure 8-4 A Typical Web User Interface for Cisco WLAN Controllers
Menu Bar Buttons Administrative Tools

Selector Area Main Data Page

The web user interface has five main areas:

■ Administrative tools

■ Menu bar

■ Buttons

■ Selector area

■ Main data page
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244 Chapter 8: Wireless LAN QoS Implementation

The menu bar has the following selections available:

■ Monitor

■ WLANs

■ Controller

■ Wireless

■ Security

■ Management

■ Commands

■ Help

The Controller option from the web user interface menu bar provides access to many pages,
including the QoS Profiles page. On the QoS Profiles page, you can view the names and
descriptions of the QoS profiles, and you can edit each of the profiles by clicking on the Edit
button. If you select the Bronze profile, for example, and click on the Edit button, you see a page
like the one shown in Figure 8-5.

Figure 8-5 Edit QoS Profile Page
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Configuring Wireless LAN QoS 245

Even though the EDCF sets the priority or access category for each profile, in the Edit QoS Profile
page (shown in Figure 8-5), you can set the average data rate, burst data rate, average real-time
rate, and burst real-time rate, as parts of per-user bandwidth contract. You can set these fields to 0
up to 60,000 bits per second; their default value is 0, which means that the feature is off.

You can configure two Over the Air QoS fields on the Edit QoS Profile Page for the profile you are
editing: Maximum RF Usage Per AP (%), and Queue Depth. The maximum RF usage per AP
for each profile is set to 100 (%) by default. If the queue depth for a profile is reached, packets of
that profile are dropped at the AP. The default queue depth values can vary from one controller
model to another. The controller example used in the ONT courseware has these queue depth
default values: 100 for Platinum, 75 for Gold, 50 for Silver, and 25 for Bronze.

On the Edit QoS Profile page, under the Wired QoS Protocol heading, you can select the protocol
type as either 802.1P or None. Selecting 802.1P activates 802.1P Priority Tags, and selecting
None deactivates 802.1P Priority Tags (default). If you select 802.1P for protocol type, you can
then set 802.1P Tag for the wired connection to a number from 0 to 7. The default mappings for
the four access categories are 6 for Platinum, 5 for Gold, 3 for Silver, and 1 for Bronze.

WLANs is another option from the web user interface menu bar (see Figure 8-4). It allows you to
create, configure, and delete WLANs on your controller. The WLANs menu bar provides you with
these selections:

■ WLANs

■ WLANs > New

■ WLANs > Edit

■ WLANs > Mobility Anchors

■ AP Groups VLAN

To configure existing WLANs, you must click on Edit. A page similar to the one shown in Figure
8-6 is displayed.
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246 Chapter 8: Wireless LAN QoS Implementation

Figure 8-6 WLANs > Edit Page

On the WLANs > Edit page, for each WLAN ID, you can set the Quality of Service field to one
of Platinum (voice), Gold (video), Silver (best effort), or Bronze (background). On the same
page, you can set the general WMM or 802.11e policy for the interaction between wireless client
and the AP to Disabled, Allowed, or Required:

■ Disabled—This setting means that WMM or 802.11e QoS requests are ignored.

■ Allowed—This setting means that QoS is offered to WMM or 802.11e-capable clients.
Default QoS is offered to non-WMM/802.11e-capable clients.

■ Required—This setting means that all clients must be WMM/802.11e compliant to use this
WLAN ID.
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Foundation Summary 247

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should help
solidify some key facts. If you are doing your final preparation before the exam, the information in
this section is a convenient way to review the day before the exam.

Wireless LANs (WLANs) are extensions to wired LANs; the same protocols and applications can
run on both. The media access method of WLAN is carrier sense multiple access collision avoid
(CSMA/CA). Wireless (802.11) uses distributed coordinated function (DCF) to avoid collision.
DCF is based on RF carrier sense, inter-frame spacing (IFS), and random wait timers.

To continue to support real-time applications such as voice and video that have specific requirements
such as minimum dedicated bandwidth, maximum delay, maximum jitter, maximum packet loss,
and so on, the wireless components of a network must also offer QoS capabilities and features.
802.11e, which was in draft but is now a standard, provides QoS extensions to 802.11 wireless.
Wi-Fi Alliance released a Wi-Fi Multimedia (WMM) standard while the 802.11e was in the
process of being approved as a standard. WMM has four access categories compared to the eight
priority levels of 802.11e. Table 8-5 shows the mapping between 802.11e priorities and the WMM
access categories.

Table 8-5 Mapping of 802.11e Priority Levels to WMM Access Categories

WMM Access Category 802.11e Priority Level
Voice (Platinum) 6 or 7
Video (Gold) 4 or 5
Best-Effort (Silver) 0 or 3
Background (Bronze) 1 or 2

802.11e and WMM replace DCF with Enhanced DCF (EDCF). In addition to prioritizing/
categorizing the data (see Table 8-5), 802.11e/WMM uses different minimum wait times and
random back-off times for traffic with different priorities (access categories). Figure 8-7 provides
a pictorial representation of this method.
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248 Chapter 8: Wireless LAN QoS Implementation

Figure 8-7 WLANs QoS RF Back-Off Timings
PB
Platinum/Voice Priority-Based Random
DIFS MWT
Queue (Priority 6/7) Back-Off Time
(Slots)

Priority Based Minimum Wait Time

PB
Gold/Video Priority-Based Random
DIFS MWT
Queue (Priority 4/5) Back-Off Time
(Slots)

PB
Silver/Best-Effort Priority-Based Random
DIFS MWT
Queue (Priority 0/3) Back-Off Time
(Slots)

Bronze/Background Priority-Based Minimum
DIFS Priority-Based Random Back-Off Time
Queue (Priority 1/2) Wait Time (Slots)

The Cisco Split-MAC architecture separates the real-time aspects of the 802.11 protocol from its
non-real-time/management aspects. Instead of an autonomous access point (AP), the Split-MAC
architecture uses lightweight access points (LWAPs), which will handle real-time functions, and
WLAN controllers, which will handle the non-real-time functions. A WLAN controller can have
one or more LAPs associated with it. The wireless LAN controller and LWAP communicate using
Lightweight Access Point Protocol (LWAPP) over a Layer 2 (Ethernet) or a Layer 3 (UDP/IP)
network. Table 8-6 shows the main real-time tasks assigned to the LWAP and the main non-real-
time tasks assigned to the wireless LAN controller within the Cisco Split-MAC model.

Table 8-6 Division of Functions Between Wireless LAN Controller and LWAP

802.11/MAC Real-Time Functions 802.11/MAC Non-Real-Time Functions
Performed at the LWAP Performed at the WLAN Controller
Beacon generation Association/disassociation/reassociation
Probe transmission and response 802.11e/WMM resource reservation
Power management 802.1x EAP
802.11e/WMM scheduling and queuing Key management
MAC layer data encryption/decryption Authentication
Control frame/message processing Fragmentation
Packet buffering Bridging between Ethernet and WLAN
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Foundation Summary 249

To provide end-to-end QoS within a network that includes wireless subsets, the LWAPP data units
moving between the wireless LAN controller and the lightweight access point (LWAP) have Layer
2 (802.1p) or Layer 3 (DSCP) fields. Figure 8-8 shows how the WLAN controllers and LWAPs
accomplish the mapping of QoS markings (within the Split-MAC architecture).

Figure 8-8 Mapping of QoS Fields by LAPs and WLAN Controllers
Multilayer
(Wired)
LAN Switch
802.1Q
Trunk
IP Packet

802.1p 802.1p
DSCP ... DSCP ...
(CoS) (CoS)

Wireless LAN Mapped Copied 1
Mapped 4 Controller

DSCP ... DSCP ... 802.1p DSCP ... DSCP ...
(Outer) (Inner) (Outer) (Outer) (Inner)

LWAPP Data Unit LWAPP Data Unit
Encapsulating an IP Packet LWAPP Encapsulating an IP Packet
Tunnel
Mapped 3 Mapped 2

LAP

802.11e DSCP 802.11e DSCP
... ...
(or WMM) (Inner) (or WMM) (Inner)

Wireless
Client

Packet marking translations among DSCP, 802.1p, and 802.11e priorities performed at wireless
LAN controller and LWAP are based on Table 8-7.

Table 8-7 QoS Markings Mapping Table

802.1p 802.11e
Cisco 802.1Q/P Priority-Based Traffic Type IP DSCP Priority Priority
Network control/reserved 56–62 7 7
Inter-network control/IP routing 48 6 7
Voice 46 (EF) 5 6
continues
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250 Chapter 8: Wireless LAN QoS Implementation

Table 8-7 QoS Markings Mapping Table (Continued)

802.1p 802.11e
Cisco 802.1Q/P Priority-Based Traffic Type IP DSCP Priority Priority
Video 34 4 5
(AF41)
Voice control 26 3 4
(AF31)
Background (Gold) 18 2 2
(AF21)
Background (Silver) 10 1 1
(AF11)
Best-effort 0 (BE) 0 0 or 3

The Controller option from the web user interface menu bar provides access to many pages,
including the QoS Profiles page. On the QoS Profiles page, you can view the names and
descriptions of the QoS profiles, and you can edit each of the profiles by clicking on the Edit
button. Table 8-8 shows the fields within the Edit QoS Profiles page.

Table 8-8 Fields Shown on the QoS Profiles Page of the Web User Interface

Field Explanation/Default Value/Value Range
Description QoS profile description.
Average Per-User Contract Data 0 to 60,000 bits per second.
Rate
Average data rate for non-UDP traffic.

Default 0 = OFF.
Burst Per-User Contract Data Rate 0 to 60,000 bits per second. Operator-defined.

Peak data rate for non-UDP traffic.

Default 0 = OFF.
Average Per-User Contract Real- 0 to 60,000 bits per second.
Time Rate
Average data rate for UDP traffic.

Default 0 = OFF.
Burst Per-User Contract Real-Time 0 to 60,000 bits per second.
Rate
Peak data rate for UDP traffic.

Default 0 = OFF.
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Foundation Summary 251

Table 8-8 Fields Shown on the QoS Profiles Page of the Web User Interface (Continued)

Field Explanation/Default Value/Value Range
Maximum QoS RF Usage per AP 1 to 100%.

Maximum air bandwidth available to a class of clients.

Default = 100%.
QoS Queue Depth Depth of queue for a class of client.

Causes packets with a greater value to be dropped at the
access point.

(25 for Bronze, 50 for Silver, 75 for Gold, and 100 for
Platinum on some controllers.)
Wired QoS Protocol 802.1P activates 802.1P priority tags, and None
deactivates 802.1P priority tags (default).
802.1P Tag 802.1P priority tag for the wired connection.

0 to 7.

Used for traffic and LWAPP packets.

1 for Bronze, 3 for Silver, 4/5 for Gold, and 6 for
Platinum.
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252 Chapter 8: Wireless LAN QoS Implementation

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. How does distributed coordinated function (DCF) accomplish collision avoidance?
2. What is the standard (or draft name) for the wireless QoS of IEEE?
3. What is the Wi-Fi Alliance specification for wireless QoS?
4. Describe the relationship between 802.11e priorities and WMM access categories.
5. What contention mechanism does 802.11e use to provide prioritized RF access?
6. Describe the Split-MAC architecture.
7. List at least three of the real-time MAC functions that are assigned to the LWAP in the Split-
MAC architecture.
8. List at least three of the non-real-time MAC functions that are assigned to the wireless LAN
controller in the Split-MAC architecture.
9. Which protocol is used between the wireless LAN controller and the lightweight access point
in the Split-MAC architecture?
10. From which page of the web user interface of the wireless LAN controller can you examine
and modify QoS profiles?
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This chapter covers the
following subjects:

■ Overview of WLAN Security

■ 802.1x and EAP Authentication Protocols

■ Configuring Encryption and Authentication
on Lightweight Access Points
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CHAPTER 9
Introducing 802.1x and Configuring
Encryption and Authentication on
Lightweight Access Points

This chapter is composed of three sections. In the first section, you are provided with an intro-
duction to wireless security, its issues, and how it has evolved. In the next section, the 802.1
extensible authentication protocol (EAP) and some of its popular variants are presented.
Wireless protected access (WPA and WPA2) and 802.11i security standards are also presented
in this section. The final section of this chapter shows how you can navigate through the graphic
user interface of a wireless LAN controller (WLC) using a web browser to set up various
authentication and encryption options on lightweight access points (LWAP).

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 10-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 9-1 outlines the major topics discussed in this chapter and the “Do I Know This Already?”
quiz questions that correspond to those topics.

Table 9-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score

“Overview of WLAN Security” 1–4

“802.1x and EAP Authentication Protocols” 5–9

“Configuring Encryption and Authentication on Lightweight 10
Access Points”

Total Score (10 possible)

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this
chapter. If you do not know the answer to a question or are only partially sure of the answer,
mark this question wrong for purposes of the self-assessment. Giving yourself credit for an
answer you correctly guess skews your self-assessment results and might provide you with a
false sense of security.
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256 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 6 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.
■ 7–8 overall score—Begin with the “Foundation Summary” section and then follow up with
the “Q&A” section at the end of the chapter.
■ 9 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.
1. Which of the following is not an issue or a weakness of initial WLAN security approaches?
a. Relying on SSID as a security measure
b. Relying on MAC filters
c. Overhead of mutual authentication between wireless clients and access control/authenti-
cation servers
d. Usage of static WEP
2. Which of the following is not considered a weakness of WEP?
a. With enough data captured, even with initialization vector used, the WEP key can be
deducted.
b. WEP is vulnerable to dictionary attacks.
c. Because with basic WEP the wireless client does not authenticate the access point, the
client can be victimized by rogue access points.
d. The WEP usage of certificates is not convenient for some customers.
3. Which of the following organizations developed LEAP to address the shortcomings of WEP?
a. Wi-Fi Alliance Group
b. Cisco
c. IEEE
d. Microsoft
4. Which of the following organizations developed WPA?
a. Wi-Fi Alliance Group
b. Cisco
c. IEEE
d. Microsoft
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“Do I Know This Already?” Quiz 257

5. Which of the following is not a required component for 802.1x authentication?
a. External user database
b. Supplicant (EAP-capable client)
c. Authenticator (802.1x-capable access point)
d. Authentication server (EAP-capable RADIUS server)
6. Which of the following is not a LEAP feature?
a. Usage of PKI
b. Fast, secure roaming with Cisco or Cisco-compatible clients
c. True single login with an existing username and password using Windows NT/2000
Active Directory (or Domain)
d. Support for a wide range of operating systems (such as Microsoft, Macintosh, Linux,
and DOS)
7. Which of the following is not an EAP-FAST feature?
a. Provides full support for 802.11i, 802.1x, TKIP, and AES
b. Supports Windows single sign-on for Cisco Aironet clients and Cisco-compatible clients
c. Uses certificates (PKI)
d. Supports password expiration or change (Microsoft password change)
8. Which of the following is an EAP-TLS feature?
a. It uses PKI.
b. Its supported clients include Microsoft Windows 2000, XP, and CE, plus non-Windows
platforms with third-party supplicants such as Meetinghouse.
c. It permits a single logon to a Microsoft domain.
d. All of the above.
9. Which of the following is not true about PEAP?
a. It builds an encrypted tunnel in Phase 1.
b. Only the server authentication is performed using PKI certificate.
c. All PEAP varieties support single login.
d. Cisco Systems, Microsoft, and RSA Security developed PEAP.
10. When you use a web browser to access a WLC GUI to modify or configure the encryption and
authentication settings of a wireless LAN, which item of the main toolbar should you click
on first?
a. Security
b. Configure
c. WLAN
d. Management
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258 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Foundation Topics

Overview of WLAN Security
Affordability, ease of use, and convenience of wireless devices, wireless local-area networks
(WLAN), and related technologies have caused a substantial increase in their usage over recent
years. At the same time, the number of reported attacks on wireless devices and networks has
surged. Hackers have access to affordable wireless devices, wireless sniffers, and other tools.
Unfortunately, the default wireless security settings are usually open and vulnerable to intrusion
and attacks. For example, if encryption is not enabled, sensitive and private information sent over
a wireless LAN can easily be sniffed (captured). One of the common methods that hackers use is
called war driving. War driving refers to the process whereby someone drives around with a laptop
equipped with a wireless network interface card (NIC), looking for vulnerable wireless devices
and networks. Best practices require that authentication and encryption be used to protect wireless
client data from security and privacy breaches. User authentication allows the network devices to
check and ensure legitimacy of a user and protect the network from unauthorized users trying to
gain access to the network and all the confidential data/files. Encryption is used so that, if someone
captures data during transit through sniffing, for example, he cannot read it. The illegitimate
capturer of data needs to know the key and the algorithm used to encrypt the data to decrypt it.

WLAN Security Issues
The main security problem with wireless LANs is and has been that the available security features
are not enabled and used. However, for those who have been interested and keen to secure their
wireless networks, the available features have not always been as sophisticated as they are today.

Service Set Identifier (SSID) is the method for naming a wireless network. The SSID configuration
of a client must match the SSID of the wireless access point (AP) for the client to communicate
with that AP. However, if the client has a null SSID, it can request and acquire the SSID from the
AP. Unless the AP is configured not to broadcast its SSID, the AP responds to the wireless client
request and supplies the SSID to the client; the client can then associate to that AP and access the
wireless network. Some people mistakenly think that if the AP is configured not to broadcast its
SSID, they have a secure wireless LAN; that is not true. When a legitimate wireless client with the
correct SSID attempts to associate with its AP, the SSID is exchanged over the air unencrypted;
that means that an illegitimate user can easily capture and use the SSID. The conclusion is that
SSID should not be considered a wireless security tool. SSID is used to logically segment wireless
clients and APs into groups.
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Overview of WLAN Security 259

Rogue APs impose threats to wireless LANs. A rogue AP is illegitimate; it has been installed
without authorization. If an attacker installs a rogue AP and clients associate with it, he can easily
collect sensitive information such as keys, usernames, passwords, and MAC addresses. Unless the
client has a way of authenticating the AP, a wireless LAN should have a method to detect rogue
APs so that they can be removed. Furthermore, attackers sometimes install rogue APs intending
to interfere with the normal operations and effectively launch denial of service (DoS) attacks.

Some wireless LANs use MAC filters. Using MAC filters, the wireless LANS check the wireless
MAC address of a client against a list of legitimate MAC addresses before granting the client
access to the network. Unfortunately, MAC addresses can be easily spoofed, rendering this
technique a weak security feature.

The 802.11 Wired Equivalent Privacy (WEP), or basic 802.11 security, was designed as one of the
first real wireless security features. WEP has several weaknesses; therefore, it is not recommended
for use unless it is the only option available. For example, with enough data captured, hacking
software can deduct the WEP key. Because of this weakness, usage of initialization vector (IV)
with WEP has become popular. The initialization vector is sent to the client, and the client uses it
to change the WEP key, for example, after every packet sent. However, based on the size of the IV,
after so much data is sent, the cycle begins with the initial key again. Because the IV is sent to the
client in clear text and the keys are reused after each cycle, with enough data captured, the hacker
can deduct the WEP key. WEP has two other weaknesses. First, it is vulnerable to dictionary
attacks because, using dictionary words, the hackers keep trying different WEP keys and might
succeed in guessing the correct WEP key. Second, using WEP, the wireless client does not
authenticate the AP; therefore, rogue APs can victimize the client.

Evolution of WLAN Security Solutions
802.11 WEP using 40-bit keys shared between the wireless AP (AP) and the wireless client was
the first-generation security solution to wireless authentication and encryption that IEEE offered.
WEP is based on the RC4 encryption algorithm (a stream cipher) and supports encryption up to
128 bits. Some vendors, such as Cisco Systems, supported both 40-bit and 128-bit keys on their
wireless devices; an example would be Cisco Aironet 128-bit devices. RC4 vulnerabilities, plus
the WEP usage of static keys, its weak authentication, and its nonscalable method of manually
configuring WEP keys on clients, soon proved to be unacceptable, and other solutions were
recommended.

To address the shortcomings of WEP, from 2001 to 2002, Cisco Systems offered a wireless
authentication and encryption solution that was initially called Lightweight Extensible
Authentication Protocol (LEAP). LEAP had negative connotations for some people; therefore,
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260 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Cisco Systems decided to rename it Cisco Wireless EAP. In brief, this solution offered the
following improvements over WEP:

■ Server-based authentication (leveraging 802.1x) using passwords, one-time tokens, Public
Key Infrastructure (PKI) certificates, or machine IDs

■ Usage of dynamic WEP keys (also called session keys) by reauthenticating the user
periodically and negotiating a new WEP key each time (Cisco Key Integrity Protocol, or
CKIP)

■ Mutual authentication between the wireless client and the RADIUS server

■ Usage of Cisco Message Integrity Check (CMIC) to protect against inductive WEP attacks
and replays

In late 2003, the Wi-Fi Alliance Group provided WPA as an interim wireless security solution until
the IEEE 802.11i standard becomes ready. WPA requires user authentication through preshared
key (PSK) or 802.1x (EAP) server-based authentication prior to authentication of the keys used.
WPA uses Temporal Key Integrity Protocol (TKIP) or per-packet keying, and message integrity
check (MIC) against man-in-the-middle and replay attacks. WPA uses expanded IV space of 48
bits rather than the traditional 24-bits IV. WPA did not require hardware upgrades and was
designed to be implemented with only a firmware or software upgrade.

In mid-2004, IEEE 802.11i/WPA2 became ready. The main improvements to WPA were usage of
Advanced Encryption Standard (AES) for encryption and usage of Intrusion Detection System
(IDS) to identify and protect against attacks. WPA2 is more CPU-intensive than WPA mostly
because of the usage of AES; therefore, it usually requires a hardware upgrade.

802.1x and EAP Authentication Protocols
IEEE developed the 802.1x standard, called Extensible Authentication Protocol (EAP), so that
LAN bridges/switches can perform port-based network access control. 802.1x was therefore
considered a supplement to the IEEE 802.1d standard. The 802.1x (EAP) standard was quickly
discovered and adopted for wireless LAN access control. Cisco Systems has supported the 802.1x
authentication since December 2000.

Cisco Systems, Microsoft, and other vendors have developed several variations of EAP; different
clients support one or more of those EAP varieties. 802.1x leverages many of the existing
standards. Following are a few of the important EAP features and benefits:

■ The RADIUS protocol with a RADIUS server can be used for AAA centralized authentication.
Users are authenticated based on usernames and passwords stored in an active directory
available in the network (based on RFC 2284). The RADIUS server or Cisco Access Control
Server (ACS) can use this directory. See Figure 9-1 in this chapter.
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802.1x and EAP Authentication Protocols 261

■ Authentication is mutual between the client and the authentication server (RADIUS Server).
The client software, which is required by the authentication protocols to participate in the
authentication process, is commonly referred to as a supplicant.

■ 802.1x can be used with multiple encryption algorithms, such as AES, WPA TKIP, and WEP.

■ Without user intervention, 802.1x uses dynamic (instead of static) WEP keys. These WEP
encryption keys are derived after authentication.

■ One-time password (OTP) can be used to encrypt plaintext passwords so that unencrypted
passwords do not have to be sent over insecure connections/applications such as Telnet and
FTP.

■ 802.1x supports roaming in public areas and is compatible with existing roaming
technologies.

■ Policy control is centralized, as is management of the user database.

The components that are required for 802.1x authentication are an EAP-capable client (the
supplicant), 802.1x-capable AP (the authenticator), and EAP-capable RADIUS server (the
authentication server). Optionally, the authentication server may use an external user database.
Figure 9-1 shows these components.

Figure 9-1 801.2x (EAP) Authentication Components
Supplicant Authenticator Authentication Server

EAP-Capable 802.1x-Capable EAP-Capable External
Client Access Point RADIUS Server User Database
(Optional)

The EAP-capable client requires an 802.1x-capable driver and an EAP supplicant. The supplicant
may be provided with the client card, be native in the client operating system, or be obtained from
the third-party software vendor. The EAP-capable wireless client (with the supplicant) sends
authentication credentials to the authenticator. The authenticator is usually located at the enterprise
edge, between the enterprise network and the public or semipublic devices. The authenticator
sends the received authentication credentials to the authentication server. The authentication
server refers to a user database to check the validity of the authentication credentials and to
determine the network access level of a valid user. Some examples of authentication servers are
Cisco Secure ACS, Microsoft IAS, and Meetinghouse Aegis. The local RADIUS database or an
external database such as Microsoft Active Directory can be used for authentication.
Authentication does not always use a RADIUS database or an external database; for example,
Cisco IOS can perform local authentication based on the usernames and passwords stored in a
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262 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

device configuration (running-config). Please note however that local authentication is neither a
scalable nor a secure authentication option.

EAP Authentication Protocols
802.1x does not provide LAN access to a client that is attempting access through a LAN switch
port or a wireless AP until the client has been authenticated. Many authentication protocols are
variations of EAP and work within the framework of 802.1x. The most popular protocols used in
Cisco wireless networking environments are briefly discussed in the following sections.

Cisco LEAP
Cisco LEAP is one of the 802.1x authentication types for WLANs and, like the other EAP types,
it is supported by Wi-Fi WPA and WPA2. Cisco LEAP supports strong mutual authentication
between the client and a RADIUS server using a logon password as the shared secret, and it
provides dynamic per-user, per-session encryption keys. Cisco LEAP is included with all Cisco
wireless products, Cisco Aironet products, and Cisco-compatible client devices.

Following are the important capabilities that LEAP provides, making it somewhat unique
compared to the other EAP variations:

■ Fast, secure roaming (Layer 2 and Layer 3) with Cisco or Cisco-compatible clients

■ True single login with an existing username and password using Windows NT/2000 Active
Directory (or Domain)

■ Support for a wide range of operating systems (such as Microsoft, Macintosh, Linux, and
DOS)

Following are the client operating systems that Cisco LEAP supports:

■ Microsoft Windows 98, XP, and CE

■ Mac OS (9.X or 10.X)

■ Linux (Kernel 2.2 or 2.4)

■ DOS

Following are the RADIUS servers and user databases that Cisco LEAP supports:

■ Cisco Secure ACS and Cisco Network (Access) Registrar

■ Meetinghouse Aegis

■ Interlink Merit
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802.1x and EAP Authentication Protocols 263

■ Funk Odyssey Server and Funk Steel-Belted

■ Products that use the Interlink Networks server code (such as LeapPoint appliances)

Following are the Cisco wireless devices that Cisco LEAP supports:

■ Cisco Aironet autonomous APs and LWAPs

■ Cisco WLAN controllers

■ Cisco Unified Wireless IP Phone 7920 handset

■ Workgroup bridges, wireless bridges, and repeaters

■ Many Cisco and Cisco-compatible WLAN client devices

Figure 9-2 displays the Cisco LEAP authentication process. A wireless client can only transmit
EAP traffic (no other traffic type) until a RADIUS server authenticates it. The authentication can
be initiated by the client Start message or by the AP Request/Identity message. Either way, the
client responds to the AP with a username. When the AP receives the username, it encapsulates it
in the Access Request message (a RADIUS message type) and sends it to the RADIUS server. In
the next two steps, the RADIUS server authenticates the client, and then the client authenticates
the RADIUS server through a challenge/response process (through the AP).

Figure 9-2 Cisco LEAP

Client Access Point RADIUS Server Windows
NT/AD
Controller
Start
Access Point Blocks All Requests
Request/Identity Until Authentication Completes

Identity Identity

RADIUS Server Authenticates Client

Client Authenticates RADIUS Server
Derive Derive
Key Key

Key Management
WPA or CCKM Key Management Used
Protected Data Session
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264 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

In the challenge/response process, one party sends a challenge (a randomly generated bit sequence)
to the other, and the other party sends a response back. The response is generated using an
algorithm such as MD5, which takes the challenge, plus a password that both parties share, and
perhaps other input such as a session ID. The benefit of the challenge/response process is that the
shared password is not sent from one party to the other.

When the RADIUS server and the client successfully authenticate each other, they submit a
Success (RADIUS) message to each other (through AP). Next, the RADIUS server and the client
generate a pairwise master key (PMK). The RADIUS server sends its PMK to the AP so that the
AP stores it locally for this particular client. Finally, the client and the AP, using the PMKs each
hold, perform a four-way handshake that allows them to exchange encrypted traffic and have a
protected data session.

EAP-FAST
Extensible Authentication Protocol-Flexible Authentication via Secure Tunneling (EAP-FAST)
was developed by Cisco Systems and submitted to the Internet Engineering Task Force (IETF) in
2004. Cisco LEAP requires use of strong passwords; for a customer who cannot enforce a strong
password policy and does not want to use certificates, migrating to EAP-FAST is a good solution
because it provides safety from dictionary attacks. EAP-FAST is standards based (nonproprietary)
and is considered flexible and easy to deploy and manage. Some of the main features and benefits
of EAP-FAST are as follows:

■ Supports Windows single sign-on for Cisco Aironet clients and Cisco-compatible clients

■ Does not use certificates or require PKI support on client devices but does provide for a
seamless migration from Cisco LEAP

■ Supports Windows 2000, Windows XP, and Windows CE operating systems

■ Provides full support for 802.11i, 802.1x, TKIP, and AES

■ Supports WPA and WPA2 authenticated key management on Windows XP and Windows
2000 client operating systems

■ Supports wireless domain services (WDS) and fast secure roaming with Cisco Centralized
Key Management (CCKM)

■ Supports password expiration or change (Microsoft password change)

EAP-FAST consists of three phases:

Phase 0 (provision PAC)—In this phase, the client is dynamically provisioned with a
Protected Access Credential (PAC) through a secure tunnel. Phase 0 is considered
optional, because PAC can be manually provided to the end-user client. PAC is used in
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802.1x and EAP Authentication Protocols 265

Phase 1 of EAP-FAST authentication. PAC consists of a secret part and an opaque part.
It has a specific user ID and an authority ID associated with it.
Phase 1 (establish secure tunnel)—In this phase, the Authentication, Authorization, and
Accounting (AAA) server (such as the Cisco Secure ACS v. 3.2.3) and the client use PAC
to authenticate each other and establish a secure tunnel.
Phase 2 (client authentication)—In this phase, the client sends its credentials to the
RADIUS server through the secure tunnel, and the RADIUS server authenticates the
client and establishes a client authorization policy.
Figure 9-3 displays the EAP-FAST authentication process. A wireless client can transmit only
EAP traffic (no other) until a RADIUS server authenticates it. First, the client sends an EAP over
LAN (EAPOL) start frame to the AP, and the AP returns a request/identity to the client.

Figure 9-3 EAP-FAST

Client Access Point RADIUS Server External
User
Database
Start
Access Point Blocks All Requests
Request/Identity Until Authentication Completes

Identity Identity
Authentication
Server-Side

Establish a Secure Tunnel (PAC and TLS)
A-ID A-ID

PAC-Opaque PAC-Opaque
Authentication
Client-Side

Server Authenticates Client

Key Management
WPA or CCKM Key Management Used
Protected Data Session

Next, the client sends its network access identifier (NAI) address to the AP, which in turn sends it
to the RADIUS server. The client and the server then perform mutual authentication using Phase
1 and Phase 2 of EAP-FAST process, and the RADIUS server sends a session key to the AP in a
Success packet.
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266 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

After that, the client and the RADIUS server negotiate and derive a session key. (This process
varies depending whether the client is using WEP or 802.11i.) The client and the AP use these keys
during this session.

At the end of the session, the client sends an EAPOL-logoff packet to the AP, returning it to the
preauthentication state (filtering all but EAPOL traffic).

EAP-TLS
Extensible Authentication Protocol-Transport Layer Security (EAP-TLS) uses the Transport
Layer Security (TLS) protocol. TLS is an IETF standard protocol that has replaced the Secure
Socket Layer (SSL) protocol. TLS provides secure communications and data transfers over public
domains such as the Internet, and it provides protection against eavesdropping and message
tampering. EAP-TLS uses PKI; therefore, the following three requirements must be satisfied:

■ The client must obtain a certificate so that the network can authenticate it.

■ The AAA server needs a certificate so that the client is assured of the server authenticity.

■ The certification authority server (CA) must issue the certificates to the AAA server(s) and
the clients.

EAP-TLS is one of the original EAP authentication methods, and it is used in many environments.
However, some customers are not in favor of using PKI and certificates for authentication purposes.
The supported clients for EAP-TLS include Microsoft Windows 2000, XP, and CE, plus non-
Windows platforms with third-party supplicants, such as Meetinghouse. EAP-TLS also requires a
supported RADIUS server such as Cisco Secure ACS, Cisco Access Registrar, Microsoft IAS,
Aegis, and Interlink. One of the advantages of Cisco and Microsoft implementation of EAP-TLS
is that it is possible to tie the Microsoft credentials of the user to the certificate of that user in a
Microsoft database, which permits a single logon to a Microsoft domain.

Figure 9-4 displays the EAP-TLS authentication process. The wireless client associates with the
AP using open authentication. The AP restricts (denies) all traffic from the client except EAP
traffic until the RADIUS server authenticates the client. First, the client sends an EAPOL start
frame to the AP, and the AP returns a request/identity to the client.
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802.1x and EAP Authentication Protocols 267

Figure 9-4 EAP-TLS

Client Access Point RADIUS Server CA
Start
Access Point Blocks All Requests
Request/Identity Until Authentication Completes

Identity Identity

Server Certificate Server Certificate

Client Certificate Client Certificate
Encrypted
Exchange

Random Session Keys Generated

Key Management
WPA Key Management Used
Protected Data Session

Second, the client sends its NAI address to the AP, which in turn sends it to the RADIUS server.
The client and the server then perform mutual authentication using an exchange of digital certificates,
and the RADIUS server sends a session key to the AP in a Success packet.

Third, the RADIUS server and the client negotiate and derive the session encryption; this process
varies depending on whether the client is using WEP or 802.11i. The client and the AP use these
keys during this session.

At the end of the session, the client sends an EAPOL-logoff packet to the AP, returning it to the
preauthentication state (filtering all but EAPOL traffic).

PEAP
Protected Extensible Authentication Protocol (PEAP) is yet another 802.1x authentication type for
WLANs, submitted by Cisco Systems, Microsoft, and RSA Security to the IETF as an Internet
Draft. With PEAP, only the server authentication is performed using PKI certificate; therefore,
installing digital certificates on every client machine (as is required by EAP-TLS) is not necessary.
The RADIUS server must have self-issuing certificate capability, you must purchase a server
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268 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

certificate per server from a PKI entity, or you must set up a simple PKI server to issue server
certificates.

PEAP works in two phases. In Phase 1, server-side authentication is performed, and an encrypted
tunnel (TLS) is created. In Phase 2, the client is authenticated using either EAP-GTC or EAP-
MSCHAPv2 within the TLS tunnel. The two implementations are called PEAP-GTC and PEAP-
MSCHAPv2. If PEAP-GTC is used, generic authentication can be performed using databases
such as Novell Directory Service (NDS), Lightweight Directory Access Protocol (LDAP), and
OTP. On the other hand, if PEAP-MSCHAPv2 is used, authentication can be performed using
databases that support MSCHAPv2, including Microsoft NT and Microsoft Active Directory.
PEAP-MSCHAPv2 supports single sign-on, but the Cisco PEAP-GTC supplicant does not
support single logon.

Figure 9-5 displays the PEAP authentication process. The wireless client associates with the AP
using open authentication. The AP restricts (denies) all traffic from the client except EAP traffic
until the RADIUS server authenticates the client.

Figure 9-5 PEAP

Client Access Point RADIUS Server External
User
Database
Start
Access Point Blocks All Requests
Request/Identity Until Authentication Completes

Identity Identity
Authentication
Server-Side

Server Certificate Server Certificate

Pre-Master Secret Pre-Master Secret

Encrypted Tunnel Established
Authentication
Client-Side

EAP in EAP Authentication

Key Management
WPA Key Management Used
Protected Data Session
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802.1x and EAP Authentication Protocols 269

As stated earlier, PEAP goes through two phases. As shown in Figure 9-5, in Phase 1, or the server-
side authentication phase, the client authenticates the server using a CA to verify the digital
certificate of the server. Then the client and server establish an encrypted tunnel. In Phase 2, or the
client-side authentication phase, the client submits its credentials to the server inside the TLS
tunnel using either EAP-GTC or EAP-MSCHAPv2.

Next, the RADIUS server sends the session key to the AP in a Success packet, and the RADIUS
server and client negotiate and derive a session encryption key. (This process varies depending
whether the client is using WEP or 80211i.) The client and the AP use the session key during this
session.

At the end of the session, the client sends an EAPOL-logoff packet to the AP, returning it to the
preauthentication state (filtering all but EAPOL traffic).

WPA, 802.11i, and WPA2
WPA is a standards-based security solution introduced by Wi-Fi Alliance in late 2003 to address
the vulnerabilities of the original 802.11 security implementations (WEP). The IEEE standard for
security, IEEE 802.11i was ratified in 2004.

The most important features/components of WPA that you need to know and remember are as
follows:

■ Authenticated key management—WPA performs authentication using either IEEE 802.1x
or PSK prior to the key management phase.

■ Unicast and broadcast key management—After successful user authentication, message
integrity and encryption keys are derived, distributed, validated, and stored on the client and
the AP.

■ Utilization of TKIP and MIC— Temporal Key Integrity Protocol (TKIP) and Message
Integrity Check (MIC) are both elements of the WPA standard, and they secure a system
against WEP vulnerabilities such as intrusive attacks.

■ Initialization Vector Space Expansion—WPA provides per-packet keying (PPK) via IV
hashing and broadcast key rotation. The IV is expanded from 24 bits (as in 802.11 WEP) to
48 bits.

Figure 9-6 displays the WPA (and 802.11i) authentication process. First, the client and the AP
exchange the initial association request (probe request) and agree to a specific security capability.
Next, the client and the authentication server (RADIUS server) perform the standard 802.1x
authentication. Upon successful authentication, the authentication server generates and sends a
master key to the AP; the client generates the same master key. These are called the PMK, which
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270 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

can be generated as a result of an 802.1x authentication process between the client and the server.
The PMK can also be generated based on a 64-HEX character PSK.

Figure 9-6 WPA and 802.11i Authentication and Key Management

Client Authenticator Authentication Server

Security Capability Discovery

802.1x Authentication

802.1x Key Management RADIUS-Based (PMK) Key Distribution

Four-Way Key Handshake

Two-Way Group Key Handshake

After completion of 802.1x authentication and 802.1x key management, the client and the AP
perform a Four-Way Key Handshake and exchange a nonce, a WPA information element, a
pairwise transient key (PTK), and MIC key information. This ensures validity of the AP and
creates a trusted session between the client and the AP.

The final step is the two-way key handshake that the client and the AP exchange. The purpose of
this handshake is to derive a group transient key (GTK), which provides a group key plus MIC
keys (used for checking data integrity).

Following are the main shortcomings and issues of WPA:

■ Even though WPA uses TKIP, which is an enhancement to 802.11 WEP, it relies on the RC4
encryption. (RC4 has known shortcomings.)

■ WPA requires AP firmware support, software driver support for wireless cards, and operating
system support (or a supplicant client). There is no guarantee that the manufacturers of all
these components that you own will release upgrades to support WPA. Furthermore, because
some vendors do not support mixing WEP and WPA (Wi-Fi Alliance does not support mixing
WEP and WPA either), an organization wanting to deploy WPA has to replace a significant
number of wireless infrastructure components.

■ WPA is susceptible to a specific DoS attack; if an AP receives two successive packets with
bad MICs, the AP shuts down the entire basic service set (wireless service) for one minute.
Furthermore, if small and noncomplex PSKs are used instead of 802.11i or EAP, an attacker
who performs dictionary attacks on captured traffic can discover them.
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802.1x and EAP Authentication Protocols 271

Less than a year after the release of WPA by Wi-Fi Alliance, IEEE ratified the 802.11i standard
(June 2004). 802.11i provides stronger encryption, authentication, and key management strategies
for wireless data and system security than its predecessor, 802.11 WEP. Following are the three
main components added by 802.11i:

■ 802.1x authentication

■ AES encryption algorithm

■ Key management (similar to WPA)

WPA2, the next generation or supplement to WPA, was developed by Wi-Fi Alliance and is
interoperable with IEEE 802.11i. WPA2 implements AES as per the National Institute of
Standards and Technology (NIST) recommendation, using Counter Mode with Cipher Block
Chaining Message Authentication Code Protocol (CCMP). Following are the key facts about
WPA2:

■ It uses 802.1x for authentication. (It also supports PSKs.)

■ It uses a similar method of key distribution and key renewal to WPA.

■ It supports Proactive Key Caching (PKC).

■ It uses Intrusion Detection System (IDS).

Because of the nature of RF medium, the wireless standards mandate that IDS works at physical
and data link layers. Wireless IDS addresses wireless and standards-based vulnerabilities with the
following capabilities:

■ Detect, locate, and mitigate rogue devices.

■ Detect and manage RF interference.

■ Detect reconnaissance.

■ Detect management frames and hijacking attacks.

■ Enforce security configuration policies.

■ Perform forensic analysis and compliance reporting as complementary functions.

WPA and WPA2 have two modes: Enterprise mode and Personal mode. Within each mode is an
encryption support and user authentication. Products that support both the PSK and the 802.1x
authentication methods are given the term Enterprise mode. Note that for 802.1x authentication,
an AAA/RADIUS server is required. Enterprise mode is targeted at medium to large medium to
large environments, such as education and government departments. Products that only support
PSK for authentication and require manual configuration of a PSK on the AP and clients are given
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272 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

the term Personal mode. (No authentication server is required.) Personal mode is targeted at small
business environments such as small office, home office (SOHO). Table 9-2 displays the
authentication and encryption methods that WPA and WPA2 use in Enterprise and Personal
modes.

Table 9-2 WPA/WPA2 Enterprise and Personal Modes

Mode WPA WPA2

Enterprise mode Authentication: IEEE 802.1x/EAP Authentication: IEEE 802.1x/EAP

Encryption: TKIP/MIC Encryption: AES-CCMP

Personal mode Authentication: PSK Authentication: PSK

Encryption: TKIP/MIC Encryption: AES-CCMP

Even though WPA2 addresses the security shortcomings of WPA, an enterprise must consider the
following WPA2 issues while evaluating and deciding to migrate to WPA2:

■ The wireless client (supplicant) must have a WPA2 driver that is EAP compatible.

■ The RADIUS server must support EAP.

■ Because WPA2 is more CPU-intensive than WPA (mostly due to usage of AES encryption),
hardware upgrades are often required (rather than just firmware upgrades).

■ Some older devices cannot be upgraded, so they might need to be replaced.

Configuring Encryption and Authentication on Lightweight
Access Points
In this section, you will learn how to navigate through the GUI of a WLC (Cisco WLC2006,
specifically) to configure encryption and authentication on a lightweight AP (Cisco AP1020,
specifically). The specific tasks shown are configuring open authentication, static WEP
authentication, WPA with PSK, web authentication, and 802.1x authentication.

Open Authentication
Open authentication means that you are interested neither in authenticating the client/user nor in
encrypting the data exchanged between the wireless client and the network. This type of setting is
often used in public places or hotspots such as airports, hotels, and lobbies for guest wireless
access (to the Internet, for example). To set up open authentication, open a web browser page to
your WLAN controller (using its name or IP address), log on, and click on the WLAN option on
the main toolbar.
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Configuring Encryption and Authentication on Lightweight Access Points 273

After you are on the WLAN page, you can set up a new wireless LAN by clicking on New or
change the settings on an existing WLAN by clicking on Edit beside the name of an existing
WLAN. The default method for authentication is 802.1x. This protects your WLAN against
accidentally setting it up with open authentication. Figure 9-7 shows the page that you will see if
you choose to modify an existing WLAN by clicking on Edit.

Figure 9-7 Configuring Open Authentication

As you can see in Figure 9-7, on the right side of the WLAN > Edit page is a drop-down list with
the title Layer 2 Security under the Security Policies section. To set up for open authentication,
you must select None from the drop-down list. (Remember that the default is 802.1x.)

Static WEP Authentication
To set up a WLAN for static WEP authentication, you must go to the WLAN > Edit page. On the
right side of this page, in the Security Policies section, select Static WEP from the Layer 2
Security drop-down list. After you select this option, the Static WEP options are displayed on the
bottom of this page. (See Figure 9-8.)
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274 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Figure 9-8 Configuring Static WEP Authentication

As you can see in Figure 9-8, a section with the Static WEP Parameters heading is displayed on
the bottom of the WLAN > Edit page. You can configure up to four keys using the Key Index
drop-down list. For each key, you can select its size from the Key Size drop-down list. In the
Encryption Key box, you can type the value for each key. For each key, you can select ASCII or
HEX as the key format from the Key Format drop-down list. Note that each WLAN is associated
to only one key index; therefore, with a maximum of four key indexes available from the drop-
down list, you can set up a maximum of four wireless LANs with the Static WEP option.

WPA Preshared Key
WPA PSK authentication is also configured on the WLAN > Edit page. From the Layer 2
Security drop-down list under the Security Policies section, you must select WPA (or WPA1 +
WPA2 depending on your software version). If you select the WPA1 + WPA2 (or WPA) option,
the appropriate fields for setting up the WPA parameters are displayed on the bottom of the WLAN >
Edit page, as shown in Figure 9-9.
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Configuring Encryption and Authentication on Lightweight Access Points 275

Figure 9-9 Configuring WPA PSK

NOTE Please note that in the figure that is in the ONT courseware, WPA is chosen from the
Layer 2 Security drop-down list, and the bottom of the page has a WPA Parameters section
instead. The reason for the discrepancy between Figure 9-9 of this book and the figure that is in
the ONT courseware is the software version difference on the wireless controller.

To set up for WPA PSK, under the WPA1 + WPA2 Parameters section, you must select the WPA1
Policy check box. For WPA1 encryption, you can choose either the AES or TKIP check box.
Next, from the Auth Key Mgmt drop-down list, you must select PSK. Finally, on the last line in
the WPA1 + WPA2 Parameters section, you must type the PSK in the long text box provided. Note
the PSK format drop-down list allows you to specify the format of the PSK as either ASCII or
HEX.

NOTE Again, the figure in the ONT courseware shows that after you select WPA from the
Layer 2 Security drop-down list, a WPA Parameters section displays on the bottom of the
WLAN > Edit page. Within that section, you are asked to click and enable a Pre-Shared Key
check box and then type a PSK in the long text box provided.
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276 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Web Authentication
To authenticate users through a web browser interface, you must configure web authentication and
its corresponding parameters. If a user has a web browser open (HTTP) and attempts to access the
WLAN, he is presented a login page. The login page is customizable; you can configure the logos
and the text on the login page. Web authentication is usually used for guest access; the data
exchanged between the wireless client and the AP is not encrypted, nor is there MIC or per-packet
authentication. Therefore, the client is open to attacks such as packet modification and hijacking.
As of the writing of this book, the web authentication feature is available on Cisco 4400 WLCs
and Cisco Catalyst 6500 Wireless Service Modules (WiSM), but it is not available on Cisco 2000
WLCs or Cisco Integrated Services Routers wireless LAN controller modules. With web authen-
tication, the maximum simultaneous authentication limit is 21; the total local web authentication
user limit is 2500.

To set up web authentication, you must navigate to the WLAN > Edit page. Under Security
Policies in the Layer 3 security section, you will find a Web Policy check box that you must enable
(see Figure 9-10).

Figure 9-10 Configuring Web Authentication

Below the Web Policy check box, you must choose between Authentication or Passthrough
options. If you select Authentication, the users are prompted for a username and password when
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Configuring Encryption and Authentication on Lightweight Access Points 277

they attempt to access the network. The username and password are verified against the internal
user database of WLC; if no match is found, the username and password are verified from an
external RADIUS server if one is configured. If Passthrough is selected, the user is not prompted
for a username and password; however, if the Email Input check box (which is beneath the
Passthrough option) is enabled, the users are prompted for their e-mail address. The last option
you have under Layer 3 security is selecting an access list from the Preauthentication ACL drop-
down list to be used against the traffic exchanged between the wireless client and the WLC.

To customize the login page for web authentication, you must click the Security option in the
main toolbar. From the security options listed on the left side of this page, click the Web Login
Page option. You are then presented with a page similar to the one shown in Figure 9-11.

NOTE In the ONT courseware, either because of a WLC hardware/software difference or
because of typing error, you are asked to go to Management > Web Login Page instead of
Security > Web Login Page.

Figure 9-11 Customizing the Web Login Page

As shown in Figure 9-11, on the Web Login Page, you have three choices for Web Authentication
Type: Internal (Default), Customized (Downloaded), and External (Redirect to external
server). If you choose the external or customized types, you must then enter a URL in the
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278 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Redirect URL after login box below. Otherwise, if you want the default authentication page of
the WLC, select the Internal (Default) option. The other options you have are selecting to show
or hide the Cisco logo, and entering a headline and a message. These options are only available if
you select external or customized web authentication types. An example for the headline would
be “AMIRACAN Inc. Wireless Network,” and an example for the message would be “Access is
only offered to authorized users. Please enter your username and password.”

802.1x Authentication
802.1x authentication is the default setting. To change the setting from other options back to
802.1x, you must navigate to the WLAN > Edit page and select 802.1x from the Layer 2 Security
drop-down list under the Security Policies section. After you select this option, on the bottom
of the WLAN > Edit page, a section with the 802.1x Parameters heading is displayed (see
Figure 9-12).

Figure 9-12 802.1x Authentication

Under the 802.1x Parameters section, you are presented with a drop-down list, giving you a choice
of None, 40 bits, 104 bits, and 128 bits WEP encryption for 802.11 data encryption. Note that
802.11 standards only support 40/64-bit and 104/128-bit keys; 128/152-bit keys are only supported
by 802.11i, WPA, and WPA2-compliant clients. It is also important to note that Microsoft
Windows XP clients only support 40-bit and 104-bit WEP keys.
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Configuring Encryption and Authentication on Lightweight Access Points 279

From the Layer 2 Security drop-down list under the Security Policies section, you can also select
WPA1 + WPA2. As stated earlier, on some hardware/software, WPA and WPA2 might be
presented as separate options. If you intend to use WPA with 802.1x, select the WPA1 + WPA2
(or WPA) option. In response, the WLAN > Edit page displays the WPA1 + WPA2 Parameters
section on the bottom, as shown in Figure 9-13.

Figure 9-13 WPA with 802.1x

Next, under the WPA1 + WPA2 Parameters section, enable the WPA1 Policy check box and
choose the AES or TKIP check box for WPA1 encryption. Finally, make sure you choose 802.1x
(not PSK) from the Auth Key Mgmt drop-down list so that the RADIUS server performs
authentication.

To configure a WLAN for WPA2 security with dynamic keys, from the Layer 2 Security drop-
down list under the Security Policies section, select WPA1 + WPA2. (On some hardware/
software, WPA and WPA2 might be presented as separate options.) In response, the WLAN > Edit
page displays the WPA1 + WPA2 Parameters section on the bottom, as shown in Figure 9-14.
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280 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Figure 9-14 WPA2 with Dynamic Keys

Next, under the WPA1 + WPA2 Parameters section, enable the WPA2 Policy check box and
choose the AES or TKIP check box for WPA2 encryption. Finally, make sure you choose 802.1x
(not PSK) from the Auth Key Mgmt drop-down list so that the RADIUS server performs
authentication.

If you enable both the WPA1 Policy and the WPA2 Policy check boxes, you are effectively setting
up your WLAN for WPA compatibility mode. WPA compatibility mode supports both WPA and
WPA2 clients and allows them to use the same SSID. Selecting both AES and TKIP for WPA2
Encryption allows support of legacy hardware that does support WPA2 but not AES.

NOTE In the ONT courseware, it shows that you can select WPA2 from the Layer 2 Security
drop-down list under the Security Policies section (instead of WPA1 + WPA2); this is because
of a hardware/software difference. Next, the ONT courseware states that a section titled WPA2
Parameters appears on the bottom of the WLAN > Edit page. In the WPA2 Parameters section,
you are then presented with the choice of enabling any of the following three options:
■ WPA2 Compatibility Mode

■ Allow WPA2 TKIP Clients
■ Pre-Shared Key
This note has been added so that you are prepared for a possible question in the certification
exam, should it be based on software/hardware variances.
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Foundation Summary 281

Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Following are the traditional wireless local-area network (WLAN) security issues:

■ Reliance on Service Set Identifier (SSID) as a security feature

■ Vulnerability to rogue access points (AP)

■ Reliance on MAC filters as a security feature

■ Usage of Wired Equivalent Privacy (WEP)

Following are the shortcomings of WEP:

■ The distribution of WEP keys to clients is not scalable.

■ WEP keys can be deducted if enough data is captured (even with IV).

■ WEP is vulnerable to dictionary attacks.

■ WEP does not provide protection against rogue APs.

The main features and benefits of 802.1x/EAP are as follows:

■ Usage of RADIUS server for AAA centralized authentication

■ Mutual authentication between the client and the authentication server

■ Ability to use 802.1x with multiple encryption algorithms, such as Advanced Encryption
Standard (AES), wireless protected access (WPA), Temporal Key Integrity Protocol (TKIP),
and WEP

■ Without user intervention, the ability to use dynamic (instead of static) WEP keys

■ Support of roaming
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282 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Following are the required components for 802.1x authentication:

■ EAP-capable client (the supplicant)

■ 802.1x-capable AP (the authenticator)

■ EAP-capable RADIUS server (the authentication server)

Table 9-3 displays important features of the main EAP variants discussed in this chapter.

Table 9-3 Comparison of Main EAP Variants

PEAP-
Feature Cisco LEAP EAP-FAST EAP-TLS PEAP-GTC MSCHAPv2

User authentication Windows NT Windows NT OTP, LDAP, OTP, LDAP, Windows NT
database and server domains, domains, Active Novell NDS, Novell NDS, domains,
Active Directory, LDAP Windows NT Windows NT Active
Directory (limited) domains, domains, Directory
Active Active
Directory Directory

Requires server No No Yes Yes Yes
certificates

Requires client No No Yes No No
certificates

Able to use single Yes Yes Yes No Yes
sign-on using
Windows login

Works with fast Yes Yes No No No
secure roaming

Works with WPA Yes Yes Yes Yes Yes
and WPA2

Following are the most important features/components of WPA:

■ Authenticated key management—WPA performs authentication using either IEEE 802.1x
or preshared key (PSK) prior to the key management phase.

■ Unicast and broadcast key management—After successful user authentication, message
integrity and encryption keys are derived, distributed, validated, and stored on the client and
the AP.

■ Utilization of TKIP and MIC— Temporal Key Integrity Protocol (TKIP) and Message
Integrity Check (MIC) are both elements of the WPA standard and they secure a system
against WEP vulnerabilities such as intrusive attacks.
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Foundation Summary 283

■ Initialization vector space expansion—WPA provides per-packet keying (PPK) via
initialization vector (IV) hashing and broadcast key rotation. The IV is expanded from 24 bits
(as in 802.11 WEP) to 48 bits.

The main shortcomings and issues of WPA are as follows:

■ Even though WPA uses TKIP, which is an enhancement to 802.11 WEP, it relies on the RC4
encryption. (RC4 has known shortcomings.)

■ WPA requires AP firmware support, software driver support for wireless cards, and operating
system support (or a supplicant client). It is not guaranteed that the manufacturers of all these
components that you own will release upgrades to support WPA.

■ WPA is susceptible to a specific denial of service (DoS attack); if an AP receives two
successive packets with bad MICs, the AP shuts down the basic service set for one minute.

■ If small and noncomplex PSKs are used instead of 802.11i or EAP, an attacker who performs
dictionary attacks on captured traffic can discover them.

Following are the key features of WPA2:

■ It uses 802.1x for authentication. (It also supports PSKs.)

■ It uses a similar method of key distribution and key renewal to WPA.

■ It supports Proactive Key Caching (PKC).

■ It uses Intrusion Detection System (IDS).

WPA and WPA2 have two modes: Enterprise mode and Personal mode. Each mode has encryption
support and user authentication. Table 9-4 displays the authentication and encryption methods that
WPA and WPA2 use in Enterprise and Personal modes.

Table 9-4 WPA/WPA2 Enterprise and Personal Modes

Mode WPA WPA2

Enterprise mode Authentication: IEEE 802.1x/EAP Authentication: IEEE 802.1x/EAP

Encryption: TKIP/MIC Encryption: AES-CCMP

Personal mode Authentication: PSK Authentication: PSK

Encryption: TKIP/MIC Encryption: AES-CCMP
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284 Chapter 9: Introducing 802.1x and Configuring Encryption and Authentication

Following are some of the issues that an enterprise must consider while evaluating and deciding
to migrate to WPA2:

■ The wireless client (supplicant) must have a WPA2 driver that is EAP compatible.

■ The RADIUS server must support EAP.

■ Because WPA2 is more CPU-intensive than WPA (mostly due to usage of AES encryption),
hardware upgrades are often required (rather than just a firmware upgrade).

■ Some older devices cannot be upgraded, so they might need to be replaced.

To set up or change the authentication and encryption settings for your WLANS (LWAPs), open
a web browser page to your WLAN controller (using its name or IP address), log on, and click
on the WLAN option on the main toolbar. Next, click on Edit for an existing WLAN; the
WLAN > Edit page appears. The Security Policies section on the WLAN > Edit page allows
you to set up Layer 2 and Layer 3 security settings.
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Q&A 285

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. What is a rogue access point, and what are its dangers?
2. Specify at least two weaknesses of basic 802.11 (WEP) security.
3. Specify at least two benefits of LEAP over the basic 802.11 (WEP).
4. Specify at least one benefit and one drawback of WPA2 over WPA.
5. Provide at least three important features and benefits of 802.1x/EAP.
6. What are the required components for 802.1x authentication?
7. What is the role of EAP client supplicant?
8. Specify at least three of the main features and benefits of EAP-FAST.
9. What are the three phases of EAP-FAST?
10. Provide at least two important features or facts about EAP-TLS.
11. Provide at least two important features or facts about PEAP.
12. Specify at least two important features of WPA.
13. What are the three key security features that the 802.11i standard has offered?
14. Provide at least two important features/facts about WPA2.
15. List at least three services that wireless IDS provides to address RF and standards-based
vulnerabilities.
16. What are the two modes of WPA and WPA2?
1763fm.book Page 286 Monday, April 23, 2007 8:58 AM

This chapter covers the
following subjects:

■ The Need for WLAN Management

■ CiscoWorks Wireless LAN Solution Engine

■ Cisco Wireless Control System
1763fm.book Page 287 Monday, April 23, 2007 8:58 AM

CHAPTER 10
WLAN Management

This chapter provides an understanding of the network manager’s tools to discover, configure,
and monitor the various components in a WLAN solution. Cisco offers autonomous and
lightweight access points (LWAP), which can both be centrally managed. Centralization
simplifies WLAN management and improves scalability. Lightweight access points and their
associated controllers can be managed using the Cisco Wireless Control System (WCS).
Autonomous access points can be managed using the CiscoWorks Wireless LAN Solution
Engine (WLSE).

Additional capabilities are available when centrally managing lightweight access points, for
example, when WCS brings real-time device location tracking to life using the Cisco RF
Fingerprinting technology. This is one of the many benefits customers can experience using
WCS.

“Do I Know This Already?” Quiz
The purpose of the “Do I Know This Already?” quiz is to help you decide whether you really
need to read the entire chapter. The 10-question quiz, derived from the major sections of this
chapter, helps you determine how to spend your limited study time.

Table 10-1 outlines the major topics discussed in this chapter and the “Do I Know This
Already?” quiz questions that correspond to those topics.

Table 10-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section Covering These Questions Questions Score
“The Need for WLAN Management” 1–4
“CiscoWorks Wireless LAN Solution Engine” 5–8
“Cisco Wireless Control System” 9–13
Total Score (13 possible)
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288 Chapter 10: WLAN Management

CAUTION The goal of self-assessment is to gauge your mastery of the topics in this chapter.
If you do not know the answer to a question or are only partially sure of the answer, mark this
question wrong for purposes of the self-assessment. Giving yourself credit for an answer you
correctly guess skews your self-assessment results and might provide you with a false sense of
security.

You can find the answers to the “Do I Know This Already?” quiz in Appendix A, “Answers to the
‘Do I Know This Already?’ Quizzes and Q&A Sections.” The suggested choices for your next step
are as follows:

■ 9 or less overall score—Read the entire chapter. This includes the “Foundation Topics,”
“Foundation Summary,” and “Q&A” sections.

■ 10–11 overall score—Begin with the “Foundation Summary” section and then follow up
with the “Q&A” section at the end of the chapter.

■ 12 or more overall score—If you want more review on this topic, skip to the “Foundation
Summary” section and then go to the “Q&A” section. Otherwise, proceed to the next chapter.

1. The Cisco Unified Wireless Network unique approach addresses all layers of the WLAN
network through what five interconnected elements?
a. Access points, mobility platform, network unification, world-class network manage-
ment, and unified advanced services
b. Client devices, access points, mobility platform, network unification, and unified
advanced services
c. Client devices, access points, network unification, world-class network management,
and unified advanced services
d. Client devices, mobility platform, network unification, world-class network manage-
ment, and unified advanced services
2. What are the two Cisco WLAN implementations?
a. Centralized and decentralized
b. Thick and thin
c. Autonomous and lightweight
d. None of the above
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“Do I Know This Already?” Quiz 289

3. Control and radio monitoring is accomplished based on which of the following?
a. The controller providing power mitigation
b. The end solution being autonomous or lightweight
c. Both A and B
d. Neither A nor B
4. Centralized WLAN management is performed by using which of the following?
a. CiscoWorks WLSE for both lightweight and autonomous implementations
b. Cisco WCS for both lightweight and autonomous implementations
c. CiscoWorks WLSE for autonomous implementations and Cisco WCS for lightweight
implementations
d. Cisco WCS for autonomous implementations and CiscoWorks WLSE for lightweight
implementations
5. CiscoWorks WLSE discovery process requires routers, switches, and access points to be
properly configured with what protocol(s)?
a. SNMP and LWAPP
b. CDP and LWAPP
c. CDP and SNMP
d. LWAPP only
6. CiscoWorks WLSE Express supports what modes of setup?
a. Manual only
b. Manual and Automatic
c. Automatic only
d. Manual, Automatic, and Assisted
7. What are the two features of WLSE that enforce optimization and high availability?
a. Auto Re-Site Survey and Assisted Site Survey
b. ACLs and QoS
c. ACLs and MTU
d. Assisted Site Survey and QoS
8. What are the two versions of CiscoWorks for WLANs?
a. Demo and Registered
b. Registered and WLSE Express
c. WLSE and WLSE Express
d. WLSE and Demo
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290 Chapter 10: WLAN Management

9. How many Cisco wireless LAN controllers and access points is the Cisco WCS designed to
handle?
a. 50 Cisco wireless LAN controllers and 1500 access points
b. 1 Cisco wireless LAN controller and 50 access points
c. 0 Cisco wireless LAN controllers and 500 access points
d. 50 Cisco wireless LAN controllers and 500 access points
10. How many versions of the Cisco WCS exist?
a. 1
b. 2
c. 3
d. Depends on license purchased
11. True or False: Some Cisco WCS features might not function properly if you use a web
browser other than Internet Explorer 7.0 on a Windows workstation.
a. True
b. False
12. What is the most secure way to configure controller management?
a. Through the controller-dedicated service port
b. Through the console cable
c. Through SNMP v3
d. Through SSH
13. When Cisco wireless LAN controller detects a rogue access point, what does it immediately
do?
a. Sends an SNMP trap
b. Sends an e-mail notification to the recipient entered in the controller
c. Notifies Cisco WCS, which creates a rogue access point alarm
d. Both A and B
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The Need for WLAN Management 291

Foundation Topics

The Need for WLAN Management
WLAN management is one piece of a puzzle for network managers to understand. WLANs
address the business drivers such as mobile users, Wi-Fi enabled notebooks, and anytime,
anywhere access. WLAN management helps Network Managers plan for scalable WLANs that
are both centralized and secure.

WLAN management within the Cisco Unified Wireless Network is composed of five elements.
Those elements are fundamental to building successful enterprise-class WLANs that are scalable,
centralized, and secure.

Cisco Unified Wireless Networks
The Cisco Unified Wireless Network is a total-enterprise solution composed of five
comprehensive elements. The Cisco Unified Wireless Network enables the use of advanced
wireless services and addresses security concerns. It also addresses deployment, control, and the
management of WLAN components and RF.

Following are the five elements of Cisco Unified Wireless Network:

■ Client devices—Use the Cisco Compatible Extensions program to help ensure
interoperability. The Cisco Compatible Extensions program delivers services such as wireless
mobility, QoS, network management, and enhanced security.

■ Mobility platform—Provides ubiquitous access in any environment indoors or out. The
LWAPs are dynamically configured and managed by wireless LAN controllers (WLC)
through LightWeight Access Point Protocol (LWAPP).

■ Network unification—Creates seamless integration into the routing and switching
infrastructure. The WLCs are responsible for functions such as RF management, n+1
deployment, and Intrusion Prevention System (IPS).

■ World-class network management—Enables WLANs to have the equivalent LAN security,
scalability, reliability, ease of deployment, and management via Cisco Wireless Control
System (WCS). Cisco WCS provides features for design, control, and monitoring.
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292 Chapter 10: WLAN Management

■ Unified advanced services—Support new mobility applications, emerging Wi-Fi
technologies, and advanced threat detection and prevention capabilities such as wireless VoIP,
future unified cellular, location services, Network Admission Control (NAC), the Self-
Defending Network, Identity Based Networking Services (IBNS), Intrusion Detection
Systems (IDS), and guest access.

Following are Cisco WLAN products supporting the Cisco Unified Wireless Network:

■ Client devices—These include the Cisco 7920 IP Phone, PDAs, and client cards for
notebooks. Cisco client device compatibility is higher than 90 percent, reducing conflicts or
issues.

■ Mobility platform—Lightweight access points (AP) include the 1500, 1300, 1240AG,
1230AG, 1130AG, and 1000. Bridges include the 1400 and 1300.

■ Network unification—WLCs include the 4400 and 2000. Catalyst devices include the 6500
WiSM, ISR, and 3750 integration.

■ World-class network management—Cisco WCS provides features for design, control, and
monitoring.

■ Unified advanced services—Cisco Wireless Location Appliance, WCS, Self-Defending
Network (SDN), NAC, Wi-Fi phones, and RF firewalls.

Cisco WLAN Implementation
Cisco offers two WLAN implementations. The first is the autonomous WLAN solution based on
autonomous APs, and the second is the lightweight WLAN solution based on LWAPs and WLCs.
Table 10-2 compares the two WLAN solutions.

Table 10-2 Comparison of WLAN Implementation Solutions

Autonomous WLAN
Category Solution Lightweight WLAN Solution
Access Point Autonomous APs LWAPs
Control Individual configuration on each Configuration via Cisco WLC
AP
Dependency Independent operation Dependent on Cisco WLC
WLAN Management Management via CiscoWorks Management via Cisco WCS
WLSE and Wireless Domain
Services (WDS)
Redundancy AP redundancy Cisco WLC redundancy
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The Need for WLAN Management 293

The two WLAN solutions have different characteristics and advantages:

■ Autonomous APs— Configuration is accomplished on each AP. Each AP places RF control,
security, and mobility functions within the local configuration. Individual configuration is
required because each AP operates independently. However, centralized configuration,
monitoring, and management can be done through CiscoWorks WLSE. WDS provides the
radio monitoring and management communication between the autonomous APs and
CiscoWorks WLSE.

■ LWAPs—Configuration, monitoring, and security are accomplished via the WLAN controller.
The LWAPs depend on the controller for control and data transmission. However, Remote-
Edge Access Point (REAP) mode does not need the controller for data transmission. Cisco
WCS can centralize configuration, monitoring, and management. Cisco WLAN controllers
can be implemented with redundancy within the WLC groups.

Without centralized WLAN management both implementations eventually have scalability issues.
However, LWAPs and their associated WLAN Controllers provide a more scalable solution for
WLANs than autonomous APs. In fact, the growth and management of autonomous APs becomes
an important concern since independently managing APs increases operational costs and staffing
requirements. Moreover, correlating and forecasting across the enterprise WLAN becomes more
difficult due to the lack of visibility and/or personnel time. Client handoff times decrease between
APs and real-time applications such as voice and video start to suffer.

Security starts to lose effectiveness because of the growth and no centralized management.
Detection and mitigation of denial of service (DoS) attacks across an entire WLAN are not
possible. Interferences cannot be viewed on a systemwide basis because of the lack of centralized
management. Each autonomous AP is a single point of enforcement for security policies across
Layer 1, Layer 2, and Layer 3. Security is at risk when an AP is stolen or compromised because
the passwords, keys, and community strings all reside within the local configuration.

Regardless of which implementation is chosen, Cisco provides a centralized WLAN management
solution.
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294 Chapter 10: WLAN Management

WLAN Components
Figure 10-1 provides a clear hierarchy of the components that are required to build a WLAN.

Figure 10-1 WLAN Components

Autonomous Lightweight
Wireless Clients
Solution Solution

Autonomous Lightweight
Access Points
Access Points Access Points

Wireless Domain Cisco Wireless
Control
Services (WDS) Controller

Cisco Wireless
Cisco Wireless Solution WLAN
Control
Engine (WLSE) Management
System (WCS)

PoE Switches, Network PoE Switches,
Routers Infrastructure Routers

DHCP, DNS, DHCP, DNS,
Network Services
AAA AAA

Client devices are the most obvious of the WLAN components. Client devices come in many
forms such as PDAs, IP phones, notebooks, and bar-code scanners.

Access Points are another obvious WLAN component—either autonomous or lightweight. The
APs are used to build the WLAN infrastructure. Configuration is performed independently on the
autonomous APs. Lightweight APs are configured through their associated LAN controller.

Control is the WLAN component that provides device control and radio monitoring. Control and
radio monitoring are specific to the end solution implementation. The autonomous AP solution
uses Wireless Domain Services (WDS). All WDS configured APs aggregate their information
through WDS which sends it to the WLSE. The lightweight APs use their associated LAN
controllers via LWAPP.

WLAN management is the WLAN component that addresses how large-scale deployments are
centrally managed. Autonomous APs use CiscoWorks WLSE and lightweight APs use Cisco WCS
management.

The network infrastructure WLAN component includes the routers and switches that interconnect
all the APs, controllers, management, and servers together.
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CiscoWorks Wireless LAN Solution Engine 295

Network services is the last WLAN component in Figure 10-1. Network services function to
provide services such as Dynamic Host Configuration Protocol (DHCP), Domain Name System
(DNS), and Authentication, Authorization and Accounting (AAA)—DHCP, DNS, and AAA.

NOTE Cisco Aironet bridges operate at the MAC address layer (data link layer).

CiscoWorks Wireless LAN Solution Engine
CiscoWorks WLSE is part of the CiscoWorks network management products. CiscoWorks WLSE
provides centralized management for autonomous APs. WLANs benefit from the WLSE major
features such as configuration, fault and policy monitoring, reporting, firmware, and radio
management. In addition, the RF and device-management features help reduce operating expenses
and deployment. CiscoWorks WLSE covers fault, configuration, and performance management,
which are three of the FCAPS (Fault, Configuration, Accounting, Performance, and Security)
management tools. Proper Cisco Discovery Protocol (CDP) and Simple Network Management
Protocol (SNMP) configuration on all switches, routers, WDS, and APs is required for the
CiscoWorks WLSE discovery process to work. After the devices are discovered, a decision is
required on whether to manage them through CiscoWorks WLSE.

WLSE Software Features
Network management of system-wide autonomous APs through CiscoWorks WLSE has these
major software features:

■ Configuration—One CiscoWorks WLSE console supports up to 2500 APs. Configuration
changes can be performed in mass, individually, or in defined groups as desired or on a
schedule time. All Cisco Aironet APs are supported.

■ Fault and policy monitoring—WLSE monitors device faults and performance threshold
conditions such as memory, CPU, associations, Lightweight Extensible Authentication
Protocol (LEAP) server responses, and policy configuration errors.

■ Reporting—WLSE provides the capability to e-mail, print, and export reports. Client,
device, and security information can all be tracked and reported.

■ Firmware—WLSE performs centralized firmware upgrades. Upgrades can be done in mass,
individually, or in defined groups as desired or on a scheduled time.

■ Radio management—WLSE assists in management of the WLAN radio environment. Radio
management features include parameter generation, network status, and reports.
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■ CiscoWorks WLSE administration—WLSE administration includes status by means of
WLSE log files, software (WLSE system software), security (authentication modules, SSH,
Telnet access), backup and restore (WLSE data), diagnostics (WLSE test and reports),
connectivity tools, and redundancy (managment of redundant WLSEs.) Two WLSE devices
can create a highly available WLAN management solution. CiscoWorks WLSE supports
warm-standby redundancy.

■ Deployment wizard—WLSE provides a deployment wizard that discovers, uploads
configurations, and manages all deployed APs.

NOTE You can configure a CiscoWorks WLSE backup server to take over wireless
management if there is a primary CiscoWorks WLSE failure.

WLSE Key Benefits
Managing autonomous APs and bridges through CiscoWorks WLSE provides centralized
management and RF visibility for the WLAN. This provides many key benefits, such as the
following:

■ Improved WLAN security—Wireless IDS with rogue AP detection handles security threats
such as malicious intruders, ad hoc networks, excess 802.11 management frames that signal
denial-of-service (DoS) attacks, and man-in-the-middle attacks.

■ Simplified AP deployment—Deployment Wizards automatically apply configuration
policies to new APs.

■ RF visibility—WLSE provides information and displays to show RF coverage, received
signal strength indicator (RSSI) displays, rogue AP location, and roaming boundaries of the
WLAN.

■ Dynamic RF management—WLSE offers self-healing, assisted site survey, automatic re-
site survey, and interference detection capabilities within the WLAN.

■ Simplified operations—Threshold-based monitoring, reporting, template-based con-
figuration, and image updates are all features designed to simplify operations.

CiscoWorks WLSE and WLSE Express
Two versions of CiscoWorks WLSE are available based on the network sizes: WLSE and WLSE
Express.

WLSE is for medium to large enterprise WLAN solutions with up to 2500 managed devices.
WLSE requires an external AAA server such as a Cisco ACS server since the WLSE does not
include one.
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CiscoWorks WLSE Express includes AAA providing security services that support 802.1x LEAP,
Protected Extensible Authentication Protocol (PEAP), Extensible Authentication Protocol-
Flexible Authentication via Secure Tunneling (EAP-FAST), and Extensible Authentication
Protocol-Transport Layer Security (EAP-TLS). The user directory supports Lightweight
Directory Access Protocol (LDAP), Microsoft Active Directory, and a local user database. In
addition, user authentication mechanisms are supported for both wired and wireless. WLAN IDS
features are also supported.

WLSE Express is designed for small to medium businesses with up to 100 WLAN devices. In
addition, service providers with public WLAN (PWLAN) hot spot management would use WLSE
Express because of the smaller number of devices.

CiscoWorks WLSE and CiscoWorks WLSE Express both support the following WLAN devices:

■ Cisco Aironet autonomous APs and bridges

■ AP- and Cisco Catalyst Series Wireless LAN Services Module (WLSM)-based WDS

CiscoWorks WLSE and CiscoWorks WLSE Express both support the following protocols:

■ Secure Shell (SSH)

■ HTTP

■ Cisco Discovery Protocol (CDP)

■ Simple Network Management Protocol (SNMP)

■ CiscoWorks WLSE and CiscoWorks WLSE Express both integrate with CiscoWorks wired
management tools and third-party NMSs. Fault notification and forwarding can be integrated
via SNMP traps and syslog messages. In addition, CiscoWorks WLSE and CiscoWorks
WLSE Express both provide the ability to export data via Simple Object Access Protocol
(SOAP) Extensible Markup Language (XML) application programming interface (API).

Simplified WLSE Express Setup
CiscoWorks WLSE Express supports two modes of setup:

■ Automatic—DHCP is enabled by default. DHCP options 66 and 67 provide the TFTP IP
address and filename. A special configuration file can be downloaded automatically, making
the WLSE Express ready for use.

■ Manual—CiscoWorks WLSE Express can be manually configured with setup scripts and by
entering CLI commands.
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WLSE Configuration Templates
CiscoWorks WLSE supports performance optimization and high availability beyond the basic
configuration and monitoring. The configuration is performed through a browser or web-based
GUI. Templates ease the configuration and deployment of the WLAN environment. Several
templates exist, such as these:

■ Plug-and-play deployment

■ Automatic configuration of APs added to CiscoWorks WLSE

■ Automatic RF configuration of APs

■ Calculation of optimal RF configurations by APs

WLSE IDS Features
CiscoWorks WLSE includes intrusion detection features, such as these:

■ Rogue APs are automatically shut down when they are detected and located by disabling the
switch ports.

■ Ad hoc network devices are detected in addition to rogue APs.

■ Man-in-the-middle attacks are detected via Message Integrity Check (MIC) failures.

■ AP configuration monitoring ensures that security policies are always enforced.

■ Sensor-mode APs can add enhanced features to the WLAN.

WLSE Summary
All the features CiscoWorks WLSE offers help improve the day-to-day WLAN management.
CiscoWorks WLSE is a solution providing performance optimization and high availability for
autonomous WLAN networks. Following are two features of WLSE that enforce optimization and
high availability:

■ Auto re-site survey—This feature can optimize the WLAN environment by selecting a more
effective channel and adjusting the power levels. The most effective results come from performing
a client walkabout during the assisted site survey. The assisted site survey is highly recom-
mended but not required.

■ Self-healing—This feature allows CiscoWorks WLSE to detect AP failures and compensate
by automatically increasing the power and cell coverage of the others nearby. Moreover, when
the AP comes back, it recalculates the power and channel selections. This minimizes the client
impact and maintains availability.
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CiscoWorks WLSE supports centralized configuration, firmware, and radio management. These
can save time and resources normally required to operate large deployments of APs not centrally
managed. Moreover, CiscoWorks WLSE pulls all the configurations, images, and manage-
ment information into one location. The templates simplify large-scale implementations by auto-
configuration of new APs. The security policies minimize the security vulnerabilities due to rogue
APs and misconfigurations. Upon detection, CiscoWorks WLSE sends out an alert. CiscoWorks
WLSE is capable of monitoring AP utilization and client association and reporting the information
to help in capacity planning and troubleshooting. CiscoWorks WLSE proactively monitors APs,
bridges, and 802.1x EAP servers and provides improved WLAN uptime. Table 10-3 briefly
summarizes this information.

Table 10-3 CiscoWorks WLSE Features and Benefits

Feature Benefit
Centralized configuration, firmware, and radio Reduces WLAN total cost of ownership by saving
management time and resources required to manage large
numbers of APs
Autoconfiguration of new APs Simplifies large-scale deployments
Security policy misconfiguration alerts and Minimizes security vulnerabilities
rogue AP detection
AP utilization and client association reports Helps in capacity planning and troubleshooting
Proactive monitoring of APs, bridges, and Improves WLAN uptime
802.1x EAP servers

Cisco Wireless Control System
Cisco WCS is an advanced centralized WLAN solution for LWAPs. It provides configuration,
firmware, radio management, and IDS for LWAP and their associated controllers. The same
configuration, performance monitoring, security, fault management, and accounting options found
on the individual controllers also exist on the WCS. It is designed to support 50 Cisco WLCs and
1500 APs.

Administrators can define operator permissions within the administration menu where accounts
and maintenance tasks are located. Features like autodiscovery help simplify configuration and
reduce data entry errors. WCS administration is accessible via HTTPS and supports SNMPv1,
SNMPv2, and SNMPv3. Cisco WCS uses SNMP for controller communications.

WCS runs on both Microsoft Windows and Linux platforms. The WCS implementation can either
be run as a normal application or as a service that is always running even after reboot.
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Cisco WCS has three versions:

■ WCS Base

■ WCS Location

■ WCS Location + 2700 Series Wireless Location Appliance

WCS Location Tracking Options
The three WCS tracking options are increasingly enhanced with features. Tracking refers to the
management of wireless assets and how each version can help improve on that task.

The simplest version of Cisco WCS, WCS Base, informs managers which AP a device is
associated with. This allows managers to have an approximation of the device location. The
optional version, called WCS Location, is the second level of WCS. It provides users with the RF
fingerprinting technology and can provide location accuracy to within a few meters (less than 10
meters 90 percent of the time; less than 5 meters 50 percent of the time). The third and final option,
the one with the most capabilities, is called WCS Location + 2700 Series Wireless Location
Appliance. The WCS Location + 2700 Series Wireless Location Appliance provides the capability
to track thousands of wireless clients in real time.

With these advanced location-tracking capabilities, the Cisco Unified Wireless Network is an ideal
platform for helping to enable key business applications that take advantage of wireless mobility,
such as asset tracking, inventory management, and enhanced 911 (e911) services for voice. By
incorporating indoor location tracking into the wireless LAN infrastructure itself, Cisco reduces
the complexities of wireless LAN deployment and minimizes total cost of ownership.

WCS Base Software Features
Cisco WCS Base is a full-featured software product for WLAN monitoring and control. Wireless
client data access, rogue AP detection to the nearest Cisco AP, and containment are examples that
are offered in Cisco WCS Base.

Cisco WCS graphical views provide the following:

■ Autodiscovery of APs as they associate with controllers

■ Autodiscovery and containment or notification of rogue APs

■ Map-based organization of AP coverage areas
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■ User-supplied campus, building, and floor plan graphics that provide locations and status of
managed APs, RF coverage maps as well as location to the nearest AP, and coverage hole
alarms

Cisco WCS Base also provides system-wide control of the following:

■ Configuration for controllers and managed APs using customer-defined templates

■ Status and alarm monitoring of all managed devices with automated and manual client
monitoring and control functions

■ Automated monitoring of rogue APs, coverage holes, security violations, controllers, and APs

■ Event log information for data clients, rogue APs, coverage holes, security violations,
controllers, and APs

■ Automatic channel and power level assignment using radio resource management (RRM)

■ User-defined audit status, missed trap polling, configuration backups, and policy cleanups

WCS Location Software Features
Cisco WCS Location includes the WCS Base features with some enhancements. WCS Location
has the ability to use the historical location data management of the location appliance. WCS
Location also features the on-demand monitoring of any single device using RF fingerprinting
technology, providing high location accuracy. Any rogue AP, client, or device tracking can be
performed on-demand within 10 meters or 33 feet using RF fingerprinting.

WCS Location + 2700 Series Wireless Location Appliance Features
Cisco Wireless Location Appliance scales on-demand location tracking to a new level, significantly
improving the functionality of Cisco WCS Location. Whereas WCS Location could track one
on-demand device, the Cisco Wireless Location Appliance can track up to 1500 devices simul-
taneously. It can record historical information that can be used in capacity management and
trending.

WCS System Features
The Cisco WCS operating system manages all data client, communications, and system admin-
istration functions and performs radio resource management (RRM) functions. Moreover, WCS
manages systemwide mobility policies using the operating systems security solution and
coordinates all security functions using the operating system security framework.
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Cisco WCS User Interface
Three user interfaces exist for Cisco WCS. The first is a full featured CLI that can be used to
configure and monitor individual controllers. The second interface is the industry standard SNMP.
Cisco WCS supports SNMPv1, SNMP 2c, and SNMPv3. The third interface is a full featured
HTTPS web browser interface. It is hosted by Cisco controllers and can be used to configure and
monitor individual controllers and their associated access pointAPs.

The Cisco WCS user interface is where the administrator can create, modify, and delete user
accounts; change passwords; assign permissions; and schedule periodic maintenance tasks. The
administrator creates usernames and passwords, assigning them to predefined permissions groups.

In addition, the administrator can configure operating system parameters, monitor real-time
operations, create and configure coverage area layouts, and perform troubleshooting tasks via
HTTPS.

The user interface has four menus on each screen. A general description of each menu function
follows:

■ Monitor—See a top-level description of all devices

■ Configure—Configure APs, controllers, and templates

■ Administration—Schedule tasks such as backups, device status, network audits, and
location server synchronization

■ Location—Configure the Cisco Wireless Location Appliances

Cisco WCS System Requirements
Cisco WCS is supported under Microsoft Windows 2000, Windows 2003, and Red Hat Enterprise
Linux ES v.3 servers as either a normal application or a service.

Minimum server requirements are as follows:

■ Windows 2000 Service Pack 4 (SP4) or greater, Windows 2003 SP1 or greater, or Red Hat
Enterprise Linux ES v.3

■ Up to 500 APs: 2.4-GHz Pentium with 1-GB RAM

■ More than 500 APs: dual processors (at least 2.4 GHz each) with minimum 2-GB RAM

■ 20-GB hard drive

NOTE The minimum client requirement is Internet Explorer 6.0 with SP1 or later.
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WCS Summary Pages
The WCS Network Summary (Network Dashboard) page is displayed after logging in success-
fully. It is a top-level overview of the network with information about controllers, coverage
areas, APs, and clients. Systems configuration and devices can be added from this page. Access
the Network Summary page from other areas by choosing Monitor > Network Summary.
Figure 10-2 shows a sample WCS Network Summary page.

Figure 10-2 WCS Network Summary Page

The Network Summary page is an at-a-glance view that is ideal for an operational monitoring
environment. In the lower-left portion of the page is the Alarm Monitor, which shows the received
alarms from all the controllers. The Alarm Monitor reflects the current state of alarms needing
attention. They are usually generated by one or more events. Alarms can be cleared, but the event
remains. The alarm color codes are given in Table 10-4.

Table 10-4 WCS Alarm Color Codes

Color Code Type of Alarm
Clear No alarm
Red Critical alarm
Orange Major alarm
Yellow Minor alarm
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The WCS Controller Summary page provides visibility for the supported 50 Cisco WLCs and
1500 APs. To access this page, select Monitor > Devices > Controllers. (The Monitor Controllers
> Search Results page is the default.) The WCS Controller Summary page provides detailed
information about the specific controller, such as the IP address, controller name, location,
mobility group name, and reachability. Figure 10-3 shows a sample WCS Controller Summary
page.

Figure 10-3 WCS Controller Summary Page

Wireless Location Appliance
The Cisco Wireless Location Appliance is part of the Cisco Unified Wireless Network using LAN
Controllers and LWAPs that can track the location of devices to within a few meters. The Cisco
Wireless Location Appliance solution uses an advanced technology called RF fingerprinting to
track thousands of devices, increasing visibility and control of the airspace. The RF fingerprinting
technology performs location computations using the site survey results and the RSSI information
to improve the location accuracy over other location methods.
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The appliance can provide location-based alerts for business policy enforcement as well as
location trending, rapid problem resolution, and RF capacity management using its stored location
data. Cisco Wireless Location Appliances are servers that compute, collect, and store historical
location data for up to 1500 devices, including laptops, Voice over IP (VoIP) telephone clients,
radio frequency identification (RFID) asset tags, rogue APs, and rogue clients.

The centralized management of WCS is extended via the capabilities and easy-to-use GUI of the
Cisco Wireless Location Appliance, which makes setup fast and easy. This is after an initial
configuration using the CLI console. After the Location Appliance configuration is complete, the
location server communicates directly with the LAN Controllers.

The Cisco Wireless Location Appliance will collect the assigned operator-defined location data.
The collected information is used for tracking up to 1500 devices for a 30-day period.

Wireless Location Appliance Architecture
The Cisco Wireless Location Appliance architecture is designed so it can interact with WCS as
a client. This allows WCS to centrally control and provide visualization services. The Cisco
Wireless Location Appliance utilizes the same LWAPs as wireless client and Wi-Fi tag location
“readers.” The readers or Cisco LWAPs collect the RSSI information and send it to the Cisco
WLCs. The aggregate RSSI information is then sent to the associated Cisco Wireless Location
Appliance via SNMP. The Cisco Wireless Location Appliance uses the aggregated RSSI
information and performs location computations.

An RF prediction and heat map can be generated once the network maps and APs are added. The
site floor plans can then graphically display all the wireless devices. The WCS visualization
provides immediate asset location application for many administrators.

Wireless Location Appliance Applications
All enterprises could benefit from RF capacity planning and location-based security as well as
maintaining asset visibility. The location information can be made available to third-party app-
lications via SOAP XML APIs on the appliance. A multitude of specialized wireless applications
can be created based on such features as these:

■ Visibility and tracking of mobile devices by using Wi-Fi tags—Anything you can put a
Wi-Fi tag on is manageable, such as computer equipment, office furniture, business phones,
and trade tools. Any asset can be quickly located within the WLAN.

■ Workflow automation and people tracking—This involves automating awareness of
inbound or outbound deliveries or shoppers coming to the register. Police, firefighters,
security personnel, and children can all be tracked at any time.
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■ Telemetry—This involves delivering information in a serialized format containing variable
information, such as car and truck mileage or inventory changes.

■ WLAN security and network control—This involves containing information and
awareness by locating rogue APs, rogue clients, and secure network control.

■ RF capacity management and visibility—Integrating and reviewing location-based trend
reports for RF traffic patterns allows for an improvement to capacity management.

WCS Configuration Examples
The WCS configuration first requires an authorized login. Several configuration steps must take
place after the initial authorized login, such as adding devices and site maps.

WCS Login Steps
The Cisco WCS Server login involves three major steps:

Step 1 Start Microsoft Internet Explorer version 6.0 or later.
Step 2 Enter https://localhost in the address bar when the Cisco WCS user
interface is on a Cisco WCS server. Enter https://wcs-ip-address when the
Cisco WCS interface is on any other workstation.
Step 3 Enter your username and password on the login page. The default username
is root, and the default password is public.

NOTE Some Cisco WCS features might not function properly if you use a web browser other
than Internet Explorer 6.0 or later.

Changing the Root Password
The following are the steps to change the root password:

Step 1 Log in as root.
Step 2 Select Administration > Accounts.
Step 3 From the User Name column, click root.
Step 4 Enter a new password in the New Password text box, and retype the new
password in the Confirm New Password text box.
Step 5 Click Submit.
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Adding a Wireless LAN Controller
The first step when adding a WLC is gathering the IP address of the controller service port. Use
the following steps to add the controller:

Step 1 Log into Cisco WCS.
Step 2 Choose Configure > Controllers from the All Controllers page.
Step 3 Click the Select a Command drop-down menu, choose Add Controller,
and click GO.
Step 4 Enter the controller IP address, network mask, and required SNMP settings
in the Add Controller fields (see Figure 10-4).

Figure 10-4 Adding a WLC

Step 5 Click OK.

NOTE Cisco WCS displays the Please Wait dialog box during the initial contact and while it
is being added to the Cisco WCS database. Control is returned to the Add Controller page again
upon success.

Controller management through the dedicated service port of the controller improves security.
Some controllers do not have dedicated service ports, such as the Cisco 2000 Series WLC, which
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must use the controller management interface. Moreover, if a controller service port is disabled,
the management interface of the controller must be used.

An issue might arise in which the WCS cannot communicate with the controller. A Discovery
Status dialog box appears with a message “No response from device, check SNMP.” A few checks
can verify the correct settings:

■ A bad IP address on the controller service port

■ A blocked network path can be verified by pinging the controller from the WCS server

■ SNMP mismatch between the controller and Cisco WCS

You can continue to add or return additional controllers to the All Controllers page by choosing
Configure > All Controllers.

Configuring Access Points
To view a summary of all Cisco LWAPs in the Cisco WCS database, choose Configure > Access
Points. This page allows you to add third-party APs and remove selected Cisco LWAPs. When a
WLC is added to WCS, it automatically adds all the LWAPs, too.

NOTE There is no need to add Cisco LWAPs to the Cisco WCS database. The operating
system software automatically adds Cisco LWAPs as they associate with existing Cisco WLCs
in the Cisco WCS database.

The All Access Points page displays the AP name, radio type, map location, controller, port,
operational status, and alarm status. Figure 10-5 shows the All Access Points page.
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Figure 10-5 All Access Points Page

WCS Map
Cisco WCS can use real floor, building, and campus plans to view the physical and RF
environments together. This section discusses adding a campus map and a new building.

Adding a Campus Map
Use the following steps to add a campus map:

Step 1 Save the map in a format such as .png, .jpg, .jpeg, or .gif. Do not worry
about the size, because WCS will manage it.
Step 2 Browse to the map and import it from anywhere in the file system.
Step 3 Choose the Monitor tab.
Step 4 Choose Maps.
Step 5 From the Select a Command drop-down menu, choose New Campus and
click Go.
Step 6 On the New Campus page, enter the campus name and contact.
Step 7 Choose Browse, and select the campus graphic name.
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Step 8 Choose Maintain Aspect Ratio so that WCS does not distort the map.
Step 9 Enter the horizontal and vertical span size in feet.

NOTE The campus horizontal and vertical spans should be larger than any building or floor
plan to be added to the campus.

Step 10 Click OK.
Cisco WCS displays the Maps page, which lists maps in the database along with map types and
their status. Figure 10-6 shows a sample Maps page.

Figure 10-6 Maps Page

A WCS map can start out as either a building or a campus map. The building map can be a single
entity or part of the campus map. Moreover, the campus map can have an outdoor coverage area.

Adding a New Building
Buildings can be added without maps. Use the following steps to add a building:

Step 1 Choose the Monitor tab.
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Step 2 Choose Maps.
Step 3 When you choose the desired campus, WCS displays the Campus page (see
Figure 10-7).

Figure 10-7 Campus Page

Step 4 From the Select a Command drop-down menu, choose New Building and
click Go.
Step 5 Create a virtual building to organize related floor plan maps by entering the
building name, contact, number of floors and basements, and the horizontal
and vertical span size in feet.

NOTE Ctrl-Left-Select is an alternative key stroke combination to resize the bounding area
in the upper left corner of the campus map. When changing the bounding size, the building
horizontal span and vertical span parameters automatically adjust to match your changes.

Step 6 Choose Place to put the scaled rectangular building on the campus map.
Figure 10-8 shows the placement of a new building.
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Figure 10-8 Adding a New Building

Step 7 Move the building to the desired location.
Step 8 Choose Save.

NOTE A hyperlink that is associated with the building links the corresponding Maps page.

Rogue Access Point Detection
The process flow of a rogue AP being detected in a WLAN environment is based on the LWAPs
already being powered up and associated to their controllers. The WLC detects a rogue AP and
immediately notifies WCS, which creates a rogue AP alarm that appears in the lower-left corner
of the user interface pages. Simply selecting the indicator displays the Rogue AP Alarms page.

Rogue Access Point Alarms
The alarms for rogue APs are naturally listed on the Rogue Access Point Alarms page. This page
details the severity, the rogue MAC address, the vendor, the radio type, the strongest AP RSSI, the
date and time, the channel number, and the SSID. You can view further details by clicking the link
in the Rogue MAC Address column. Then you see the associated Alarms > Rogue AP MAC
Address page. To view rogue AP information using the menu bar, choose Monitor > Alarms >
Rogue AP Alarms.
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You can handle the alarms by checking a box to the left of the severity and manage them using the
Select a Command drop-down menu. The choices available are Assign to Me, Unassign, Delete,
Clear, or Email Notification.

Rogue Access Point Location
To see the rogue AP calculated location on a map, choose Map from the Rogue AP MAC Address
page, or from the menu bar choose Monitor > Maps > Building Name > Floor Name. A small
skull-and-crossbones indicator appears at the calculated location. The calculated location is to the
nearest AP based on the strongest RSSI with WCS Base. WCS Location compares the RSSI signal
strength from multiple APs to pinpoint the most probable location using RF fingerprinting
technology.
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Foundation Summary

The “Foundation Summary” is a collection of information that provides a convenient review of
many key concepts in this chapter. If you are already comfortable with the topics in this chapter,
this summary can help you recall a few details. If you just read this chapter, this review should
help solidify some key facts. If you are doing your final preparation before the exam, the
information in this section is a convenient way to review the day before the exam.

Following are the five elements of Cisco Unified Wireless Network:

■ Client devices—Use the Cisco Compatible Extensions program helps ensure inter-
operability. The Cisco Compatible Extensions program delivers services such as wireless
mobility, QoS, network management, and enhanced security.

■ Mobility platform—Provides ubiquitous access in any environment indoors or out. The
lightweight access points (LWAP) are dynamically configured and managed by wireless LAN
controllers (WLC) through LightWeight Access Point Protocol (LWAPP).

■ Network unification—Creates seamless integration into the routing and switching
infrastructure. The WLCs are responsible for functions such as RF management, n+1
deployment, and Intrusion Prevention System (IPS).

■ World-class network management—Enables wireless local-area network (WLANs) to
have the equivalent LAN security, scalability, reliability, ease of deployment, and
management via Cisco Wireless Control System (WCS). Cisco WCS provides features for
design, control, and monitoring.

■ Unified advanced services—Support new mobility applications, emerging Wi-Fi tech-
nologies, and advanced threat detection and prevention capabilities such as wireless VoIP,
future unified cellular, location services, Network Admission Control (NAC), the Self-
Defending Network, (Identity Based Network Services(IBNS), Intrusion Detection Systems
(IDS), and guest access.

Cisco offers two WLAN implementations: autonomous and lightweight. Table 10-5 contrasts the
two solutions.

Table 10-5 Comparison of WLAN Implementation Solutions

Category Autonomous WLAN Solution Lightweight WLAN Solution
Access Point Autonomous APs LWAPs
Control Individual configuration on each AP Configuration via Cisco WLC
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Table 10-5 Comparison of WLAN Implementation Solutions (Continued)

Category Autonomous WLAN Solution Lightweight WLAN Solution
Dependency Independent operation Dependent on Cisco WLC
WLAN Management Management via CiscoWorks Management via Cisco WCS
WLSE and Wireless Domain
Services (WDS)
Redundancy AP redundancy Cisco WLC redundancy

CiscoWorks WLSE is a management tool for WLANs with autonomous APs. It is designed to
centralize management, reduce total cost of ownership, minimize security vulnerabilities, and
improve WLAN uptime. Features and benefits of Cisco WLSE are summarized in Table 10-6.

Table 10-6 CiscoWorks WLSE Features and Benefits

Feature Benefit
Centralized configuration, firmware, and radio Reduces WLAN total cost of ownership by saving
management time and resources required to manage large
numbers of APs
Autoconfiguration of new APs Simplifies large-scale deployments
Security policy misconfiguration alerts and Minimizes security vulnerabilities
rogue AP detection
AP utilization and client association reports Helps in capacity planning and troubleshooting
Proactive monitoring of APs, bridges, and Improves WLAN uptime
802.1x EAP1 servers

1 EAP = extensible authentication protocol

CiscoWorks WLSE supports Secure Shell (SSH), HTTP, Cisco Discovery Protocol (CDP), and
Simple Network Management Protocol (SNMP). CiscoWorks WLSE comes in two versions:
CiscoWorks WLSE and WLSE Express. CiscoWorks WLSE supports up to 2500 WLAN devices.
WLSE Express supports up to 100 WLAN devices. The WLSE Express setup option is either
Automatic or Manual.

Cisco WCS is a Cisco WLAN solution network-management tool that is designed to support 50
Cisco WLCs and 1500 APs. Cisco WCS supports SNMPv1, SNMPv2, and SNMPv3.

Cisco WCS comes in three versions:

■ Cisco WCS Base—The base version of Cisco WCS can determine which AP a wireless
device is associated with.
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316 Chapter 10: WLAN Management

■ Cisco WCS Location—Cisco WCS Location is the base plus Cisco RF fingerprinting
technology.

■ Cisco WCS Location + 2700 Series Wireless Location Appliance—Cisco Wireless
Location + 2700 Appliance tracks thousands of devices in real time, enabling key business
applications such as asset tracking, inventory management, and e911.

The Cisco Wireless Location Appliance provides simultaneous device tracking and data collection
for capacity management or location trending:

■ The Cisco Wireless Location Appliance is an innovative, easy-to-deploy solution that uses
advanced RF fingerprinting technology to simultaneously track thousands of 802.11 wireless
devices from directly within a WLAN infrastructure.

■ Cisco 2700 Series Wireless Location Appliances are servers that enhance the high-accuracy
built-in Cisco WCS location abilities by computing, collecting, and storing historical location
data for up to 1500 laptop clients, palmtop clients, Voice over IP (VoIP) telephone clients,
radio frequency identification (RFID) asset tags, rogue APs, and rogue AP clients.

To access the Cisco WCS Network Summary page, choose Monitor > Network Summary.

To access the Cisco WCS Controller Summary details, choose Monitor > Devices > Controllers.

The default Cisco WCS username is root, and the default password is public.

The WLC detects rogue APs and immediately notifies the WCS, which in turn creates a rogue AP
alarm in the lower-left corner of the user interface pages. To view rogue AP information using the
menu bar, choose Monitor > Alarms > Rogue AP Alarms. The alarms choices available are
Assign to Me, Unassign, Delete, Clear, or Email Notification. To see the location of a rogue
AP, either choose Map from the Rogue AP MAC Address page, or choose Monitor > Maps >
Building Name > Floor Name from the menu bar. A small skull-and-crossbones indicator
appears at the calculated location.
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Q&A 317

Q&A

Some of the questions that follow challenge you more than the exam by using an open-ended
question format. By reviewing now with this more difficult question format, you can exercise your
memory better and prove your conceptual and factual knowledge of this chapter. The answers to
these questions appear in Appendix A.

1. Discuss the different characteristics and advantages of the two WLAN solutions.
2. Does WLSE support both lightweight and autonomous access points?
3. When do you use CiscoWorks WLSE versus WLSE Express?
4. Discuss the platform support for Cisco WCS.
5. What are the three WCS versions for tracking wireless devices?
6. When does Cisco WCS listen for rogue access points?
7. How do you add lightweight access points to the WCS database?
8. Does Cisco WCS support SNMPv3?
9. What page is displayed upon a successful WCS login?
10. What is the Cisco WCS default username and password?
11. For the WLAN components, what are the WLAN Management solutions?
12. What happens to Rogue APs when they are detected?
13. What are the differences between WCS base, Location, and Location + Appliance (including
how many clients can be tracked)?
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Part IV: Appendix

Appendix A Answers to the “Do I Know This Already?” Quizzes and Q&A
Sections
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APPENDIX A
Answers to the “Do I Know
This Already?” Quizzes
and Q&A Sections

Chapter 1

“Do I Know This Already?” Quiz
1. D

2. C
3. B
4. A
5. D
6. D
7. B
8. A
9. B
10. C
11. A
12. D
13. C
14. D
15. A
16. C
17. B
18. D
19. A
20. B
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322 Appendix A: Answers to the “Do I Know This Already?” Quizzes and Q&A Sections

Q&A
1. The benefits of packet telephony networks include these:
■ More efficient use of bandwidth and equipment

■ Lower transmission costs

■ Consolidated network expenses

■ Improved employee productivity

■ Access to new communication devices

2. Following are the components of a packet telephony (VoIP) network:
■ Phones

■ Gateways

■ Multipoint control units

■ Application servers

■ Gatekeepers

■ Call agents

■ Video end points

3. The analog interfaces through which legacy analog devices can connect to a VoIP network
include these:
■ FXS

■ FXO

■ E&M

4. The digital interface options to connect VoIP equipment to PBXs or the PSTN include the
following:
■ BRI

■ T1/E1 CAS

■ T1/E1 CCS

5. The three stages of a phone call are:
1. Call setup
2. Call maintenance
3. Call teardown
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Chapter 1 323

6. The two main models of call control are distributed call control and centralized call control.
Examples of distributed call control include H.323 and SIP. An example of centralized call
control is MGCP.
7. The steps for converting analog signals to digital signals include the following:
1. Sampling
2. Quantization
3. Encoding
4. Compression (optional)

8. Following are the steps for converting digital signals to analog signals:
1. Decompression of the samples, if compressed
2. Decoding
3. Reconstruction of the analog signal from PAM signals

9. The sampling rate must be at least twice the maximum signal frequency. Because the
maximum voice frequency over a telephone channel was considered 4000 Hz, based on the
Nyquist theorem, a rate of 8000 samples per second is required.
10. The two main quantization techniques are linear quantization and logarithmic quantization.
11. µ-Law is used in the United States, Canada, and Japan. A-Law is used in other countries. Both
methods are quasi-logarithmic; they use logarithmic segment sizes and linear step sizes
within each segment. For communication between a µ-Law and an A-Law country, the µ-Law
country must change its signaling to accommodate the A-Law country.
12. The main codec/compression standards and their bandwidth requirements are as follows:
■ G.711 (PCM)—64 Kbps

■ G.726 (ADPCM)—16, 24, or 32 Kbps

■ G.728 (LDCELP)—16 Kbps

■ G.729 (CS-ACELP)—8 Kbps

13. MOS stands for mean opinion score. It is a measurement of voice quality derived from the
judgment of several subscribers. The range of MOS scores is 1 to 5, where 5 is the perfect
score for direct conversation.
14. DSP stands for digital signal processor. It is a specialized processor used for the following
telephony applications: voice termination, conferencing, and transcoding.
15. The TCP/IP protocols that are responsible for transporting voice are RTP (12 bytes), UDP (8
bytes), and IP (20 bytes).
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324 Appendix A: Answers to the “Do I Know This Already?” Quizzes and Q&A Sections

16. RTP provides sequence numbering (reordering) and time-stamping to complement UDP.
17. cRTP stands for Compressed RTP. cRTP reduces the IP, UDP, and RTP headers from 40 to 2
bytes (without a checksum), and to 4 bytes (with a checksum). cRTP provides significant
bandwidth savings, but it is only recommended for use on slow links (less than 2 Mbps).
18. Packet rate, packetization size, IP overhead, data link overhead, and tunneling overhead
influence the bandwidth requirements of VoIP.
19. The packet rate and packetization period are reciprocal. For example, if the packetization
period is 20 milliseconds (0.020 seconds), the packet rate is equal to 1 over 0.020, or 50
packets per second.
20. The sizes of Ethernet, 902.1Q, Frame Relay, and Multilink PPP (MLP) overheads are as
follows:
■ Ethernet—18 bytes

■ 802.1Q+Ethernet—4 + 18 = 22 bytes

■ Frame Relay—6 bytes

■ MLP—6 bytes

21. Following are the tunneling and security protocols and their associated overheads:
■ IPsec transport mode—30 to 53 bytes

■ IPsec Tunnel mode—50 to 73 bytes

■ L2TP—24 bytes

■ GRE—24 bytes

■ MPLS—4 bytes

■ PPPoE—8 bytes

22. Following are the steps necessary to compute the total bandwidth for a VoIP call:
1. Determine the codec type and packetization period.
2. Gather the link information. Determine whether cRTP, any type of tunneling, or IPsec is
used.
3. Calculate the packetization size or period.
4. Add all the headers to the packetization size.
5. Calculate the packet rate.
6. Calculate the total bandwidth.
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Chapter 1 325

23. VAD stands for voice activity detection. It suppresses the transmission of silence; therefore,
it might result in up to 35 percent bandwidth savings. The success of VAD depends on the
types of audio, the level of background noise, and other factors.
24. The components of the enterprise voice implementations include the following:
■ Gateways

■ Gatekeepers

■ IP phones

■ Cisco Unified CallManager

25. On a Cisco router, voice gateway functions include the following:
■ Connect traditional telephony devices

■ Convert analog signals to digital and vice versa

■ Encapsulate digital voice into IP packets

■ Perform voice compression

■ Provide DSP resources for conferencing and transcoding