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					                                                                                Introduction


Introduction to Realtimepublishers
by Don Jones, Series Editor

For several years, now, Realtime has produced dozens and dozens of high-quality books that just
happen to be delivered in electronic format—at no cost to you, the reader. We’ve made this
unique publishing model work through the generous support and cooperation of our sponsors,
who agree to bear each book’s production expenses for the benefit of our readers.
Although we’ve always offered our publications to you for free, don’t think for a moment that
quality is anything less than our top priority. My job is to make sure that our books are as good
as—and in most cases better than—any printed book that would cost you $40 or more. Our
electronic publishing model offers several advantages over printed books: You receive chapters
literally as fast as our authors produce them (hence the “realtime” aspect of our model), and we
can update chapters to reflect the latest changes in technology.
I want to point out that our books are by no means paid advertisements or white papers. We’re an
independent publishing company, and an important aspect of my job is to make sure that our
authors are free to voice their expertise and opinions without reservation or restriction. We
maintain complete editorial control of our publications, and I’m proud that we’ve produced so
many quality books over the past years.
I want to extend an invitation to visit us at http://nexus.realtimepublishers.com, especially if
you’ve received this publication from a friend or colleague. We have a wide variety of additional
books on a range of topics, and you’re sure to find something that’s of interest to you—and it
won’t cost you a thing. We hope you’ll continue to come to Realtime for your educational needs
far into the future.
Until then, enjoy.
Don Jones




                                                 i
                                                                                                               Table of Contents


Introduction to Realtimepublishers.................................................................................................. i
Chapter 1: Forward…Into the Past!.................................................................................................1
VoIP vs. IPT.....................................................................................................................................3
The Uni-Media Phase ......................................................................................................................3
The Multi-Media Phase....................................................................................................................4
           VoIP .....................................................................................................................................5
Technical Definition vs. Market Definition.....................................................................................5
VoIP is Revolutionary......................................................................................................................6
VoIP Business Aspects Not Clear....................................................................................................7
           IPT........................................................................................................................................9
Major Changes ...............................................................................................................................12
           Telephony Becomes an Application ..................................................................................13
           Responsibility Shifts from Carrier to End-User Organizations .........................................14
The IPT Life Cycle ........................................................................................................................15
           Planning and Assessment...................................................................................................16
           Pre-Deployment Testing and Implementation ...................................................................17
           Ongoing Operations and Optimization ..............................................................................17
           Optimization ......................................................................................................................18
Summary ........................................................................................................................................19
Chapter 2: The IP Telephony Life Cycle.......................................................................................20
The Bigger Business Picture..........................................................................................................20
           Packet Telephony Is Inevitable..........................................................................................21
           Impact of VoP ....................................................................................................................27
           VoIP and IPT Return on Effort..........................................................................................28
Enterprise Telephony Migration ....................................................................................................28
           The Budget Process............................................................................................................28
                      Reuse of Existing Assets........................................................................................29
                      New Assets Needed ...............................................................................................30
           Assessing Urgency and Prioritizing Migration..................................................................31
           Migration............................................................................................................................31
                      Avoiding Migration ...............................................................................................31
                      Partial Migration ....................................................................................................32
                      Full Migration ........................................................................................................32


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           Delaying VoIP/IPT ............................................................................................................32
           Systems Integrator vs. Do It Yourself................................................................................32
           Managed Service vs. Do It Yourself..................................................................................33
           SLAs ..................................................................................................................................34
                      Penalties .................................................................................................................34
                      True Impact of SLA on Business and Operations .................................................35
The IPT Life Cycle ........................................................................................................................35
           Planning and Assessment...................................................................................................36
           Pre-Deployment Testing and Implementation ...................................................................37
           Ongoing Operations and Optimization ..............................................................................37
Managing the IPT Life Cycle ........................................................................................................38
           Enterprise-Accessible Tools ..............................................................................................38
                      MSP-Provided Customer Network Management ..................................................38
                      Manufacturer-Provided Systems............................................................................38
                      Third-Party Tools...................................................................................................39
                      Importance of Modeling, Monitoring, Measuring, and Managing ........................39
Summary ........................................................................................................................................40
Chapter 3: Planning and Assessment.............................................................................................41
Business Drivers for Change .........................................................................................................42
           Lower Costs .......................................................................................................................44
                      The Myth of Bandwidth Savings ...........................................................................45
                      Access Charge/Local .............................................................................................46
                      Long Distance ........................................................................................................46
                      Support Costs .........................................................................................................47
                      Remote/Nomadic/Mobile Workers........................................................................48
           Enabling New Strategic Applications................................................................................48
           Worker Productivity...........................................................................................................49
           Obsolescence of Older System ..........................................................................................49
                      No Longer Want to Invest in Old Technology ......................................................50
                      Manufacturer Discontinuation of Product and/or Support.....................................50
                      Increased Support and Maintenance Costs ............................................................50
                      Lack of Expansion Capability of Old System........................................................51
Setting Proper Expectations...........................................................................................................51


                                                                         iii
                                                                                                          Table of Contents


          Management Expectations .................................................................................................51
          End-User Expectations.......................................................................................................52
Inventory of Existing Capabilities .................................................................................................53
          Existing Telephony Performance Expectations Audit.......................................................54
          Existing Telephony Budgeting, Relationships and Tools Audit........................................55
          Existing Capabilities Audit ................................................................................................55
          Synchronizing Existing and New Capabilities ..................................................................58
Network Readiness Assessment ....................................................................................................59
          The Network Readiness Assessment Process ....................................................................61
                     In a Perfect World..................................................................................................62
                     In the Real World...................................................................................................62
          Understanding the Key Metrics of a Network Assessment ...............................................62
                     High Availability ...................................................................................................64
                     Call and Voice Quality...........................................................................................65
          Tips and Tricks for A Successful Network Assessment....................................................71
                     Simulating Calling Patterns ...................................................................................72
                     Simulating Gateways .............................................................................................72
                     Plan For The Future ...............................................................................................74
Assessment Tools...........................................................................................................................74
          Third-Party Tools...............................................................................................................74
                     Third-Party Tools Selection “Report Card”...........................................................75
Summary: The Road to Success ....................................................................................................76
Chapter 4: Design and Pre-Deployment Testing ...........................................................................78
Network Design Considerations ....................................................................................................79
          The Team ...........................................................................................................................79
          Assumptions.......................................................................................................................83
          The Voice/Data/Video “Triple Play” and Unified Communications ................................84
                     Real-Time ..............................................................................................................85
                     Near Real-Time......................................................................................................86
                     Background/Batch..................................................................................................86
          Interrelationship of Multi-Media Services.........................................................................86
                     Application Availability.........................................................................................86
                     Delivery/Accuracy/Loss ........................................................................................90


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                                                                                                             Table of Contents


                      Delay ......................................................................................................................92
                      Delay Variation or Jitter.........................................................................................92
                      Differentiation, Prioritization and Queue Management.........................................93
           Voice QoS..........................................................................................................................95
                      QoE ........................................................................................................................96
                      QoE vs. QoS...........................................................................................................96
                      Voice Metrics and Measurements........................................................................101
           Impact on Business Processes..........................................................................................103
                      Standardizing Positive Impacts............................................................................104
                      Avoiding Negative Impacts .................................................................................104
Translating Needs to SLAs ..........................................................................................................104
           The “Proper” SLA............................................................................................................105
           Metrics .............................................................................................................................105
           Measurement....................................................................................................................106
           Establishing Feedback Loops and Review Cycles...........................................................106
Validation of Design ....................................................................................................................107
           Network Tuning ...............................................................................................................107
           Test Bed Architecture ......................................................................................................108
                      Network Impairments ..........................................................................................108
                      Timing/Synchronization Issues............................................................................109
                      Broadband Bandwidth Measurement and Calculation ........................................110
                      Gateway, Softswitch, Session Border Controller and IP Device Capacity and
                      Performance .........................................................................................................111
Testing, Feedback, Network Modification, Testing Loop...........................................................112
                      Testing/Proof of Concept Objectives...................................................................113
                      Feature Support and End-to-End Operation ........................................................113
                      Closed Lab Environment .....................................................................................115
                      After Hours/Off Hours Familiarization and Benchmarking ................................115
                      Busy Hour Testing in a Live Network.................................................................115
                      Evaluation of Results and Ensuring End-User Acceptance.................................115
                      Fall-Back and Contingency Planning ..................................................................118
           Use of Third-Party Tools .................................................................................................118
Summary ......................................................................................................................................119
Chapter 5: Implementation and Migration...................................................................................120

                                                                         v
                                                                                                              Table of Contents


External Issues .............................................................................................................................120
           Reducing Negative Impact of Implementation/Migration...............................................120
           Business Cycles ...............................................................................................................121
           Holidays ...........................................................................................................................122
           Coordination ....................................................................................................................122
Plans.............................................................................................................................................123
           Implementation Plan ........................................................................................................125
           Training Plan....................................................................................................................125
           Test/Acceptance Plan.......................................................................................................126
           Migration Plan .................................................................................................................126
           Ongoing Operations Plan.................................................................................................127
           Contingency and Business/System Continuity Plan ........................................................127
                      Contingency Plan .................................................................................................128
                      Business/System Continuity Plan ........................................................................128
           Escalation Procedure........................................................................................................129
Internal Issues ..............................................................................................................................129
           Creation of Baseline and User Profiles............................................................................129
           Bringing All Users to Baseline ........................................................................................130
           Additional Capabilities by User Profile...........................................................................130
           Security Issues .................................................................................................................130
                      Liaison with Security Department .......................................................................131
                      Firewalls and Proxy Server Issues .......................................................................131
                      IP Addresses and Port Numbers ..........................................................................132
                      NAT and Internal and External IP Addresses......................................................133
           Internal and External Phone Number Mapping ...............................................................133
           Flash Cut vs. Parallel Operation ......................................................................................134
           Migration Priority and Order ...........................................................................................134
           Involvement of End Users in Acceptance........................................................................134
IP Telephony Implementation and Migration Case Study...........................................................135
           Business ...........................................................................................................................137
                      ROI and TCO.......................................................................................................137
                      Establish Site Profiles ..........................................................................................138
                      Rough-Order-of-Magnitude Pricing ....................................................................139


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                      Management Review and Site Prioritization .......................................................139
                      Initial Migration/Green Field Plan and Renegotiation of Favorable Contracts...139
                      Joint Migration/Green Field Planning Effort.......................................................139
                      Contracts and Delivery ........................................................................................140
                      Inventory and Fixed Asset Accounting and Disposition of Old Equipment .......140
                      Implementation of Changes to Business Operations ...........................................140
           Technical..........................................................................................................................142
                      Review of Standards’ Status and Maturity ..........................................................142
                      Develop Private Net, IP VPN, and Carrier/PTT Strategies .................................142
                      Private Backbone Design/Upsize for Voice Traffic and Backbone Re-engineering143
                      Establish Common Standards and Test Plan .......................................................143
                      Establish Private Net, IP VPN, and Carrier/PTT Demo/Test Capabilities ..........143
                      Training/Staging for Migration Teams ................................................................143
                      Training for Support Teams.................................................................................144
                      Review of Planning and New Telephony Project Rollouts .................................144
           Legal ................................................................................................................................144
                      Closing Contracts for Old Telephony System .....................................................144
                      New Contracts......................................................................................................144
           Regulatory........................................................................................................................145
                      Compliance with National Law and Telco/PTT Regulation................................145
           Contact Centers/Call Centers...........................................................................................146
                      Special Considerations and Contingency Planning .............................................146
                      Phased Implementation and Call-Taker Training ................................................146
Summary ......................................................................................................................................146
Chapter 6: Ongoing Operations ...................................................................................................147
Network Operation.......................................................................................................................147
           Capacity, Performance, and the Broadband Flip .............................................................148
                      Reliability.............................................................................................................156
                      Security ................................................................................................................156
                      Call Quality and Voice Quality............................................................................160
                      Measurement and Monitoring..............................................................................163
                      Growth Management ...........................................................................................164
           Recordkeeping and Documentation.................................................................................166


                                                                         vii
                                                                                                            Table of Contents


                      Fixed Asset Accounting.......................................................................................166
                      Shipping, Warehousing, and Tracking.................................................................167
                      Repair/Refurbishment/Return to Service/End-of-Life.........................................167
Administration .............................................................................................................................168
           End-User Cost Allocation ................................................................................................168
           Detailed Usage and Billing ..............................................................................................168
           Vendor/MSP/Carrier Billing............................................................................................169
           Cost Reduction and Network Utilization Maximization .................................................169
Provisioning .................................................................................................................................169
Maintenance.................................................................................................................................170
           Moves, Adds, and Changes..............................................................................................170
           Technology Refresh .........................................................................................................170
           Audits...............................................................................................................................170
Operations Tools..........................................................................................................................171
           Integration ........................................................................................................................171
           Operation..........................................................................................................................171
Summary ......................................................................................................................................171
Chapter 7: Optimization...............................................................................................................172
SLAs and the Scope of Optimization...........................................................................................172
The Optimization Cycle...............................................................................................................173
The Use of Software Tools ..........................................................................................................174
           Tool Ubiquity...................................................................................................................174
           Constant Testing and Monitoring ....................................................................................175
Exception Reporting and Filtering...............................................................................................175
           “What If” Scenarios .........................................................................................................175
Trending and Capacity Planning..................................................................................................176
           Bandwidth ........................................................................................................................176
           Ports/Lines .......................................................................................................................177
           Phone Numbers/Numbering Plan ....................................................................................179
           Grooming .........................................................................................................................180
                      Gateways..............................................................................................................181
                      TDM Grooming ...................................................................................................181
           VLAN Capacity ...............................................................................................................182


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                                                                                                             Table of Contents


           VPN Capacity ..................................................................................................................182
           Infrastructure....................................................................................................................182
Hardware......................................................................................................................................182
           Repair...............................................................................................................................183
           Obsolescence/Refresh ......................................................................................................184
Software .......................................................................................................................................184
           Maintenance.....................................................................................................................184
           Enhancements ..................................................................................................................185
           Upgrades ..........................................................................................................................185
           Patches and Security Audits.............................................................................................185
SLA Improvements......................................................................................................................186
           Delay ................................................................................................................................186
                      Call Setup.............................................................................................................187
                      During Call...........................................................................................................187
                      Call Tear Down/Release ......................................................................................187
           Delay Variation................................................................................................................187
           Sample and Packet Loss...................................................................................................188
           Availability ......................................................................................................................188
Telephony Service Availability ...................................................................................................189
           GoS ..................................................................................................................................189
                      Blocking/Non Blocking Access...........................................................................190
                      Success Rates of Call Setup.................................................................................191
                      Related Gateway Issues .......................................................................................191
Prices and Costs ...........................................................................................................................191
           Hardware Costs................................................................................................................191
           Service and Support Costs ...............................................................................................192
           In-Sourcing vs. Outsourcing ............................................................................................193
IP Contact Center Optimization...................................................................................................193
Summary ......................................................................................................................................194
Chapter 8: Sweating the Details and Assuring Success...............................................................195
Legal and Regulatory Issues ........................................................................................................195
           General Considerations....................................................................................................196
                      Countrywide Bans of VoIP Services or Service Providers..................................197


                                                                         ix
                                                                                                             Table of Contents


                      Penalties for Non-Compliance.............................................................................197
                      Emergency Services.............................................................................................198
                      Security and Wiretapping/Call Recording ...........................................................198
                      Records Retention................................................................................................199
                      Taxes and Fees.....................................................................................................199
           Country-Specific Regulatory Issues ................................................................................199
           Contractual Issues ............................................................................................................199
           Patent Issues.....................................................................................................................200
Technical Issues ...........................................................................................................................201
           Voice Band Data ..............................................................................................................201
           Numbering Plans..............................................................................................................202
                      Voice VPNs .........................................................................................................202
                      E.164 Compliance................................................................................................202
                      URIs/URLs ..........................................................................................................203
                      ENUM and Other Numbering Issues...................................................................204
           Interworking Issues with IPT Service Providers .............................................................204
                      The New Demark.................................................................................................204
                      SIP-T ....................................................................................................................208
                      Electrical Power ...................................................................................................209
Business Issues.............................................................................................................................209
           Budgets ............................................................................................................................210
                      Staff......................................................................................................................210
           Repositioning/Redeployment of Old Equipment.............................................................210
           Secondary Market for Old Equipment.............................................................................210
           Donation of Old Equipment.............................................................................................211
                      Revisiting TCO and ROI Models for your organization .....................................211
Security, Privacy, and Safety .......................................................................................................212
           Emergency Calls—911, 999, 112, and So On .................................................................213
Telephony Migration Checklist ...................................................................................................213
Summary ......................................................................................................................................213
Telephony Migration Checklist ...................................................................................................214




                                                                         x
                                                                        Copyright Statement


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                                                 xi
                                                                                       Chapter 1


Chapter 1: Forward…Into the Past!
Telephony is changing. Packet-based voice is inevitable. Being in traditional telephony today is
akin to being a telegraph engineer in the late 1880s—wondering what that odd couple Bell and
Watson were doing with your telegraph cables late at night. Only in our case, there are no thick
metallic cables strung from wooden poles. Today, we have wireless signals completing the trip
that started on thin strands of glass and may end on MOCA-flavored coaxial cables or twisted
pairs of copper wire in residences or smaller fibers in the walls of offices, dorms, manufacturing
plants, and laboratories.
You are reading this guide because you already know that the traditional telephony era is
drawing to a close and you are looking for guidance on how to migrate to the next era as quickly
and painlessly as possible—maybe personally as a career move, or professionally on behalf of an
enterprise or other organization. Chapters 2 through 8 of this guide are all about how to do just
that. But first we must set the stage.
Speaking of setting the stage: several years ago, I saw a Broadway musical called Pippin, about
the son of Charlemagne. Recalling a Broadway musical about the life of the son of Charlemagne
must seem an odd way of starting a book on the successful deployment of Voice over IP (VoIP)
and IP Telephony (IPT), but stick with me, there are some interesting parallels. For one, the story
of Pippin is one that is fairly well known, albeit with its share of misconceptions and historical
fuzziness, similar to the 100+ year history of telephony. The musical tells the story from a
unique, yet enlightened, and enlightening, point of view: it chronicles Pippin’s personal search
for excellence and his desire to make an everlasting contribution. Lastly, at the very beginning of
the musical, the Lead Player requests, “I beg you; cast all previous misconceptions aside and
accept what we enact for you today.” In this same spirit of embracing a new way of looking at
things you think you already know, cast aside your previous misconceptions as you read this
guide. This book represents a fresh, and I hope enlightened and enlightening, view of the next
evolutionary step in human communications.
This guide explores beyond simply replacing dial tone on circuits with dial tone in packets,
though it addresses this basic function because basic dial tone is often all that an organization
needs. This book challenges a lot of what the market has been feeding us and a lot of what we
think we know and “remember.” The view is a simplistic one, really, which asks questions such
as “Why are we doing this?” and “What is our Return on Investment?” This book represents a
view based upon experience, based upon listening at least as much as talking, and based upon
thinking—probably way too much thinking—about why things are being done the way they are
and less about starry-eyed amazement at what technology can do and how it will transform
society and forever change the way humans interact with each other.
This book is written neither for “Bell-head” nor “packet-head” but rather for the next generation
of communications professional who will recognize these former terms only in their historical
context, “back when” there was a difference. Today’s communications professional is more
interested in session initiation, and has a protocol for such things; they are interested in voice as
one of a rich palette of communications methods to solve a specific set of problems in the bigger
picture of instantaneous access to information generically called multi-media, or often called the
voice, data, and video triple play. Oh, and this is the first and last time I will mention H.323
because after this chapter we will only be looking ahead, not backwards.



                                                  1
                                                                                       Chapter 1


To further set the scene, please allow me to enumerate the observations that form the foundation
for this book. The basic underlying premises:
   •   In the early phases of VoIP, there has been a tendency to accept mediocrity for cost
       savings and to embrace lower quality for convenience, but this will soon give way to
       quality as a differentiator of products and services.
   •   “Free” is a great concept but is not sustainable, and VoIP is not often as cost-effective, or
       as nearly “free,” as advertised.
   •   VoIP is a stop along the journey to packet-based telephony, not the destination.
   •   IPT is the real objective for business and organizational users and will revitalize, rather
       than kill, all the best aspects of traditional telephony.
   •   Telephony is becoming an application like spread sheets, word processing, browsers, and
       Instant Messaging, and, like the other applications, has common elements as well as
       special requirements needed to make it truly successful.
   •   End-user organizations are shifting to some degree to playing the role of
       telecommunications carrier for their end users or, at the very least, providing a stronger,
       less transparent, liaison role between their user community and managed service
       providers.
   •   Internal customers must be handled with the best practices that world-class carriers and
       service providers over the years have applied to their external customers, including
       Service Level Agreements (SLAs) and the underlying measurement and reporting of
       performance and service troubleshooting.
   •   Traditional telephony knowledge and skills are as valuable now as they have ever been
       and are increasing in value as the number of people who possess this special knowledge
       is decreasing in the job market.
   •   You might as well plan your trip to packet-based telephony and make the migration as
       smooth and painless as possible because, voluntary or not, you are going there anyway.
   •   There is no need for most organizations to rush to packet telephony. Older telephony
       systems should be kept around as long as they make business sense and as long as they
       can be coaxed into providing dial tone.
This chapter begins with a high-level overview of electronic human voice communications;
subsequent chapters delve into the details of what is needed to make the future better than the
past, how—specifically—to implement business voice communications, and how to measure
performance and use it as a part of a feedback loop for service optimization. The emphasis will
be on midsize to large, national to global enterprises, but smaller users, carriers, and service
providers will find their share of readily accessible knowledge in these words, as well. After
reading this chapter, you will understand the basis for the rest of the book and will be better able
to apply what you read in the remaining chapters.




                                                  2
                                                                                      Chapter 1



VoIP vs. IPT
The dramatic shifts and uncertainty in today’s telephony marketplace can be attributed to two
distinct phases of telephony technology deployment, each occurring at key points in time, with a
very important transitional period occurring between the two. The two phases are the “circuit
phase” and the “packet phase,” with an intervening transitional period I will simply call VoIP.
To make the point clear, you can also call the circuit phase the uni-media phase and the packet
phase the multi-media phase.
This is not to say that no one did multi-media with circuits, quite to the contrary; rather, the
distinction between the two phases focuses on the capabilities inherent in a single connection.
The uni-media phase, a term invented by this author solely as a vehicle for making this
comparison, is characterized by a single circuit delivering a single type of information: a circuit
or channel for voice; another, separate, circuit or channel for data; and another, separate, circuit
or channel for video. The treatment of each circuit as a single information pathway, largely
independent of each other in transmission and possibly synchronized at the end points is
fundamentally different than the multi-media view of treating all connection types as variations
on the packet theme with appropriate allowances being made along the way for the needed
Quality of Service (QoS) for each. In the uni-media phase the separation between different types
of media is more rigid and enforced at a much lower layer in the network.


The Uni-Media Phase
The uni-media phase of telephony is defined as beginning March 10, 1876—the invention of the
telephone—up until mid-1994 when engineer/hobbyists who would go on to form VoIP pioneer
VocalTec made the first PC-to-PC voice call from Princeton, NJ to Israel and VoIP emerged.
The only major change in telephony up until that time was the shift from the analog network to
the digital network. Though VoIP represents a major change, the fundamental achievement was
still a point-to-point call from one human to another.
The uni-media phase of telephony is characterized by fast, reliable, guaranteed QoS circuit
switching, which was originally analog and eventually digital. The first phase delivers a 99.99+
percent reliable range of services to most areas of the world through a vast, carefully constructed
global network based on standards that segregate the circuits into their own, dedicated, protected
uni-media lanes in the network highway. The uni-media phase is also characterized by a rigid,
standardized, hierarchically structured geographically aligned numbering system designed in five
layers that allows any-to-any connectivity globally with a minimum number of steps.




                                                  3
                                                                                       Chapter 1



The Multi-Media Phase
The multi-media phase is disruptive and challenges the established order but goes beyond VoIP.
VoIP was thought, at one point, to be the multi-media phase, but VoIP has turned out to be more
of a “living laboratory” and has become a transitional step from uni-media to multi-media and
not a full-blown revolution unto itself. More than anything else, the advent of VoIP kicked off
the new thinking about the future viability of circuit-switched telephony in light of customer
demands for integrated multi-media applications, such as presence and unified messaging, and
put us on our current trajectory toward all-packet/non-circuit multi-media networks.
The multi-media phase cannot rightfully be called the multi-media phase “of Telephony”
because at least two other media are involved, each having a variety of flavors. Beyond voice,
there is data and video, and each has real-time, near-real-time, and non-real-time variants. The
multi-media phase is represented by the ad hoc use of the global Internet and other IP intranet
and extranet networks for a mixture of telephony, data, and video in a service package often
referred to as the “triple play.” This direction is as true of residential or personal communications
as it is of business communications, the difference being that in the residential context, the triple
play has a decidedly entertainment spin, and in the business realm, the emphasis is on strategic
and tactical applications that will either save or make money.
At the present time, early in the multi-media phase, Internet telephony services are immature,
network reliability is lower than in the uni-media phase, and QoS—and the resulting Quality of
Experience (QoE), are measured differently than in the uni-media phase, yet there are some
compelling reasons to move to a packet-based architecture. Among those reasons are true on-
demand multi-media applications; transparent service ubiquity over metallic, wireless, and fiber
delivery systems; and lower cost/higher efficiency consolidated (I dare not say converged)
backbones, all of which show a lot more business benefit and potential longevity than the early
VoIP cost-focused discounting models that have already failed.
As we move further into the multi-media phase, next-generation carriers will start to compete on
services, not price, over both PSTN and IP network infrastructures. This is the hallmark of the
next phase—a value-added services framework that delivers enhanced telecommunications
capabilities to the marketplace as services, not as boxes and wires that must be further integrated
by customers to provide services. And with a scalable and flexible enhanced service architecture
solution riding atop these new-generation networks, it will actually be possible for the IP-centric
network implementations to increase their competitive advantage over circuit-switched network
providers and beat the analyst projections for market share and revenue growth.




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VoIP
Traditional telephony had changed little in concept or common practice from the very first
telephone call in 1876 until the first VoIP call in 1994. The major breakthroughs had been the
introduction of digital switches, implementation of digital telephony, use of fiber optics for
communications, and the use of out-of-band signaling methods, such as SS-7/CCS-7 and the
ISDN D channel. These are all evolutionary improvements on the basic theme of an end-to-end
circuit switched call. Other important recent changes have occurred in the regulatory
environment. These implementations were made possible by advances such as fiber optic
networking and SS-7 signaling.
Were it not for the advent of the Internet and IP-based telephony efforts, traditional telephony
would probably have stayed the course for many more decades to come. However, traditional
telephony and IP networking did cross paths, which, concurrent with deregulation in the United
States and elsewhere, caused a number of tangible changes.
The largest change has been the precipitous decline in the cost of circuit-switched long distance
service, in not only the United States but also in many other countries around the globe since
1995 to present. Where 20-to-25-cent/minute prices for long distance calls were common in the
U.S. in 1995, now sub-2-cent/minute prices are the norm with companies such as major disruptor
Vonage offering residential flat-rate plans to the U.S., Canada, and Puerto Rico for US$24.99 per
month and similar business plans for $49.99.


Technical Definition vs. Market Definition
A careful analysis of the foregoing discussion speaks to the issue of the two distinct definitions
for VoIP. One definition, a market definition, includes a general, collective, label of VoIP for all
the market, legal, and regulatory forces and technologies that have caused the revolutionary and
dramatic changes in global phone calling, freeing the consumer or business user from having to
understand the details. The second definition, the technical definition, includes a vast number of
technological innovations that have enabled the business definition but of which the thing called
VoIP is only small part. There is not, for instance, a VoIP protocol, per se, but rather a number of
choices, all of which will be examined, contrasted, and compared in subsequent chapters.




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VoIP is Revolutionary
VoIP began in what can best be described as the hobbyist realm. Early VoIP software was
produced in spare bedrooms, garages, and office cubicles and used existing hardware that was
cobbled together to accomplish the task of putting voice over the ubiquitous and inexpensive
Internet without any regard for, or in most cases knowledge of, traditional telephony switching,
signaling, or other systems considered vital to high-quality, highly reliable universal telephony.
Use of the primarily flat-rate Internet, instead of the metered telephone network, allowed VoIP
users to avoid paying long-established tariffs on the traditional local, national, and global voice
networks. The desire was to put voice on the Internet as one more application of the globally
connected Internet, which could lower costs and, potentially, put more people in touch with each
other at a lower cost than keyboard actuated IP tools such as email alone had been able to do.
Calls in this phase were mainly PC-to-PC.
The second step of the transition was “toll bypass,” and was predominantly the use of existing
telephone hardware with customized software, which allowed avoidance of toll charges and a
growing perception that VoIP was basically regular toll voice at a lower price. Calls in this phase
continued from PC-to-PC, phone-to-PC, and PC-to-phone, but began to include phone-to-phone
calls using the Internet for the long-haul, long distance part of the call. International calls were
offered at particularly attractive pricing. Toll bypass naturally morphed into “free voice” for
everyone.
The result of the widespread news of “free voice” was that traditional carriers slashed their prices
to retain customers and avoid disappearing completely, which turned out to be an over-reaction
on their parts, but appeared to be an entirely sane move at the time. In retrospect, we see that
price reductions were mandatory, but a less instantaneous, more gradual approach would have
still worked in the marketplace and would have presented less risk to the financial health of the
traditional purveyors of voice services. In turn, lower prices for traditional telephony had a
significant dampening effect on new world carriers planning to implement VoIP networks
because most of the initial economic models of early VoIP implementers were predicated on
providing a lower quality and reliability service, compared with traditional circuit-switched
phone networks, but at significant discounts. These plans were put into question when the
traditional, high-quality and reliability services became available but at competitive prices.
The biggest problem with VoIP is that residential and business customers alike are learning
through difficult, real-world experience that VoIP is not “what you were getting from the phone
company, just cheaper.” VoIP lacks the reliability, scalability, standards, signaling, services,
ubiquity, and capability of the uni-media phase. What VoIP pundits fail to realize is that it is not
possible to reproduce in a weekend what it took over a century to produce in the uni-media phase
and then to easily apply it to multi-media applications, many of which are not yet fully
understood. VoIP, though, does provide a real-world telephony laboratory and a natural bridge to
the multi-media phase, where the real issues of transitioning to a new world order of Telecom are
addressed.




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It turns out, however, that the keys to success in the coveted multi-media phase lie not as much
in the hardware but in the higher-layer application software architecture, including the Session
Initiation Protocol (SIP), IP Multi-Media Subsystem (IMS), and Parlay-compliant Applications
Programming Interfaces (APIs)—and there is still a lot to be done before the current state of the
art replicates what we currently have in the uni-media phase, let alone surpasses it and brings
new and engaging applications to light.
This guide will use VoIP in the more generic sense, but will focus on more specific items under
the umbrella of IPT. Why? Because VoIP was originally intended for PC-to-PC voice messaging
and was never intended to replace global telephony. The idea of “free voice” was compelling and
simple, yet the model is not sustainable. There is still the underlying fact that someone has to pay
for the global Internet to carry voice and that as a niche, hobbyist toy it is possible to sneak some
voice in with data, but in the quantities that are needed to fully shift global telephony traffic to
the Internet, the voice packets will be noticed and must pay their own way.
VoIP is really more about the edge of the network. VoIP, in our definition, is a very visible,
tangible thing and requires new devices and a change in the traditional ways of communicating.
It requires PC-based agent software, SIP phones, or PDAs with VoIP capability, but is this not
just a new way of providing dial tone? And, if so, and this author feels that it is, is there anything
compelling about VoIP from a business standpoint? Does it enable you to do anything that you
have not been able to do before? Not in its early stages, but it holds out the hope of advanced
applications in the future and it is the new, advanced applications that are needed to make this
thing generically labeled as “VoIP” to be successful.
Multi-media and wireless are crucial to VoIP success, as VoIP plays an important role in voice-
enabling multi-media and the traditional data-only wireless systems already in widespread use in
business and personal communications.


VoIP Business Aspects Not Clear
VoIP may or may not cost less than traditional telephony, but that is not always the main
objective. Take the case of Heritage Bank of Alabama. Heritage was faced with a choice of not
being able to adequately support their growing client base or investing in old PBX technology or
taking a risky step as an early adopter of VoIP. Although VoIP systems have dropped in price
and the quality and variety of sources of implementation expertise have grown since Heritage
first took the leap, it is still true that the changes in training, operations, culture, and user
acceptance all represent big hurdles. It is also true, as shown in the following case study, that a
shift to VoIP is often inopportune and does not always yield the compelling 30 to 40 percent
benefit demanded by today’s businesses. In many cases, the issues are more practical than a
compelling savings and are more about the survival of the business because voice
communications is a fundamental service without which a business cannot function. Shown in
this light, a savings of less than $10,000 per year for a regional bank is better than a loss but is
not the focus.




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                                 Growing Up with VoIP at Heritage Bank
Heritage Bank, based in Birmingham, Alabama, realized some cruel facts about being forced into VoIP
prematurely by exhausted traditional technology. There are more lessons than may be obvious for any
company that is not considering the age and obsolescence of the existing system.
ROI
The manufacturer gave Heritage Bank price concessions in compensation for substantial problems early
in their implementation cycle. The ROI presented takes into account the price concessions but not price
reductions since 2002 and, therefore, may represent a reasonable expectation of ROI for a system that is
implemented in today’s competitive pricing environment.
The following rough-order-of-magnitude ROI analysis shows the hard savings the bank has realized from
their consolidation of service providers and shift to VoIP. In addition to the hard dollar savings, there are
increases in customer satisfaction, system supportability, and security and enhanced internal user
confidence that are impossible to measure. Heritage Bank’s ROI has two components, both of which
complement the other: service consolidation and VoIP.
It is customary and prudent to separate the ROI components to clearly show the impact of the service
consolidation and VoIP separately. It can be argued that VoIP can be implemented separately from a
service consolidation and vice versa. In most cases, this is true, but in the case of Heritage Bank, there
were so many inter-related elements that the project must be taken as a whole.
A further argument could be made that the bank could utilize VoIP-enabled low-cost voice services to
further lower their costs. This may well be a future initiative but the bank is extremely happy with the
quality of their current service and support presently being provided and will approach such an initiative
very warily. Such an option was unrealistic in 2002.
A 46-month ROI is high; the norm today being 24 months or less. What the ROI hides, however, is that
the old system was at its limits: it was unable to sustain Heritage’s growth. An outside observer could
quickly tell the strategic importance of this project to the bank by the fact that they spent just more than
US$200,000 for consulting and services.
TCO
A TCO comparison of the old key system and the new VoIP system provides further evidence that the
bank’s move was more strategic than simply cost savings and correlates well with the ROI analysis. Total
savings after 5 years, not adjusted for inflation or any other economic factors, was a mere $40,000, but
the bank is still in business and thriving.




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IPT
IPT was born of VoIP and proved compelling for traditional carriers, Internet Service Providers
(ISPs), and end-user organizations with stricter TCO and ROI guidelines for new technology
adoption. Right out of the gate, IPT was intended for both PC and non-PC device
interconnection, was designed to replace traditional telephony because it incorporated key
functions ignored by VoIP—such as signaling, and can be used as a transitional technology
harmoniously and seamlessly providing a telephony environment of mixed legacy and VoIP
devices.
IPT is complex but well understood and can be successful simply by providing traditional dial-
tone over cost-reduced multi-media packet infrastructure. IPT can be used as a stepping stone to
a future multi-media and wireless environment, or can be used to create a mixed environment.
IPT can provide cost reductions of 20 to 40 percent over traditional telephony, and TCO and ROI
models are well understood, as shown in the two examples that follow—one is an example of a
packet voice infrastructure at a county school system and the other is a PC-based application at a
global consulting firm.
                        IPT at Paulding County Schools: A Formula for Success
Paulding County School District in Paulding County Georgia has implemented a school system-wide VoIP
setup with an IPT emphasis incorporating almost 600 phones and completely replacing their older PBX
and CENTREX service and the entire physical infrastructure in all schools.
ROI
There are three ways to approach the ROI for this project. The first model is Voice Rides Free, the
second is Port Allocation, and the third is Capacity Allocation.
Sources of Savings: Although the system provides additional “soft” benefits, the ROI and TCO
calculations are treated as if this is a straight functional replacement of a phone system. No savings will
be calculated for the move of PRIs to USLEC—this could be achieved without a migration to VoIP.
Voice Rides Free: The first model assumes that the system needed to be installed for data and, because
it was so substantially overbuilt, the VoIP traffic represents a small impact. Therefore, the ROI can be
calculated as the cost for VoIP systems and phones of $450,000 divided by the $33,000 per month
savings to yield an ROI of 13.6 months.
Port Allocation: The alternative to Voice Rides Free is for both voice and data to share the costs of the
system. In this first allocation model, Port Allocation, costs are allocated based on the total number of
ports. If you were allocating the costs of a road system, this would be the equivalent of splitting the total
cost by the number of driveways, regardless of the type of vehicle that would use the road. With a total
allocated cost of $716,850 and a savings of $33,000 per month, the ROI would be 21.7 months.
Using these figures, an organization considering a new system can calculate the rough costs of their own
converged voice and data solution. Keep in mind that financial modeling shows these costs to be roughly
consistent with the costs of a traditional PBX, so these figures are not valid for comparing VoIP and PBX.
If a change of telephony system is mandated, the numbers show a move to VoIP is viable, while a more
reasonable decision point might be the answer to the question “Are there compelling reasons for a move
to a new telephony system now or can we wait?”
Capacity Allocation: Capacity-based allocation takes into account the demands placed on the network by
voice and data and shares the cost of both network use and network idle capacity. If the network were a
road system, this is the equivalent of allocating costs based upon the size of the vehicles using the road.
Based upon an average demand of 384 thousand kilobits per second (kbps) per data user and average
demand of 70kbps per voice user, we could allocate 18 percent (70/384ths) of the infrastructure costs (18
percent of $1,452,000 or $261,360) to voice, add the $450,000 cost of VoIP equipment, and divide the
total of $711,360 by the $33,000 of savings per month to yield an ROI of 21.55 months.


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TCO
TCO is always a tricky calculation but a very useful one, especially when used to compare two or more
systems for which the TCO has been calculated using the same algorithm. In the case of the Paulding
County School District VoIP project, the system life of 5 years will be used for calculation of all recurring
costs. Because there is no dedicated staff, no personnel overhead will be calculated, but the outside
maintenance fees will be considered. A useful figure for comparison purposes is the total cost/user, and
this will be calculated based upon 450 users. Fixed costs will be based upon the Port Allocation method
shown for the ROI calculation.
Based upon these rough order of magnitude calculations, for budgetary purposes, the cost to provide
VoIP-based phone service for a single Paulding County School District telephony user is $2953 for 5
years, or $49 per month; a figure substantially below the $74 per month cost of the Centrex service being
replaced.
There are two forces at play. The first is the usual impact of replacing a 10-plus year-old technology with
a new technology: the double impact of a lower price and more capability. The second force is the
financial impact of Paulding County’s very favorable monthly fiber cost. With normal fiber pricing, a
budgetary figure of $80 to $100 per month per user to provide VoIP-based phone service would be more
realistic for an organization of similar size, which would mean that even traditional Centrex may be more
cost effective and the cost difference may have to be justified based upon strategic application benefits or
on a “Voice Rides Free” strategy, which also makes sense for many organizations. This example clearly
highlights the importance of keeping infrastructure costs under control.

The Paulding County Schools example is a compelling one, especially when considering the
TCO on a per-user basis. Calculating TCO per user allows a direct comparison of costs from the
IPT system to the traditional system and provides very convincing evidence that Paulding
County Schools made the right decisions along the way. This example shows that much of the
value of IPT projects can come from the infrastructure part of the project and that all aspects of
the system implementation are inextricably interwoven and must be thought of differently than
traditional voice or data or video projects—projects often accomplished by multiple standalone
departments.
The next financial case study represents substantial savings with a shift to PC-based IPT. A
group of globally dispersed consultants were able to use VoIP application software to overcome
the inherent limitations of their cell phones, leveraging the global ubiquity of 802.11 LAN
standards. One can only imagine the positive shifts that will occur as WiFi hot spots continue to
proliferate globally, and how the convenience and ROI will improve as the engineers increase
their use of wireless for VoIP, data, and video.




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                         Semco Maritime Does Global VoIP—Without Phones
Semco Maritime, a global engineering services firm primarily serving the energy and shipbuilding
industries, has harnessed the power of IP to create a high-quality, money-saving, PC-based VoIP service
for their 600+ geographically dispersed employees. The two-tiered service combines MPLS and Internet-
based VPNs in a true VoIP solution.
ROI
ROI for Semco Maritime was only a secondary consideration: they required global voice connectivity but
soon learned that it was accompanied by an impressive ROI: the initial investment of less than 600K
Euros was paid back in 4.8 months. This ROI is based on the clients’ own numbers and savings figures of
200 Euros per month per engineer. In the author’s anecdotal experience, hotel room phone charges are
much higher than the 10 or so Euros per night saved on an average 20-hotel nights per month. My own
calculation is that at least 30 Euros per night would be saved, except by the most frugal and disciplined
engineer, who always makes calls from a lobby phone with a phone card and does not use the room
phone. If this were the case, the ROI would be 1.6 months.




TCO
Although the TCO calculation shown below could easily be challenged by any competent accountant, it
does provide a rough order of magnitude idea of Semco Maritime’s TCO for this application. Additional
considerations could include allocating current IP data costs to voice based upon percentage of traffic or
bandwidth and similar calculations, but those arguments could be countered by saying that the data
expense already exists and, therefore, voice rides for free. The fact remains that Semco Maritime’s total
cost per user per month for as much global telephony as they can use is in the range of 20 to 30 Euros.




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An analysis of companies moving to IPT, in most cases, shows a decided price/performance
advantage. Price/performance advantage tends to highlight the word “price,” or more importantly
for carriers, service providers, and larger end-user organizations providing services to internal
customers, the underlying “cost.” The most important aspect of the multi-media phase, however,
must be the price (or cost) to performance ratio. It seems to go, almost without saying, that the
economies of scale and benefits of integrated network measurement, troubleshooting,
management, and operations created by running voice and other enhanced multi-media services
over high-bandwidth shared IP networks should be able to provide efficiency at low cost. What
we have learned during the transitional VoIP period, however, is that it often does go without
saying, but shouldn’t, as the “converged network is more efficient and lower cost” argument
does not always work: problems and unforeseen costs scale up as well as savings in large
networks and are magnified for larger organizations—and not always in direct proportion to the
size of the network.
For many organizations who have maintained a largely unchanged traditional telephony
infrastructure, there is an immediate and positive impact of a change to a new, modernized
system simply due to Moore’s Law, which states that there is a natural price reduction
accompanied by an increase in capabilities that is simply a byproduct of the advancement of
technology. Beyond that, wild claims of “free voice” and “elimination of voice personnel” have
created unrealistic expectations that are very difficult to overcome, often leaving those who are
to finish crossing the VoIP bridge to the multi-media era with insurmountable problems,
primarily in terms of user and management expectations that are critical to a successful
transition. Without proper expectation management, any project, and especially one that impacts
such an important capability as human-to-human communication, is doomed to failure. This
includes proper expectation management in terms of cost savings, operational simplicity, voice
quality, and many other areas.


Major Changes
Sorting out all the changes and putting them into categories, we see that the two major changes
are that telephony is becoming an application, rather than a service and that end-user
organizations are accepting more responsibility for voice applications than even organizations
with historic PBX infrastructures. We can also observe that the shift from a service, provided
largely by outside organizations, to an application, traditionally managed by internal
organizations, also heralds the second major change, a shift in responsibility placing the larger
end-user organization in the role of carrier or service provider to internal clients.




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Telephony Becomes an Application
It has long been assumed by the IP crowd that the migration to Everything over IP (EoIP) is
almost an evolutionary “given,” more so, even than a revolutionary coup, but the arguments have
been more emotional than real. The argument has always pivoted around the superiority of IP
over circuits coupled with the name recognition and market acceptance of the Internet and its
underlying protocols. The piece that has been missing, however, is that the real benefits of the
Internet, and the market’s fascination with it, come from its simplistic interfaces and software
capabilities, and not from its underlying architecture. If the truth be told, wouldn’t each and
every user want a dedicated, guaranteed-performance path from themselves to each other system
with which they interact? And isn’t this exactly what a circuit offers them? No, users are not
enamored with IP and its switching characteristics; they are in love with the applications, which
can be characterized by two words: simple and multi-media. And these applications will be
offered in packages that are simple and easy, possibly under the banner of Virtual Private
Network (VPN) services, and quite possibly offered seamlessly over a variety of transports, both
wireless and wireline.
Migration from traditional telephony to the multi-media Voice over Packet model should be
strictly an internal network issue and should be transparent to the actual user of the service. The
migration should be seamless, and the only evidence that a subscriber should have that the
migration has occurred is that they now have a menu of new, meaningful multi-media services
that were not previously available.
Another key point of consideration as end-user organizations take a greater responsibility for
their voice services is that monitoring and management tools are no longer embedded in the
network but rather form a new part of the skill set the end-user organization must adopt. As the
responsibility for services shifts from the carrier and service provider to the enterprise
organization and eventually to the end user, and as the end user becomes empowered and more
sophisticated, so too, must the monitoring and management tools provided by the network. If
there is one thing that VoIP has taught us, and a mistake that cannot be repeated, is that although
it is possible to render voice waves and bits and to put them into packets that there is a lot more
to the entire puzzle than just this. Voice is an IP application, but it is not “just another
application.” We have learned that “cheap” alone won’t make voice on the net work, and that the
next phase must include much more of traditional telephony monitoring and management,
coupled with new tools, than we ever thought possible before.




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Responsibility Shifts from Carrier to End-User Organizations
The technical challenges of running advanced enhanced services across large, often multi-carrier,
IP networks are magnitudes of order higher than providing vanilla VoIP PSTN-replacement call
connectivity. When implemented by “traditional” enhanced service platform providers, the
database-centric execution of many popular enhanced services, such as phone/debit calling cards
and ubiquitous unified messaging, holds the unpleasant promise of introducing new orders of call
latency that equipment suppliers just thought they were close to “solving.”
Within the VoIP transition, the actual users have only had a glimpse of the problems, while the
service providers face scalability issues every day. The first-generation client/server service
platforms have extreme scalability limitations, typically being able to run a half dozen or so
“remote” switches before they experience melt down. There are, at present, no VoIP or multi-
media phase networks that even approach the scope or scale of traditional uni-phase networks.
Consider that the leading VoIP provider, Vonage, just passed a million subscribers in Spring
2006, and that a million is not equal to a single midsize United States city. In addition, existing
VoIP signaling protocols don’t address potentially service crippling latency issues within VoIP
networks. To more effectively meet latency requirements, multi-media networks must not only
provide fast database access but also distribute control intelligence throughout the network, a
lesson learned from uni-media.
We have also learned that multi-media networks are complex organisms requiring specialized
monitoring intelligence, and that manufacturer-provided monitoring tools are a part of the
solution but do not take into account the nuances and subtleties of the overall system. However,
generic monitoring and management tools are insufficient—to be truly effective, third-party tools
must be constructed with a knowledge of the individual manufacturer’s systems being
maintained in order to provide a system-wide, end-to-end view of system performance.
The key areas that next-generation monitoring and management tools must cover are:
   •   Pre-deployment simulation and modeling
   •   Manufacturer-specific monitoring and management
   •   Real-time business views
       •       Call detail records
       •       Calls in progress
       •       Delay-to-dial-tone rates
       •       Gateway channel utilization and loading
   •   Real-time call monitoring
       •       Phone and multi-media device availability and monitoring
       •       Successful vs. failed call completion rates
       •       Poorly performing components
       •       Service level breaches and SLA compliance
       •       Real-time interface to Manager of Managers (such as HP OpenView)


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   •   Summary and exception reporting
       •       Utilization trends over time
       •       Managed devices by company, department, and location
   •   Asset tracking
   •   Capacity planning
       •       Incoming and outgoing calls
       •       Loading by dial plan, routing rules, and gateway
       •       Bandwidth utilization
       •       Delay and delay variation
       •       Packet loss
       •       Route patterns, utilization, and availability


The IPT Life Cycle
In order to be successful in your next-generation network venture, expectations should be set as
follows. Only in so doing will the results you achieve have any hope of being judged a success
when compared with the expectations you set:
   •   Users—Different is not bad. The new system will initially provide all the important
       functions of the current system and then, after the baseline is established and the
       migration is complete, will provide important new capabilities. The system will sound
       different, but this is not necessarily a bad thing. The system voice quality will be (not as
       good as, nearly the same as, better than), fill in the blank based on your objectives, and
       the new system will be more flexible and give you more freedom while allowing you to
       stay connected.
   •   Management—The cost of the new network, when taken as a whole, may be the same or
       higher than what we are paying now but we must move forward to assure the continuity
       of our voice communications and eventually acquire advanced capabilities of strategic
       benefit, though our unique combination of geographic coverage and scale may allow us
       to realize overall raw cost savings. There will be four specific phases, each with its own
       specific objectives. Investing more time, effort, and attention in earlier phases will lower
       the overall costs and increase the value to our organization. The phases are planning and
       assessment, pre-deployment testing and implementation, ongoing operations, and
       optimization. Contrary to popular belief, VoIP is not “free voice” and VoIP is not “just
       another application.”
We will now take a closer look at each of these phases in the IPT life cycle.




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Planning and Assessment
The objectives of the planning and assessment phase are to develop a replacement system that
initially delivers replacement functionality and eventually adds important strategic functions.
The shift to the multi-media phase is characterized by a new, fresh commitment to traditional
telephony characteristics, such as reliability, security, and fraud control and customized services
deployment. In addition, desirable new aspects—such as a multi-media focus—will be available
on a large scale. All of this must be taken into account in the planning and assessment phase. It
can also not be stressed enough that management expectations must be set so that a sufficient
amount of time and resources can be invested in this phase to make the phases that follow it
successful. VoIP is not just another application that can be launched by plugging in a few IP
phones. A proper planning and assessment phase will take all of the needed factors into account.




Figure 1.1: The path from uni-media to multi-media.

“Making the move” to multi-media is not a very accurate statement as those carrier, service
provider, and end-user organizations adopting multi-media will not, in most cases, actually
‘move’ at all. In most cases, they will already be providing or using traditional and/or VoIP-
based systems and will simply transition to multi-media non-disruptively.
The first step is to ascertain the current services and features that are being utilized in the uni-
media phase to assure their accurate reproduction in the multi-media phase. A methodology for
accomplishing this step is presented in the next chapter. The next step is to move as quickly as
possible through the transition phase, though many organizations would be well advised at this
point to skip over the VoIP step and move directly to the ultimate converged IP-centric multi-
media IPT phase. This step avoids the risk of being in the transition period and reduces negative
user impact.


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Pre-Deployment Testing and Implementation
The objectives of pre-deployment testing and implementation are to assure the outcomes from
the planning and assessment phase and to determine the impact of adding the new applications to
the existing network, then to roll out the new system into the organization. The use of
sophisticated, specialized modeling and planning tools in the first two phases will allow us to
assure the maximum benefits in the shortest amount of time with the least possible disruption to
our operations.

Ongoing Operations and Optimization
Ongoing operations and optimization are two phases with opposite objectives. It is the objective
of the ongoing operations phase to ensure consistent functionality across the network, regardless
of where an individual is in the organization geographically. Ongoing operations is constantly
measuring the network’s performance against established benchmarks to ensure unwavering
compliance and absolute consistency of network services. Optimization, in contrast, has as its
primary objective improving operational aspects of the network services and adjusting the
operational baselines to reflect new baselines.
One of the key items that will require revisiting, and refreshing, is the SLA. The SLA is an
effective tool the full potential of which has never been realized. In its most basic form, the SLA
is a contractual vehicle that, at the very minimum, documents the minimum level of service that
a customer is willing to accept before they are due some penalty or refund on their service, but it
can, and should, be so much more. It can, and should, also provide a description of the actions
taken by a carrier or service provider should the user not keep up their part of the arrangements,
such as using a service class for which they have not contracted.
SLAs should also be automatic. At the present time, too few carriers and service providers have
embraced the SLA as a strategic customer relationship management tool and require the
customer to jump through hoops to get the penalties they are due. SLAs and the compliance
information supporting them should be available to the customer when and as needed. We will
see a lot more of this in the multi-media future, and sophisticated monitoring and management
tools will be required to keep customers apprised as to the QoE delivered by a network, versus
what was promised and contracted. In fact, the education process that goes along with this effort
should be initiated as early in the process as possible because the process of educating the user
on SLA compliance is also the process of educating the user on how to select multi-media
services without relying strictly on price.
The process about which we are speaking gets to the heart of the matter of creating a more
informed consumer of multi-media services in such a way as knowledge about vehicle
performance makes a more informed purchaser of cars. Mean Opinion Score, PQSM, R-Value,
and similar voice quality metrics should be as familiar to the consumer of voice services as
MPG, horsepower, and fuel options are to owners of vehicles. And this analogy makes additional
sense not just in terms of SLA performance but, as previously mentioned, in terms of a more
informed purchaser who will forego the commodity approach of “voice by the minute” or “gas
by the gallon” for a more informed purchase decision based upon metrics other than pure price.
Tools that allow the monitoring of voice traffic and related quality metrics in real-time and
provide feedback to suppliers and consumers on an exception basis, using baseline thresholds
established in the SLA, are crucial to success in the multi-media future, as they keep the focus
off price as a singular differentiator.


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One possible model that might emerge is that transport becomes a commodity over which
differentiated services flow. In this model, the prior performance of various transports,
representing some combination of different QoE offerings from one or more transport providers,
is known and the closest QoE to that needed by a given service at a moment in time can be
selected, possibly on a “pay as you go basis.” An alternative to that would be that bids would be
requested, in real time, from available transport services to provide a certain QoE at a moment in
time. How would this work in practice?
Suppose that you were in a coffee shop in Milan and needed to make a good night call home that
required high-quality, bi-directional audio and real-time video. You would have already made
whatever requirements were needed with the coffee shop—free wireless hot spot, pay-as-you-go,
used your corporate account with the wireless provider, whatever was needed—for access to the
Internet. In this case, you would select the called-to party on your PC/handheld/wireless video
phone and a request would go out for the long-haul service needed to handle your call. Because
of the low-loss, high-quality needed, you would basically be negotiating for a toll road for your
call. The negotiation would be made with a transport service that would provide what you
needed, at a negotiated price of .7 cents per minute, right now with no guarantee of this price in
the future. Your measurements from prior calls, and/or other users, would be used to validate the
bidder and your call is put through. The next call might be a highly compressed voice-only call
to check voicemail and might not require the toll road. There might be no additional cost for this
service beyond what is included in your basic monthly rate. Anything is possible in the multi-
media future.

Optimization
While the ongoing operation phase is about consistency, the optimization phase is about a
regular, constant, ongoing program of positive change in network operations. Optimization takes
the measurement and SLA compliance statistics gathered during ongoing operations and seeks
ways to improve those statistics with specific target outcomes. Desirable objectives will include
financial or operational goals, such as releasing resources on traditional telephony networks as
calls shift to the VoIP network or even reducing the amount of bandwidth used per call on the
VoIP network as newer voice coding techniques are implemented, both of which can have
favorable operational and financial outcomes. The process will involve conceptualizing
improvements, validating assumptions using modeling or simulation tools, then testing them in a
laboratory or isolated network segment before implementing the changes in the actual network.
After network performance has been assessed and any needed fine tuning done, the network
operational benchmarks are adjusted and new optimization goals are established.




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Summary
We are at the end of the first chapter which, as a standalone white paper, would simply be an
overview of the possible future of telephony: interesting and academic and about as useful, or
useless, as dozens of other white papers on the topic. But, this chapter is so much more. This first
chapter has documented the underlying beliefs, philosophies, and observations needed to
understand the rest of this guide and will be the last we will see of high-level concepts, abstract
ideas, and sweeping generalities. From this point forward, this guide will morph into more of a
practical guide, a “how to guide,” based on knowledge gained from planning and implementing
VoIP and multimedia communications for some of the largest multi-national and global
organizations.
Chapter 2 will cover the IPT life cycle, and the following six chapters will follow the life cycle
as we describe the processes, best practices, and pitfalls at each step, including planning and
assessment, design and pre-deployment testing, implementation and migration, ongoing
operations, optimization, and some very nitty-gritty details needed to assure success. Thus, if you
are new to telephony or are an industry veteran, stand by for what we intend to be the most
detailed, in-depth, and useful guide to VoIP and IPT implementation that it is possible to cram
into 200 pages.




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Chapter 2: The IP Telephony Life Cycle
Every system has a rhythm, a natural life cycle. If you can understand a system’s rhythm and “go
with the flow,” your job of managing that system will be made much easier. The traditional
telephony systems that we are now replacing have a natural rhythm, as do the IP Telephony
(IPT) systems with which they are being replaced. This chapter is about understanding the life
cycle of your enterprise IPT system and harnessing that understanding to increase the success of
your implementation as well as your financial rewards—be those rewards direct tactical benefits
such as cost savings or indirect, and often elusive, strategic benefits. This chapter is also about
squeezing as much use as you possibly can from your existing telephony systems, a move that
can make a lot of business sense regardless of your natural desire to discard the old and to start
using the shiny new thing.


The Bigger Business Picture
From the 1960s to the early to mid 1990s, a technology project often took 12 to 18 months or
more with little variation in schedule from large to small organizations. Any technology project
was a major undertaking and included an exhaustive process of a Request for Proposals,
presentations, site visits, and analysis, often by an internal staff and a cadre of outside experts. In
many cases today, however, it is only the “Fortunate 500” who still have sufficient resources to
be able to take a careful, project-oriented approach and to think and rethink the various outcomes
of possible choices or approaches—and often even those organizations will not apply sufficient
resources to assure success. These days, smaller organizations are often forced to react rather
than plan and to put out fires rather than building a fire-proof system in the first place.
The following section explores the facts surrounding a project that was carefully thought
through, argued out, and managed in a thorough and professional manner as an example. The
project described was the development of a global Voice over Packet (VoP) blueprint and
subsequent implementation for a Fortune 100 client. Although the client company is a global
energy giant operating in 162 countries, there are lessons in this story that can benefit
organizations of any size, industry, or geographic reach.
After being briefed by the client on the intended scope of the project—a global migration to
VoIP—my first step was to define terms and to redraw the scope of the project through a
collaborative, interactive, and often heated and animated debate. The initial thinking on the
client’s part was to create a new world of telephone communication that would connect all
corners of their vast global organization relying largely on their existing IP network coupled with
a new application called VoIP. Their primary motivation was to retire their old telephony gear
and in fact, all their traditional telephony headcount and associated costs. Their drivers were as
much a desire to get the benefits of “free voice” as anything else as well as to lower costs while
maintaining the same level of voice quality and connectivity.




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A key part of the success of the project redefinition effort was the rewriting of the project
objective. The objective was originally “to implement a VoIP system and reduce or eliminate
costs for internal voice communication” and was rewritten as “provide a global migration path
toward a multi-media network that will leverage existing technologies and systems, provide a
baseline of voice, data, and video services and allow implementation of important new
technologies when and where needed, leveraging existing assets, and to positively contribute to
organizational success.” Even though the updated objectives sound a lot like a really bad PhD
thesis title, the new objectives incorporated a variety of factors that must be included for the
long-term success of any project: a consideration of how the project contributes to the long-range
success of the organization and best utilizes existing assets.
This project occurred in late 2003, and one interesting lesson learned during this early phase was
how much things have changed since the 1990s. For instance, an effort to identify organization-
specific strategic and tactical benefits in order to establish milestones and success criteria for the
move to a VoP system did not get very far. The Global Network Manager explained that in their
world, tactical benefits were called “hard benefits” and hard benefits could be measured in
dollars and cents. He further explained that strategic benefits were called “soft benefits,” and that
there was no tangible way to measure strategic benefits. The project must, therefore, be
measured strictly in terms of hard, tactical benefits. As a further historical footnote, he added that
strategic benefits had not been considered since the late 1990s.

Packet Telephony Is Inevitable
Packet-based telephony is, in my strong view, inevitable, but VoIP, per se, is not. The first step
was to redefine the project as the Voice over Packet (VoP) project, as opposed to a Voice over
Internet Protocol (VoIP) project. The main difference is that VoIP originated as a way of
allowing two people to communicate by voice between two computers while VoP includes a
comprehensive toolkit that goes beyond VoIP to include other services required by traditional
telephony, such as signaling, accounting, security, and enhanced 9-1-1 emergency calling. In
addition, VoP extends the scope of communications to include a variety of services and
capabilities envisioned in initiatives such as the Session Initiation Protocol (SIP) and the Internet
Multi-Media Subsystem (IMS).
                                                   VoP
The broad term VoIP has been accepted by the industry, and consumers, to refer to a new, inexpensive
way of making telephone calls using the Internet. Although “insiders” understand that the IP network is
not always the Internet, the common definition is convenient and easy. What VoIP also means is that the
voice information is encoded and sent, historically, using proprietary methods or the H.323 protocol—and,
increasingly, via the Session Initiation Protocol (SIP). The term VoIP is exclusive and limits the range of
choices, however, the term VoP is inclusive and is a more general term meaning the coding of voice into
binary and transmitting the voice over a broader range of options—including Voice over ATM, Voice over
Frame Relay, and Voice over DSL. By considering a wider range of VoP options, not just VoIP, an
organization can often maximize use of its current networks, get around regulatory issues in certain
countries, and extend the life of existing assets.




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                                    Session Initiation Protocol (SIP)
Early versions of VoIP used proprietary “homegrown” protocols to convert voice waves into bits and
package those bits into discrete packets for transmission to a distant PC where they were depacketized
and replayed over the speakers, and eventually the headsets of the distant computer. Early efforts to
standardize the process led to the adoption of the H.323 protocol, really a family of multi-media call setup
and media negotiation protocols, from the International Telecommunications Union (ITU), a global
standards body under the United Nations. Eventually, a much more powerful multi-media control protocol,
the Session Initiation Protocol (SIP), which can define not only voice but other types of multi-media
sessions such as Instant Messaging and mobility, emerged from the Internet Engineering Task Force
(IETF), which is the de facto standards body for the Internet. SIP is generally accepted as the primary
standard protocol for VoP communications today and for the foreseeable future.


                                    Internet Multi-Media Subsystem
The Internet Multi-Media Subsystem (IMS) is an initiative of the next-generation wireless network
architects intended to create a single operational structure that incorporates wireless and wired systems
designed for multi-media communications. Although there are other, competing, approaches, IMS is a
prime example of a vision for a single architecture that incorporates static, nomadic, and mobile multi-
media systems.

Figure 2.1 is very similar to the diagram used to explain the benefits of taking a VoP approach.
Analog telephony is the base of the family tree. The importance of starting with analog
established that analog still exists in the network, and always will, because the endpoints that are
communicating—humans—communicate by modulating analog waves and, at least for the
foreseeable future, will continue to. The discussion of analog communication planted many seeds
that would bear fruit later. For the traditional “Bellheads,” the members of the team who run the
old phone network, the discussion afforded a nice walk down memory lane, jogging memories
and knowledge all but forgotten, or at least not appreciated. For the “Netheads,” the discussion
was the first of many reminders that there was much to learn about the world of voice if it were
to be successfully grafted onto their IP network.




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Figure 2.1: The voice technology family tree.

Key points about analog telephony are that it works by translating continuous analog acoustic
waves into continuous analog electrical waves and sending the electrical waves over a distance.
Transmission distance is limited because the electrical signal becomes “tired” (the correct term is
attenuated) over a distance and must be amplified to travel farther. The amplification process is
not ideal because it amplifies the desirable signal (speech) as well as undesirable background
noise. After a number of amplifications, the desirable signal is all but indistinguishable. Analog
switching is also slow and problematic and analog systems have capacity limitations. Even so,
analog is still used in today’s networks, even if just from a desktop or residential telephone to a
Private Branch eXchange (PBX) or telephone company switch.
One additional benefit of a review of traditional telephony was to provide a basis for
understanding voice quality. Regardless of the many theories, metrics, and measurements, the
true measure of voice quality has to do with how closely the analog wave arriving at the
destination, after its trip through the phone network, resembles the wave that was sent. This basic
fact is true regardless of whether the transmission system is analog or digital circuit or a digital
packet system. In fact, an early objective called “toll quality” was based on the quality of the
signal at the destination as judged by a panel of human listeners on a 5-point scale called the
Mean Opinion Score (MOS) that has been imitated but never duplicated.
After a thorough discussion of the analog era and the knowledge from that era that could be
applied to the current project, we moved to the discussion of the digital circuit era. The
comprehensive understanding of the digital circuit era that comes from operating a global digital
voice network was fresh in the minds of the Bellheads because this is the network that they were
operating every day and was the network from which they were attempting to migrate. The
Netheads, however, got another reminder of how much they did not know about voice
communications and how much there was to know.

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In this phase, we discussed the translation of analog waves, at or near the edge of the network, to
digital bits using Pulse Code Modulation (PCM) or Adaptive Differential Pulse Code Modulation
(ADPCM). We discussed the fact that, unlike voice coding techniques developed for VoIP/VoP
systems that are optimized for human speech, PCM and ADPCM are analog-to-digital schemes
that work for all sounds. In addition, although the VoIP/VoP systems would give a wide range of
clever, bandwidth efficient choices, we can still use PCM and ADPCM in the VoIP/VoP
architectures to maintain a higher voice quality. We also reviewed the global telephony network
that existed within the organization and found that Private Branch eXchanges (PBXs), literally
scaled-down telephone switches designed as the cornerstone of organizational voice networks,
connected 64Kbps voice circuits to larger switches, which connected to high-capacity fiber-
based connections formatted according to the rules of a standard called Synchronous Optical
NETworks (SONET) in the US and Canada and a very similar system call Synchronous Digital
Hierarchy (SDH) elsewhere. We also discussed the fact that regardless of the capacity of the
SONET or SDH optical systems, they were nothing more than bigger and bigger bundles of the
64Kbps voice connections, at least in the older format in which they existed with this network. In
the current network, voice was completely separate from data, video, and telemetry, and the
multi-media bits never, ever, commingled within the system.
Among other considerations was the fact that critical signaling information needed to establish
and take-down the voice calls traveled in packets in a parallel network completely separate from
the voice connections. On the plus side, the parallel signaling networks, governed by rules titled
System Signaling 7 (SS7) in the US and Canada and Common Channel Signaling 7 (CCS7)
elsewhere, were designed to be secure and virtually indestructible—without them, calls didn’t
get connected and phone company revenue did not flow. On the minus side, were the costs of
maintaining two complete networks, a circuit network for voice and a packet network for the
signaling, just to set up voice connections.
As we moved from our historical inventory to get a glimpse at the future, we had some area of
common ground after which the IP folks began to feel more comfortable. The common ground
was in the area of Voice over Fast Packet and its various branches. Unbeknownst to many, the
global telephony standards organizations had begun work on a packet-based infrastructure for
voice as early as the late 1970s with standards such as Broadband-ISDN beginning to appear in
workable form in the early 1980s.
The right side of the voice technology family tree (see Figure 2.1) is, in fact, representative of the
work of the telephony traditionalists and includes a range of technologies from Voice over ATM
to Voice over DSL, including Voice over Frame Relay and Circuit Emulation Services (see
Figure 2.2). The left side of the family tree shows initiatives, as early as 1972, to put voice into
discrete packets and send them over a shared network with data. These initiatives begin with
little-known Voice over X.25 and VoIP over X.25 and include the much more recent, and more
memorable, VoIP over the Internet projects in 1994 that were the true ancestors of VoIP as we
know it today.




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Until 1995, however, progress in packet-based voice was interesting but not of commercial
interest. It was, in fact, very much in the hobbyist realm, as we might think of ham radio, for
instance, until the invention of the packet-circuit gateway in 1995 made it possible to call from a
PC to a traditional telephone. It was not long after that calls from traditional telephones to PCs
were demonstrated and then, the service that proved most disruptive to the established traditional
telephony companies, calling from a traditional telephone to another traditional telephone
completely transparently over an IP-based bypass network. This is where the fun began and this
is, precisely, the objective of the VoP migration project: to gain the benefits of a combined voice,
data, and video network and bypass traditional telephone carriers for all internal and some
external calls while maintaining connectivity to the traditional telephone network to be able to
reach persons, be they on stationary traditional telephones or the growing population of mobile
phone users, as and when needed.
In addition to the raw technical arguments for or against a specific packet voice option,
experience and analysis showed two more important dimensions: legal/ regulatory and
operational. From a legal/regulatory standpoint, it is interesting that packet voice is not legal in
many of the jurisdictions in which this global organization operates and, even though there has
been a broad relaxation of anti-VoIP statutes, there are still nations in the world where VoIP, or
at least non-circuit voice communications, breaks certain statutes, violates specific tariffs or
contracts, or is out-right illegal. In these cases, the business and technical arguments have no
weight against the legal ones and legal and regulatory compliance is still a check-off for any
change to the telephony status quo.
As far as the operational dimension of VoP systems, the technology choices were weighed
against voice quality, delay, and delay variation issues, price, multi-plexing efficiency and
system complexity and some interesting relationships emerged (see Figure 2.2). Overall, voice
quality was deemed to be highest in the systems in which all aspects of voice transmission could
be most strictly controlled—that is, in circuit systems. The less control and the greater the
influence of other traffic, the lower the quality. Considering delay and delay variation, the further
you go away from circuits, the more delay and delay variation. Considering these items only, it
would appear that circuits are ideal and packet-based systems inadequate—that is, until price and
multi-plexing efficiency are factored in. VoIP systems that use bandwidth-saving silence
suppression and overhead compression schemes to provide the best multi-plexing, or sharing,
with data and video have the lowest overall costs though they are somewhat more complex to
operate than circuit-based systems.




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Figure 2.2: Analyzing operational dimensions of VoP systems.

                                 Multi-Protocol Label Switching (MPLS)
Multi-Protocol Label Switching (MPLS) is an approach to networking characterized by this author as “the
best of both worlds.” MPLS is a way of delivering Quality of Service (QoS) in Virtual Private Networks
(VPNs), the secure package in which many organizations are purchasing network services, by allowing a
dynamic choice to be made whether to route or switch information traversing a network. Routing, the
traditional Internet way of sending information across a network, has the benefit of not having an initial
delay to establish a path from end-to-end across the network, but each packet must be analyzed and
forwarded individually. Switching, in contrast, the traditional way that telephone networks work, does have
an initial delay but whisks each subsequent bit quickly to its destination. The real answer to which
approach is best cannot be answered until we know what information is being sent and, more specifically,
if it is a long-lived or short-lived flow.
For a short-lived flow, something like a small Web page, for instance, routing is best because each packet
is forwarded on its own: there is no need to go to all the trouble and delay of setting up a switched path
for such a small amount of information. A long-lived flow, however, such as a longer VoIP conversation or
a video over IP program, will benefit greatly from having a pre-established path and not having to go
through the requisite packet analysis and forwarding of each packet.
MPLS is “the best of both worlds” because MPLS is capable of routing all traffic until it has determined
whether a flow is long-lived or short-lived and setting up a switched path for those flows that turn out to be
long-lived. Early implementations of MPLS would always route over the available IP routing mechanism
and switch over Frame Relay, but newer variants are capable of switching over everything from ATM to
Wave Division Multi-Plexing to SONET/SDH to dedicated fibers, including Frame Relay and the IP RSVP
bandwidth reservation protocol.

In the case of the global energy giant, the impact of this analysis was felt more on the
prioritization of systems to migrate and a heightened comfort level with leaving some non-IP
systems in place for a longer period of time during migration, but had no appreciable impact on
the ultimate objective, which was still to eventually have an all-IP multi-media architecture. This
analysis did, however, support their decision to migrate much of their global IP traffic, both
voice and non-voice, over to a managed MPLS-based VPN to get the combined benefits of IP-
based applications, QoS, and security offered by such VPNs.




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In other instances, however, a broader look at VoP caused several large organizations to rethink
their VoIP objectives. In many cases, including a North America-wide system for a major
manufacturer of industrial tools, the organization realized, prior to embarking on a traditional
VoIP approach, or early enough in the cycle to retreat and regroup, that they did not really need
the power of multi-media or even magic of SIP: what they truly desired was less-expensive voice
communications at a level of quality comparable to, or just less good than, their traditional dial
network. These organizations often implemented Voice over Frame Relay or, less frequently,
VoDSL or Voice over ATM.

Impact of VoP
Having considered VoP almost in isolation, the next step was to consider the impact of VoP on
the existing network. The existing network was a predominantly data network with a rapidly
growing set of video services. The data, however, includes the traditional menu of services—
browsing, email, file transfer—and the movement of gargantuan volumes of near-real-time
telemetry data that most network operators do not have to deal with.
The initial “VoIP is just another IP application” mantra was quickly thrown out in favor of the
“VoP is a unique application with its own special requirements” mantra. One reason for this
change of mantra is that early, especially residential and consumer flavors of VoIP, used very
low-bandwidth compression algorithms which, while bandwidth efficient, provided a lower
quality, or at least a different quality of voice. Many businesses have opted to continue to use the
traditional Pulse Code Modulation (PCM) voice coding scheme even after their migration to
packet voice. Keeping PCM end-to-end, regardless of whether the bits are traveling in circuits or
packets, assures better compatibility between different systems and reduces the number of
translations between different coding schemes, which also improves voice quality. What this
means in operation is that instead of 6 to 8Kbps per voice connection, each connection consumes
80 to 110Kbps of bandwidth, though the actual utilization may be considerably less if silence
detection systems are used to not send packets containing silence.
Another impact consideration is that voice systems can provide less delay, less delay variation,
and less susceptibility to packet loss by putting fewer voice samples in each packet. Although
this approach does have its benefits, as stated earlier, it also has the drawback of taxing routers
and transmission systems by requiring a lot more work for a lot fewer bits being transmitted—
and all this is going to have to be considered when assessing the true impact of adding telephony
to the network.




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VoIP and IPT Return on Effort
The financial measurement of Return on Investment (RoI) is a calculation of the financial benefit
achieved for a specific investment of capital. An additional metric that takes into account
intangible—and, yes, admittedly, soft/strategic benefits—is Return on Effort. RoE had no value
to the global energy company, but many organizations find a consideration of RoE a good way to
take into account aspects of a move to a VoP system that cannot be captured any other way.
For instance, in one of the case studies reviewed in Chapter 1, Heritage Bank, the financial RoI
calculation was sufficiently terrible as to scream to anyone who would listen “YOU SHOULD
NOT HAVE DONE THIS!” but the RoI calculation neglected the fact that the system being
replaced by the new VoIP system was no longer available from the manufacturer, spare parts
were no longer available, and that the bank had the choice of a forklift upgrade or ceasing to add
staff, branches, and new clients. For the bank, as for many companies adopting VoIP, it is more
expensive than the current system and an upgrade does not make financial sense, but there is no
other option. RoE is also a way of identifying other intangibles, such as the school district that
was able to provide in-room phones for teachers where none had previously existed or the
pharmaceutical lab that was able to display employee and contractor photos on phones for
enhanced security.


Enterprise Telephony Migration
At this point, it has been determined that the organization will migrate to a new packet-based
telephony system. What lies ahead is the inevitable budget process, the migration itself, and,
once across the bridge, the IPT life cycle of planning and assessment, pre-deployment testing,
ongoing operations, and optimization.

The Budget Process
For the global energy company, as for many organizations giddy with excitement at the prospect
of finally being able to retire traditional voice assets and headcount and realize the benefits of
“free voice,” the excitement disappears quickly as the budget process begins.
The first realization was that the skills of the traditional telephony people are still in demand and,
in fact, are becoming more unique and valuable than their IP counterparts due to the natural
attrition of the traditional skill sets. In the case of the global energy company, and many other
companies, the following type of plan was anticipated—as soon as the first packet carrying voice
flowed from one IP phone to another, the un-needed old telephony regime could be shown out of
the building and their cost saved. What actually happened was that the best of the traditional
voice and IP team were kept on staff and the less useful, less-skilled individuals from both
groups were terminated, though nowhere near 50 percent of the staff was cut. For this situation,
it turned out that it took about 70 percent of combined prior staff to run the new, consolidated
network, though it was felt that cross-training over time to create a combined data/voice/video
multi-media support tech could provide greater than 30 percent savings on staffing costs.




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The second realization was that packet voice requires special hardware. The new hardware
would take the form of either new, specialized phone devices—at roughly the cost of earlier
digital and PBX phones—or gateway systems. The gateway systems would be needed to adapt
the traditional telephone sets to the new IP connection. In most cases, some of each new system
would be required, at least through some migratory period. This is an additional, and often
unexpected, financial burden on the organization, but a surprise that can be avoided with good
planning. Another related surprise that can be avoided is the fact that new phone sets must be
provided with AC power or powered via Power over Ethernet (PoE), both of which require
additional planning and costs.

   In some cases, organizations spend as much, or more, than they spend on the new voice over packet
   system to provide power to the phone systems because, unlike their analog predecessors, the new
   systems rely on “wall power.”

The third realization is that, like their predecessors, IP-based multi-media systems are complex
systems that require management, measurement, troubleshooting, and problem resolution—in
effect, life cycle management at each step through their operational cycle. In fact, the
management required by traditional telephony systems is often hidden “behind the scenes”
because dial tone has often been purchased as a service rather than a bunch of piece parts to be
assembled by the end user organization, such is the shift in thinking from telephony as a service
to telephony as a managed application. But do not be fooled, there are important management
functions that must be completed routinely at each step of the life cycle as will be described in
more detail later.

Reuse of Existing Assets
In the ideal situation, existing assets can be reused. This is true of human assets that can be
retrained or cross-trained in the needed disciplines. Hardware assets can often be reused as well,
with the addition of a translation system or gateway. One benefit of gateways is that they can
provide access to voice connections on existing networks as well as special features provided via
the Advanced Intelligent Networks (AINs) that traditional carriers have built over the years. The
physical positioning of the gateway also impacts the cost and ability to reuse existing assets as
well as the ease of support of the new system. Consider Figure 2.3. It might be possible, for
instance, to assess the needs of the user community and divide users into three groups: those
users who need new IP capabilities now, those who will need IP capabilities at some time in the
future, and those who simply need traditional dial tone.




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Figure 2.3: Determining which assets can be reused.

The first group would be provided with new IP phones and directly connected to the IP network.
Because this group is a subset of the entire user population, however, the cost of the new phones
is a fraction of what it would be to provide new phones for everyone, and the resource impact on
the IP network is also a fraction of what it would be if all users were migrated. Some scheduling
algorithm could be applied to the second group to determine at what point they would be
migrated to the new IP system, and the third group could simply be provided with “personal
gateways,” often called Analog Terminal Adapters, which would allow their traditional
telephones to be connected to the IP network. They would still use bandwidth on the IP network,
but the hardware cost and complexity of putting this set of users on the network would be
substantially lower than moving all users to new IP phones. At this point, the overriding
consideration might well become the calculation of the cost of maintaining the old traditional
network just to support those users who had not yet needed to migrate, and it may be advisable to
complete the migration sooner, rather than later, and retire the old network.
However, if a larger gateway was positioned on the network side of the PBX or premises switch
it is possible to reuse all the premises phone systems and wiring and attach all existing telephone
users to the IP network via the gateway. This may be an interim step until a local shared IP
softswitch is implemented or until a remote IP-attached softswitch is implemented. In any case, it
is an approach that can be used to extend the life span of the premises voice solution.

New Assets Needed
In any case, new assets will be needed in order to extend the life of the old system and to bring
new features and capabilities to the users. New assets will include, at minimum, hard or soft
phones, gateways, softswitches, MPLS-capable routers and MPLS-aware end systems, Virtual
Private Networks (VPNs) built with those tools, and additional bandwidth. New assets will
invariably include new management, monitoring, and troubleshooting tools as well as tools to
ascertain compliance of services with Service Level Agreements (SLAs).




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Assessing Urgency and Prioritizing Migration
Except in the smallest, single office situations, it is not advisable to flash cut all telephony users
to a new system. For this reason, it is necessary to assess the urgency and prioritize which areas
of the organization will be migrated in which order. As difficult as it might seem, political
pressure is the last thing that should be considered when determining the order of priority for
migration. In fact, the most visible or boisterous departments should be migrated last to assure a
minimum negative impact on the overall migration. Migration order should be based on more
reasonable considerations:
   •   Which office or region has outgrown, or is closest to outgrowing, their existing system?
   •   Which region has bandwidth or IP network infrastructure good enough to carry voice,
       and other resources to accommodate the new telephony upgrades?
   •   Which region has the most technically savvy support staff?
   •   Which region was involved with early testing and/or staging and is, therefore, most
       familiar with the new system?
   •   Which office or region is closest, because the earliest migrations are always learning
       experiences and should be expected to go slower and be more problematic?

Migration
Migration can be divided into three categories: avoiding migration, partial migration, and full
migration. Within the partial migration and full migration categories, the roll-out may be phased
based upon a variety of factors, as will be discussed later.

Avoiding Migration
Avoiding migration is usually “avoiding migration for now,” which means that the office,
division, or region under consideration either has a recently acquired system that has not yet been
depreciated or that the group will be sold or otherwise liquidated before it is required to join the
new VoP network.
One strategy adopted by the global energy company that not only eased migration but also
stretched the use of their assets, thus reducing their overall budget, was to establish regional
refurbishing and staging centers for equipment that had been taken out of service as a part of the
upgrade. As an example, let’s say that an analog key system in Tokyo no longer had expansion
capability because the upgrade components were no longer manufactured and that this is the only
reason why Tokyo was being upgraded to a new VoP system. Sydney had exactly the same issue
and the same key system. In this case, Tokyo might be upgraded to the new system, thereby
freeing components that could be used to upgrade Sydney and avoid migration to the new VoP
system, at least for now, or vice versa. If there were no immediate need for the components
removed from Tokyo, they could be refurbished and placed into an upgrade inventory for the
time when they would be needed to upgrade a system and avoid migration.




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Partial Migration
Partial migration is accomplished as previously discussed by using strategically placed gateways
to allow existing assets to be kept in place, thereby extending their useful life, especially where
only a limited subset of the new VoP capabilities were needed by the office or region being
partially migrated.

Full Migration
Full migration means that all non-VoP components are removed and replaced with their VoP
counterparts.

Delaying VoIP/IPT
It might seem a bit late in the process to ask this question, but can a move to VoIP/IPT be
delayed? One reason why it is not too late in the process is that up to this point all the planning
exercises are what military planners call “tabletop exercises”—no changes should have yet been
made, no systems implemented. Everything up to this point should be on paper, but not yet
realized in the real world.
Thus, what are the pros and cons of delaying a move to VoIP and IPT? If there are no
competitive or operational pressures to make the move, the organization should consider
delaying the move. Each day that goes by, prices for components and services get lower,
standards solidify further, and the various aspects of implementation and operations are better
understood. For each day that passes, traditional telephony components are further depreciated,
and for each day that passes, there is one more day without the potentially unnecessary
disruption to operations that virtually any change, especially the change to a system as
fundamental as your phone system, can bring. It is only prudent to stop at various times in the
planning process to revalidate your objectives and to consider any change in business climate
that would mandate either slowing down or speeding up the migration process.

Systems Integrator vs. Do It Yourself
Once the decision has been made to move to VoP and it has been determined the extent to which
it will be rolled out in the organization, it is necessary to confront the issue of phasing and
project management. Some organizations choose to migrate entire departments or offices to VoP,
while other organizations will phase-in VoP by job function—for example, phasing in support
staff first to allow them to gain experience with the technology. Once the phasing has been
determined, a project management schedule can be developed. It is at this juncture that many
organizations realize that they do not have the resources to do the task. Smart organizations
realize that if they are resource-constrained before a VoP rollout project, they must line up
additional resources before they can begin their VoIP rollout.
Some organizations attempt to be clever and hire and train college interns or other “temporary”
workers to assist during the rollout. Although the concept is right—workers will be in high
demand during the rollout but will not be necessary after migration—the approach is wrong. A
key consideration in this scenario is that knowledge gained during the rollout should be captured
in some way because it is valuable to the support process. For this reason and others, many
organizations consider engaging a systems integrator to assist in the rollout.


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When an organization weighs the systems integrator vs. DIY approach, the organization should
remember that the VoP migration is not simply a matter of plugging VoIP phones into existing
Ethernet ports and making phone calls. Instead, it is a highly complex, and highly visible, project
within the organization. The systems integrator should be evaluated based upon demonstrable
success, site visits to the systems integrator, and interviews with project managers and other key
persons. In addition, the systems integrator should be involved as early as possible in the
planning phase.
It is also important to consider the systems integrator vs. DIY debate as early in the migration
planning as possible so that the systems integrator may be considered in the budgeting process.
Although it is not possible to predict prices without more detail about an individual project, it is
not uncommon for an excellent systems integrator, who can assure project success, to cost
between twenty and thirty percent of the overall price of a project.

Managed Service vs. Do It Yourself
For small and midsized organizations, voice has largely been a “managed service” that has been
provided by a telephone company with the occasional small to medium enterprise venturing into
the “do-it-yourself” voice business. Large to very large organizations have been quite the
opposite and many have, since the late 1970s, provided their own dial tone via PBXs or technical
arrangements with similar names. As we move into the multi-media era, it seems as though many
organizations are viewing VoP as a new application for their IP intranets and Internet networks
and, therefore, increasingly as a do-it-yourself project. But, this does not need to be the case, and,
in fact, the migration to a packet-based voice system may very well be the opportunity to rethink
the way voice is delivered within the organization and between the organization and the outside
world.
If an organization does decide to choose a managed service approach, they can ignore many of
the technical details of the inner workings of the system and instead “take the temperature” from
the outside and ask the doctor how the patient is doing. This can be accomplished using service
provider-offered tools that provide a snapshot of the network or even embedded measurement
systems in the customer’s own equipment that might give a more accurate, or at least more
timely, report on the health of the overall system and its individual connections. In any case,
these tools provide nothing more than a second opinion that might—or might not—correlate with
the health ratings offered up by the service provider.
If an organization does decide to take the do-it-yourself approach, the implementation of VoP
services will require the use of measurement and monitoring tools that reveal far more
“personal” results—results that not only indicate how the system is performing but also what the
organization needs to do to repair any problems or, hopefully, the level of pride they may take in
their own good results.
In either case, the measurement and analysis tools are useless without baseline measurements
and a set of rules that typically take the form of a document called the SLA—and even though
the SLA for years has been thought of as a way of assuring compliance of the service provider,
and it still is, the definition of service provider probably needs some updating. In the managed
service scenario, the service provider is still who you think it is—“them”. But, in the do-it-
yourself scenario, the service provider has become “us,” because you are now an internal service
provider and the internal departments, divisions, regions, and employees are the consumers of
this service—they are the customers.


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A further consideration is the interplay of multiple SLAs from multiple parties. Take, for
example, a scenario in which a managed service provider is providing VoIP service on a MPLS
VPN provided by a separate provider. In this case, the two SLAs are inextricably interwoven.
Can you penalize the VoIP service provider for not connecting calls on an underlying IP network
that cannot connect across the MPLS network, or if the MPLS network does not meet specific
QoS metrics? And what if there are multiple MPLS networks provided by multiple providers?
This is an issue often seen with larger, multinational organizations. These items must be
considered to achieve a reliable and fully functional VoIP implementation. It is also possible that
these items have already been considered, and resolved, in some other networking context, so it
is worth researching before diving in.
Although the network is evolving into a complex creature delivering real-time, near real-time,
and non real-time voice, data, and video, the SLA must still be written in terms of metrics that
are meaningful to the specific services being delivered by the network to the user/consumer.
Let’s take a closer look at this critical management tool—the SLA.

SLAs
One of the key items that will require revisiting, and refreshing, is the SLA. The SLA is a
wonderful tool the full potential of which has never been realized. In its most basic form, the SLA
is a contractual vehicle which, at the very least, documents the minimum level of service that a
customer is willing to accept before they are due some penalty or refund on their service, but it
can, and should, be so much more. It can, and should, also provide a description of the actions
taken by a carrier or service provider should the user not keep up their part of the arrangements,
such as using a service class for which they have not contracted.
One possible model that might emerge is that transport becomes a commodity over which
differentiated services flow. In this model, the prior performance of various transports,
representing some combination of different Class of Service (CoS) offerings from one or more
transport providers, is known and the closest CoS to that needed by a given service at a moment
in time can be selected, possibly on a “pay as you go” basis. An alternative to that would be that
bids would be requested, in real time, from available transport services to provide a certain CoS
at a moment in time. How would this work in practice? Simply recall the example in Chapter 1
of the call from a Milan coffee shop.

Penalties
Most SLAs are lopsided, especially when it comes to penalties. In most cases, the SLA is written
in such a manner that any shortcomings on the part of the service will result in penalties from the
service provider. However, the method of proving that penalties are owed is notoriously difficult
and only the most exacting and picky end user organizations routinely see any penalties. Both of
these problems must be addressed.




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SLAs should impose a penalty on the service provider (be they external providers or an in-house
group) if they do not perform; in addition, SLAs should impose penalties on the user
organization if they don’t perform, such as if they continually ask for bandwidth beyond the
contracted rate or impose a special availability requirement for a certain location. Finally, SLA
compliance must be automatic. At the present time, too few carriers and service providers have
embraced the SLA as a strategic customer relationship management tool and require the
customer to jump through hoops to get the penalties they are due. SLAs and the compliance
information supporting them should be available to the customer when and as needed. We will
see a lot more of this in the multi-media future, and sophisticated monitoring and management
tools will be required to keep customers apprised as to the QoS delivered by a network versus
what was promised and contracted. In fact, the educational process that goes along with this
effort should be initiated as early in the process as possible because the process of educating the
user on SLA compliance is also the process of educating the user on how to select multi-media
services without relying strictly on price.

True Impact of SLA on Business and Operations
A fully integrated and automated SLA-based network monitoring and management system can
provide both long-term and “canary in the mine” types of visibility to overall and service specific
operations. In an environment of enlightened collaboration between user and service provider,
the SLA is a key tool to successful operation and a path to continuous improvement. SLA
metrics can, and should, be modified over time to reflect actual network objectives and realities.


The IPT Life Cycle
The four phases of the IPT life cycle are planning and assessment, pre-deployment testing and
implementation, ongoing operations, and optimization. These four phases are discussed in this
section.




Figure 2.4: The IPT life cycle.



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Planning and Assessment
The objectives of planning and assessment are to develop a replacement system that initially
delivers replacement functionality and eventually adds important strategic functions. The
migration process is characterized by a new, fresh commitment to traditional telephony
characteristics such as reliability, security and fraud control, and customized services
deployment as well as making desirable new aspects, such as a multi-media focus, available
organization-wide to all users who need them. All this must be taken into account in the planning
and assessment phase. It can also not be stressed enough that management expectations must be
set so that a sufficient amount of time and resources can be invested in the planning and
assessment phase to make the phases that follow it successful. VoIP is not just another
application that can be launched by plugging in a few IP phones. A proper planning and
assessment phase will take all of the needed factors into account.
To be successful in your next-generation network venture, expectations should be set as
follows—only in so doing will the results you achieve have any hope of being judged a success
when compared with the expectations you set:
   •   Users—Different is not bad. The new system will initially provide all the important
       functions of our current system and then, after the baseline is established, and the
       migration is complete, will provide important new capabilities. The system will sound
       different but this is not necessarily a bad thing. Our system voice quality will be
       ____________(not as good as, nearly the same as, better than—fill in the blank based on
       your objectives) and the new system will be more flexible and give you more freedom
       while allowing you to stay connected.
   •   Management—The cost of our new network, when taken as a whole, may be the same or
       higher than what we are paying now, but we must move forward to assure the continuity
       of our voice communications and eventually acquire advanced capabilities of strategic
       benefit, though our unique combination of geographic coverage and scale may allow us
       to realize overall raw cost savings. There will be four specific phases, each with its own
       specific objectives. Investing more time, effort, and attention in earlier phases will lower
       the overall costs and increase the value to our organization. The phases are planning and
       assessment, pre-deployment testing and implementation, ongoing operations, and
       optimization. Contrary to popular belief, VoIP is not “free voice” and VoIP is not “just
       another application.”
Organizations adopting multi-media will not, in most cases, actually ‘move’ at all. In most cases,
they will already be providing or using traditional and/or VoIP-based systems and will simply
transition to multi-media as non-disruptively as possible. The first step is to ascertain the current
services and features that are being utilized to assure their accurate reproduction in the new
system. The next step is to move as quickly as possible through the transition phase, though
many organizations would be well advised at this point to skip the VoIP step and move directly
to the ultimate converged IP-centric multi-media IPT phase. This step avoids the risk of being in
the transition period and reduces negative user impact.
A critical aspect of planning and assessment is ensuring that the new system performs at
minimum the key functions of the system being replaced and that transitioning to the new system
does not present any unexpected and unpleasant surprises. One way to do this is to do a functions
audit on the old system in the planning phase and even prior to acquisition of a new system as a
precursor to defining mandatory functionality during procurement.

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The obvious approach would be to ask the systems administrator or manager of the old system
which functions must be supported, but the result of this request usually produces an operations
manual or sales brochure listing all possible functions. It is not desirable, for numerous reasons,
to attempt to replicate all functions of the old system. A reasonable middle ground is to print
colored cards with all the key functions from the operations manual and a blank for “write your
function here” and distribute the cards to the users. When a telephony user performs a specific
function, they write their phone extension number or other identification and check off the
function performed. This would allow an audit of used phone functions to be performed. An
alternative to this would be if your PBX, phone switch, or phone service had a reporting function
that identified, at minimum, which features were programmed and, at best, how often the
function was used and by which users. After the audit, it might not be possible to guarantee that
the new VoP system will perform all the functions of the traditional system, but it will be
possible, in advance of migration, to identify potential problem areas and workarounds.

Pre-Deployment Testing and Implementation
The objectives of pre-deployment testing and implementation are to assure the outcomes from
the planning and assessment phase and to determine the impact of adding the new applications to
the existing network—and then to rollout the new system into the organization, assuring that the
systems are operating in compliance with the SLAs. The use of sophisticated, specialized
modeling and planning tools in the first two phases will allow you to assure the maximum
benefits in the shortest amount of time with the least possible disruption to your operations.

Ongoing Operations and Optimization
Onging operations and optimization are two phases with opposite objectives. It is the objective of
ongoing operations to assure consistent functionality across the network, regardless of where an
individual is in the organization geographically. Ongoing operations is constantly measuring the
network’s performance against established benchmarks to assure unwavering compliance and
absolute consistency of network services with SLAs. Optimization, in contrast, has as its primary
objective improving operational aspects of the network services and adjusting the operational
baselines to reflect new needs and requirements. Examples might be to improve voice quality and
update SLA levels, which are operational improvements, or to decommission PBXs or IP
gateways, which will provide a cost benefit.
Although ongoing operations is about consistency, the optimization phase is about a regular,
constant, ongoing program of positive change in network operations. Optimization takes the
measurement and SLA compliance statistics gathered during ongoing operations and seeks ways
to improve those statistics with specific target outcomes. Desirable objectives will include
financial or operational goals. The process will involve conceptualizing improvements and
validating assumptions using modeling or simulation tools and then testing them in a laboratory
or isolated network segment before implementing the changes in the actual network. After
network performance has been assessed and any needed fine tuning done to the changes, the
network operational benchmarks are adjusted and new optimization goals are established.




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Managing the IPT Life Cycle
Now that you have a better understanding of what the IPT life cycle encompasses, you must now
develop a better sense of how to manage the life cycle. This guide will provide much more
insight into the management process in later chapters but here will take a look at the tools that
must be in your toolkit in order to meet the challenge.

Enterprise-Accessible Tools
If you have opted for the “do-it-yourself” approach, you will have access to sophisticated tools
embedded in the routers, switches, gateways, end systems, and other components of your
networks. Even if you haven’t, there are still a rich set of tools that can be used to provide “the
big picture” as well as pinpoint areas that require a more granular focus.

MSP-Provided Customer Network Management
Managed service providers (MSPs) allow end-user organizations to view thin slices of their
network: the slices that are important to the transmission of the customer’s information, masking
information about other customers and providing a true VPN view. This capability is called
Customer Network Management (CNM)—allowing the customer to see their part of the shared
virtual network—and has been around in some shape for well over two decades. Early CNM
systems were periodic summary snapshots showing not-so-very-near-real-time (NSVNRT)
statistics that bore little resemblance to the real world or the SLAs, while more current systems,
especially those based on MPLS VPNs, can provide darn-near-real-time (DNRT) views down to
a connection level and do, in many cases, allow customers to adjust certain characteristics, such
as CoS or available bandwidth, in real time. Although not ideal, CNM is a valuable tool for end-
user organizations that must rely on third parties to provide and manage their service.

Manufacturer-Provided Systems
Manufactures provide the most accurate, most granular, and seemingly most desirable systems
embedded within their hardware. The tight coupling of hardware and operating software
provides the finest possibly granularity of analysis but, in fact, tends to be limiting and
confounding unless the entire network, from the physical layer transmission systems to the
application servers themselves, are provided by a single manufacturer and, of course, that is
rarely the case. The proprietary, manufacturer provided, tools do have their place, when
optimizing a network, for instance, or at higher levels of escalation for system-specific trouble
shooting, but lack the broader, more standardized and less platform-specific capabilities of
systems provided by third parties. It is also true that third-party systems lack any vendor bias and
can be counted on to be more impartial when a system is comprised of components from
multiple manufacturers or as a good “second opinion” in those rare cases that a system is
constructed completely from the components of a single manufacturer.




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Third-Party Tools
Third-party tools for modeling, monitoring, measuring, and managing are invaluable in that they
provide broad, cross-platform visibility into aspects of network operation that far exceed what
humans alone can do without those tools. The consistent and proper application of these tools at
various stages in the life cycle of the multi-media network spells the difference between success
and failure. The selection of these tools is the subject of the third-party tools selection report card
described in the next chapter and their application to the life cycle of the network are described
in further detail in subsequent chapters.

Importance of Modeling, Monitoring, Measuring, and Managing
As important as the right tools to dimension and manage emerging multi-media networks are,
they seem to go against conventional wisdom about how a multi-media IP network is
implemented and managed. Time after time, the biggest and most visible example of IP networks
today, the Internet, has been described as “plumbing,” or a common conduit for any type of
information, and the term plug-and-play has been used over and over to the point that it has
become ingrained in the subconscious, and almost into the genes, of ‘Net users and managers
who must make budgetary decisions. “Free Voice” is another obstacle that must be overcome in
order to be granted the funds needed to operate a network that is, consistent with another
metaphor, “the nervous system of an organization.”
What management must understand, and subsequently follow with appropriate funding, is that
the multi-media IP network, be it Internet, intranet, or extranet, requires special tools to build and
maintain its health. Experience has shown that an initial cost of 5 to 7 percent for procurement
and training, with an ongoing 10 to 12 percent of operating budget dedicated to staffing, ongoing
training, support, and network optimization will pay substantial dividends in network reliability
and cost reductions. Proper funding of personnel and systems, will ensure that organizations reap
all the rewards of their move to a new voice platform.
Put in its proper perspective, the shift from traditional telephony to its packet-based replacement
is like the differences between a 1967 Chevy and a 2007 Lexus. Both have four tires and a
steering wheel and are designed to get you from one place to another. But, what is “under the
hood” varies dramatically. Your ’67 Chevy was so simple that Floyd down at the service station
could pretty much provide any service you needed. In fact, the term “keep it humming,” relative
to automobiles, came from the fact that much of the diagnostic process involved listening to how
the car sounds. The VoP system is much more like your 2007 Lexus. It requires specialized
service at a Lexus dealer or service point that is capable of utilizing the diagnostic interface, and
Floyd at the service station wouldn’t have a hope of understanding the sophisticated,
computerized guts of this vehicle. If you have an accident, a signal is sent to a customer care
center that calls you and will dispatch 9-1-1, an ambulance, and service vehicle within seconds,
even if you are incapacitated. And what about locking your keys in the car? Well, the car can be
unlocked from the satellite, as well. As much as the Lexus may resemble the Chevy in form and
function, it is a fundamentally a different creature with different capabilities, different
management requirements, and different maintenance needs. So, too, is the case with VoP and
traditional telephony.




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Summary
This chapter has begun the practical job of looking at the key elements that further set the stage
established in Chapter 1. It has also begun looking at the key phases of the life cycle of the
packet-based telephony system and how to harness your knowledge of the natural rhythms of
that system. The next chapter will continue to dive deeper into the life cycle.
As promised, the tone and intent has shifted from one of establishing a baseline understanding of
some broad, abstract concepts, to providing specific, actionable items that you can perform for
your own situation. The next chapter will continue this approach.




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Chapter 3: Planning and Assessment
At a professional car race, the cars are powerful, fast, and full of muscle, zipping around the
track at amazing speeds; the drivers defy death and beat physics at every turn. What only the true
racing fanatic realizes, though, is that there is more to this feat than the power of the car and skill
of the driver. Considerations such as the preparation of the track, the track surface, the inflation
of the tires, the bank of the turns, the surface temperature of the track, minor variations in the
fuel mix, and other seemingly arcane considerations contribute in a very real way to the outcome
of the race.
This chapter talks as much about the track—the high-speed broadband surface over which
telephony and other multimedia content races—as it does about the vehicles—traditional and
multimedia phone calls. This chapter will also delve into some of the seemingly arcane aspects
of both, making good on the promise in the first chapter to educate phone people about IP
networks and IP people about telephony systems.
The first two chapters introduced some of the principles driving the change to VoIP and IPT and
noted that this guide will challenge the market “buzz” that has been generated by emerging and
evolving packet telephony technologies. This chapter will continue exploring both the pros and
cons of this technological shift, which should be more of an evolution than a revolution.
Planning and assessment must begin with a careful identification of the business drivers for
change, which will, in turn, provide objectives that will guide the project and allow the
measurement of the outcome. This evaluation must include thoughtful analysis of the current
methods of voice communications and both the voluntary and involuntary drivers. Voluntary
drivers include the desire to reduce cost and increase productivity as well as advances in business
applications that require the integration of new technologies. Involuntary drivers can be factors
such as manufacturer discontinuation of existing systems and an enterprise’s lack of desire to
continue investing in hardware that is rapidly becoming obsolete.
Planning and assessment also includes setting proper expectations with both users and
management. Expectation setting includes defining proper design and ongoing operational
criteria, such as technical requirements and acceptable operational thresholds. Important metrics
for all telephony applications are identified, whether for simple two-party telephone calls, voice
and/or multimedia conferencing, voicemail and presence applications. Methods for translating
those metrics into a measurable and actionable Service Level Agreement (SLA) for internal or
external service providers are also provided—whether those providers are carriers and service
providers who supply the track or service providers, systems integrators, and inside entities who
provide the vehicles that run on the track.
This chapter will also include a unique, comprehensive capabilities inventory that is designed to
ensure that every capability used in the current/old system is at least considered for inclusion in
the new system.




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Business Drivers for Change
When companies explore Internet Protocol Telephony (IPT) solutions, the focus must be on key
business areas because it is business applications and needs that will drive change, and the
meeting of business needs that will define the project and make the project successful in a
business context. As different as it may often seem an IPT project is no different than any other
project that impacts the way in which every member of an organization does their day-to-day
job. In fact, the IPT project touches the lives of every member of the organization and the way in
which they communicate with each other and with those outside the organization.
When evaluating the full range of business needs it is vital to the success of an IPT
implementation that management and user requirements are assessed and proper expectations set.
It is also important to fully understand all the business drivers behind the shift to IPT. If cost
reduction is the primary driver, there will be different factors to consider than if the primary
driver is deploying new, converged applications. Understanding the key drivers for the project
helps maintain focus on those factors that ensure success. Communicating the objectives clearly
with everyone involved, including suppliers and service providers, helps maintain a clear view of
the expected results and allows all parties to weigh in on the viability of the objectives.
When the migration from traditional telephony to VoIP and IPT began in 1995, businesses and
industry pundits alike projected incredible cost savings. These savings were primarily seen as a
natural outgrowth of consolidation. The consolidation to a single physical network and single
cable plant also led companies to envision a consolidation of staff, reducing the support
requirements for a single converged network. Much of the punditry surrounding convergence
neglected the inherent complexity of telephony itself—for years the market had been lulled into a
false sense of simplicity by the fact that anyone could lift the receiver on any phone, press the
correct series of digits, and—voila—a person anywhere in the world would lift the receiver and
answer!
Over time actual experience with VoIP has made a mockery of most claims of deep savings vs.
traditional telephony, unless, of course, users are willing to abandon most features of traditional
telephony, which is hardly a fair comparison, and use the nearly free PC-based systems.
Although the savings of VoIP circa 2007 are substantial compared with the cost of traditional
telephony circa 1995, VoIP today often demands a premium as older, depreciated gear with a
service life in the range of two decades is replaced with an unknown or 18-month to 5-year
expected service life. And one must also consider that purveyors of traditional telephony have
reduced the prices for use of their reliable, high-quality traditional systems. This leaves issues
such as manufacturer discontinuation, including both the discontinuation of lines of traditional
equipment as well as the discontinuation of the manufacturers themselves, and the desire for
new, integrated, multimedia applications in the forefront of reasons to purchase the new, often
more expensive, VoIP and IPT systems.




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As networks converge, new integrated services can provide unique, new business processes. In a
fully integrated network, the tighter relationships between data systems, telephony services and
video lead to new opportunities. Take customer service as an example. Customer data, collected
over both telephone and computer systems, can enable new methods in providing customer
service, possibly at a lower cost, but more importantly in a manner that directly increases
customer satisfaction, and indirectly increases revenues, enhances an organization’s reputation,
and raises the bar for customer service within the industry; thus, putting pressure on competitors
to meet the new challenges or lose market share.
Call center functionality, for instance, can easily be dispersed geographically, and not just
through low-cost VoIP-driven off-shore call centers but also through distributed, call-center-free
call takers working part time at home on a demand-driven basis at a lower total cost but a higher
rate per hour. It’s the technology that makes it easy to integrate teleworkers at remote locations,
at lower costs and providing higher levels of service while better meeting the personal needs of
the workers. An integrated, Web-based customer service application can handle both telephone
calls and provide Web services such as interactive online and TTY for the deaf. Those users who
would prefer to communicate via their browsers can explore a business Web site and click a
button for customer support that places a VoIP phone call to a company representative who is
working at home or in a centralized call center. With IPT in a fully integrated environment,
computer telephony integration (CTI) tools can provide the customer support representative with
comprehensive information to assist the customer. This information might include not just
customer contact and account information but also a screenshot Web page that the customer last
visited and a complete history of what this customer has been looking at on the Web site.
For example, consider the scenario that Figure 3.1 shows—an insurance service center. What
sort of reaction would this company get if, when the customer clicked “help,” a service rep were
to come online via IPT linked to a Customer Relationship Management (CRM) front end and
speak directly to the customer? “Good morning, my name is Mary. It looks like you’re trying to
find information about our new policy options for your automobile coverage. I’m sorry the Web
information wasn’t more helpful. How can I help you?”




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  Web Server

                     IPT Gateway




                                            PSTN


            CRM App
             Server
                                          Internet




Figure 3.1: Customer service and CRM in the integrated IPT environment.

The call center can just as easily support Internet users as PSTN callers. Computer telephony
integration at its finest and most developed can provide a powerful competitive edge to customer
service organizations. This evolution in the network and change to the corporate culture require
both time and focused effort. They evolve continually as new behaviors are learned. Although no
company can jump from legacy telephone systems to a fully integrated voice and data network
overnight, IPT might provide a point of entry for your company to begin moving into a
competitive converged network.

Lower Costs
As mentioned earlier, lowering cost was seen as an early driver for migration to VoIP. Cost
reduction was seen as coming from multiple fronts. First, the consolidation in infrastructure to a
singe network was viewed as a source of cost savings. The problem with this logic is that most
businesses standardized on a single, high-grade Category 5 or Category 6 twisted pair wiring
scheme years ago that can handle the needs of all forms of multimedia information. The old
dual-mode cabling of twisted pair for telephones and other types of cable for data workstations
has been replaced with a standard, unified wiring scheme. Bottom line: there is not significant
cost savings today in the physical plant.
A second area of cost savings was the reduction or elimination of telephone costs through the use
of the IP network to replace the old phone network. Organizations that started with this idea
often find that their existing IP networks are usually not large enough or robust enough to handle
the new load of telephony and multimedia demand without being enlarged and made more
redundant, and that the cost for VoIP-based outside phone services is about on par with today’s
cost of traditional services.



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A third area where substantial cost savings were envisioned was personnel costs. The general
thinking was that because voice is just another IP application, it could be put on the IP network
and managed by the data team so that all the traditional telephony people could be terminated,
thereby creating a substantial ongoing cost savings for the organization. This logic is faulty, as
organizations that do a wholesale purge of their old telephony departments go without key
services. There is just no way to replace the knowledge and expertise.
What happens is that organizations often use the migration to a new IPT system as an
opportunity to sort out their data and telephony personnel, keeping the best and brightest from
each department. What emerges, however, are single super-techs with multimedia capabilities
that are well-suited to work in any company and cost more in training, salary, and benefits. It is a
whole new era and a proper program must be developed to ensure that the needed skills are
available, and this must also include the consideration of service providers and system
integrators who will have the hiring headaches and share the cost and availability of their stables
of skilled multimedia workers over all clients.

The Myth of Bandwidth Savings
There was a widespread myth associated with early VoIP services that all you have to do is add
VoIP to your network and it will work just fine. Vendors said it will fit in between the unused
spaces in your data network. Don’t believe it! Voice service is mission critical to the operation of
any business. Anything you add to your network—whether it’s an application, new software, or
new devices—will impact performance. If your equipment vendor tells you to add their product
and your “voice will ride free,” be very skeptical. You will get what you pay for, and it’s crucial
that you protect all your business network services, whether they are voice or data.
One early driver behind packet telephony migration was to move voice traffic off the more
expensive PSTN onto a cheaper network, like the Internet. This was purely driven as a cost-
reduction measure. Although this method presents an intriguing argument, it also presents a true
“apples to oranges” comparison. The PSTN and Internet are regulated, managed, and priced
under completely different methods. The PSTN is a highly regulated environment, while the
Internet still remains, for the most part, an unregulated data service.
One approach to providing quality to a network has been to exponentially increase the carrying
capacity or bandwidth of the network. This has historically been referred to as gigabandwidth.
With advances in LAN switching technology, Gigabit Ethernet is now fairly common in the
backbones of corporate networks. 100Mbps to the desktop is commonplace. Many pundits and
technologists have argued that with the exponential increases in optical networking, we have
achieved a glut of bandwidth. Some have even proclaimed that “bandwidth is free,” although we
all seem to be paying a higher price that we would like to pay. Certainly any IT manager who
pays for WAN circuits will tell you bandwidth is not free.
Bandwidth is a finite resource in your network. CPU processing cycles are finite too. All your
business network services are additive, each consuming available resources.
An opposing argument is that providing a user with gigabandwidth is merely a temporary patch
or workaround. The Internet has done more to encourage consumption of data than the early
designers ever imagined. End users always find creative uses of technology that the inventors or
designers never dreamed of. Today, the idea of a gigabit data stream to the desktop seems to
eliminate any need for QoS mechanisms.


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The question becomes whether some traffic is more critical than other traffic. For many
businesses, telephone traffic is the most mission-critical application there is. A call center that
takes catalog orders or makes airline reservations cannot function without telephony. Assigning a
lower priority to email and Web traffic is a sound business decision for such a company. In order
to do that, some prioritization scheme is necessary, whether it’s one reviewed here or some
entirely new approach.
The best solution to QoS issues involves the well-being of the entire network. Processing power
must improve. Bandwidth must increase. Traffic engineering techniques must evolve.
Prioritization for mission-critical traffic must be implemented. There is not one single “silver
bullet” solution that can placate all the demands placed on IP and the Internet today, because the
problem is not one single problem. Only a complex solution will effectively address the myriad
performance concerns in today’s networks.

Access Charge/Local
Local access charges have been challenging for many businesses. Large enterprises might have a
distributed PBX architecture with a comprehensive “least cost routing” methodology
implemented across the corporation. These local access lines, typically T1, E1, or J1 lines, were
at one point, an incredible cost burden. The cost was offset by routing calls inside the private
voice network of a large company and dropping them off to the local telephone network as close
to the destination as possible. In essence, companies replicated the routing of the PSTN within
their own private telephone network, providing the lowest cost access based on their own
infrastructure.
Although this approach still works in the packet telephony world, the use of the ubiquitous
Internet and the lower cost of high-speed packet interfaces often require a rethinking of how calls
are routed. Many companies, however, will develop one strategy to get them through the
migration period and another strategy for the post-migration period. Any multi-national
organization must also consider the legality of packet telephony, or at least a bypass of the
national Postal, Telephone and Telegraph (PTT) authority in any country in which they work—
as well as the legality of encryption if that technique will be used to keep the packet voice
secure.

Long Distance
Long distance access was reduced for many companies through the local access trunks provided
at each remote office. Companies used creativity in trunking PBXs and key systems in smaller
offices to make the absolute best use of expensive carrier resources at the cheapest possible rate.
Many large enterprise companies built telephone networks that rivaled those of small telephone
companies.
PBXs were linked or meshed together using point-to-point circuits and proprietary protocols,
establishing a telephony network that allowed the customer to create routing priorities. Although
this setup has helped reduce the cost of long distance calling in many cases, it has added
complexity to the customer voice network and put the onus on the customer for making path
determinations about call routing.




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Intra-Company
Intra-company voice traffic has been solved as a problem for many years. Companies have
established a combination of voice trunks and voice channels on the corporate data network that
allow hauling internal telephone calls from coast to coast and around the world. PBX vendors
provided IP interface cards to simplify inter-PBX trunking over IP networks, and in many cases,
a company simply carved voice trunking capacity out of the data network bandwidth to connect
voice systems.
This approach was often presented by vendors as “free VoIP” on the data network, however, as
Chapter 2 explored, this is a fallacy. More accurately, this was sales and marketing hype that
caught the attention of some customers. What companies did find is a financial savings in using
the same carrier network infrastructure to transport both voice and data traffic. Nothing rides
free, but the savings of a single infrastructure and single billing system has proven quite
significant.

Extra Company
Telephone calls to the outside world are the lifeblood of many businesses. We call customers,
vendors, and business partners every day. As companies scrutinize expenses for inbound and
outbound telephone calls, close financial management motivates business managers to find
creative new ways to reduces costs.

Support Costs
In addition to the engineering resources addressed earlier, many organizations look to cost
savings in other support areas. Just as in the network engineering area, don’t divest yourself of
other critical support resources such as the service desk and the technicians who handle
moves/adds/changes. Instead, team your voice and data specialists together as a new unified
communications team, extracting the valuable knowledge from every resource at your disposal.
Don’t let the knowledge of your services, your network, and your requirements slip out the door
through some misguided effort to reduce costs. The cost reductions must be won over time. They
aren’t necessarily real for every business and they don’t come immediately.

Help Desk/Support
Help desk support for telephones is generally simple. Everyone knows how to use a telephone.
Many companies conducted user training at the time a new PBX was implemented and handed
out basic feature guides from their equipment vendor that explained how to use voicemail
systems and the various features of the system. In the integrated environment of VoIP/IPT, these
features may be delivered to telephone sets that can be treated like traditional telephones. More
often, integrated system introduce some element of data workstation feature management. For
example, users can often manage their own telephone set button configuration and speed dial
lists from a Web interface via their workstations.
It’s critical to remember that as the services converge onto a single infrastructure, your corporate
Help desk will be getting calls about new services and question about things they too will be
unfamiliar with. It’s vital that the Help desk and support teams be involved in the migration
process and participate in early training efforts it order to effectively support ongoing daily
business operations.




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Moves/Adds/Changes
When implementing an integrated services project, moves/adds/changes must be taken into
consideration. This is hugely important in a traditional telephone environment but even in an IP
telephony environment there are still many potential problems. Employees often move his or her
workstation, which retrieves an address from the network using Dynamic Host Configuration
Protocol (DHCP). Telephones, on the other hand, often had to be moved by the telephony
administrator, reprogramming the PBX to deliver extension number and features to a new
physical wire in the new work area. With integrated services, this process may be simplified but
it cannot be overlooked. For instance, many systems allow the users to make all needed changes
but often the users lack the technical sophistication to make the needed changes and even the
most savvy users are rarely given access to firewall and security systems that often stand in the
way of completely transparent moves adds and changes.
In many cases, the $12 per hour telephone technician—whose primary skills involved being able
to read a floor plan, locate wire pairs, and use a punch-down tool—is being replaced with an
$80-to-$120K per year IP engineer with knowledge of fire walls, IP router configuration, subnet
address masks, and an entirely new set of skills possessed by a limited number of workers.

Remote/Nomadic/Mobile Workers
Integrating voice services with VoIP solutions provides an excellent tool for augmenting remote
worker resources. Your call centers can now be staffed with people working from anywhere. For
some call centers, this provides an opportunity to hire new, less costly staff that would have
previously been unreachable. For example, stay at home mothers can now provide telephone
support with full access to corporate resources. The widespread availability of broadband
Internet, coupled with VPN technologies, provides a data linkage virtually anywhere. VoIP and
IPT solutions do the same thing for voice services.

Enabling New Strategic Applications
Strategic new business applications are many and varied. Your company might have industry-
specific applications that don’t apply to other industries. Inventory and supply chain
management, for example, is crucial in the manufacturing sector but plays a much smaller role in
the financial services sector.
Business drivers, market shifts, and process re-engineering all play a role in determining what
strategic new applications a business might deploy. CRM systems are permeating many new
businesses. Enterprise resource planning (ERP) solutions are another factor for many
organizations.
It doesn’t matter whether you’re deploying Oracle, SAP, or a Web application; the odds are high
that any strategic change in your data services applications will ripple into telephony services in
some way. In many cases, integrating telephony services will be a key business change enabler
in and of itself.




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Worker Productivity
Worker productivity will go up. Worker productivity will go down. There will be iterative cycles
of improvement. Although your staff may be provided with new tools and new options that can
enable them to work more effectively, don’t overlook the learning curve. Most people learned
basic telephone usage at a young age and have spent most of their lives dealing with telephones.
Shifting that process to a softphone running on a PC, connected to a wireless network, integrated
with a CRM application presents a new way of working.
Integrating unified communications into the corporate environment introduces the potential for a
large cultural change. It’s absolutely vital that workers be involved in understanding the changes
and how their work will be impacted each step along the way. The more engaged workers are,
they more quickly they will find ways to use the new environment to increase productivity. Most
workers are driven to get things done, but they also need to understand what the tools they have
to work with can do—and how those tools work.
Road warriors can use these technologies to enhance accessibility. Traveling executives can use
their office number remotely from a hotel room. Engineers can still work while in training or
away on a project. Integrated voice services provide the opportunity for not only enabling this
work flow but doing it all transparently. A customer calling a salesperson doesn’t need to know
that the sales rep is halfway across the country in a hotel. The office telephone number can
become truly portable. And the coming advancements in fixed mobile convergence and IP multi-
media subsystems (IMS) will couple service more tightly to the remote worker than was ever
before possible.
When evaluating systems, it’s easy to fall into the “feature/function trap.” Vendor selections are
often made based on features and functions, with cost comparisons driving final decisions. For
your workers, it isn’t just about the features that exist, but how they are used. A worker who is
used to having a 32-button telephone set that shows eight coworkers’ phone lines with status
indicators will have to change work habits to effectively use a softphone. Engage workers
throughout the entire migration process from planning to production to maximize their
productivity on day 1, day 10, and day 100 after deployment.

Obsolescence of Older System
Older systems become obsolete. Traditional Time Division Multiplexed (TDM) PBX systems
once had a practical lifetime of 10 years. Many outlived that, but many were also obsoleted early
because of technological advances.
Data networks have tended to be re-architected in a shorter life cycle. Data networks are often
redesigned every 3 to 5 years. Switching and routing technologies have advanced quickly,
making complete redesign of data networks cost effective in a shorter life-span than older PBXs.
Consider not just the obsolescence of a legacy telephone system but also the data network life
cycle when developing migration strategies to move to VoIP or IPT solutions.




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No Longer Want to Invest in Old Technology
Obsolescence is one driver, but for many companies, especially market leaders, there are
compelling business reasons to migrate to newer technology. It’s not uncommon for an
enterprise to move to new technology, not when older systems are at end of life, but when the
end is visible on the horizon.

Manufacturer Discontinuation of Product and/or Support
Manufacturers discontinue support for older products every day. In some cases, the parts become
scarce, making hardware repairs a challenge. In many cases, manufacturers will consciously
choose to end the life of a product in an attempt to encourage customers to move to newer
systems. In the history of business telephony, this was often called a “forklift upgrade.” The
analogy being that a forklift was needed to remove the old equipment to make room for a
complete new system.
If your equipment manufacturer has put your company in a “forklift upgrade” situation, it may be
important to consider how firm your relationship is with the manufacturer. Vendors play a
dangerous game with this strategy, as customers are presented a perfect opportunity to take a
system design back out to competition on the open market. This technique has backfired on
many vendors, costing them key customers.

Increased Support and Maintenance Costs
As systems age, they become more expensive to support and maintain. Parts may be less
available for hardware replacement. Technical skills to support older systems will naturally
decline as these systems phase out of useful life. If you continue to use a 10-year-old system
when much of the market has moved to other technology, the support costs to maintain an
effective staff can easily rise. What it costs to support and maintain a telephony system today
may be 30 percent more than it was when that system was a widely deployed standard
implementation.
Another cost factor might simply be technology related. Newer hardware and software often
introduce new patching and upgrade methodologies that are more cost effective than techniques
used in the past. That can drive down support/maintenance costs for new systems in comparison.
Make sure you review all the data and factor in the ongoing overhead in dollars, time, and skill
set to support each of the options you consider.




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Lack of Expansion Capability of Old System
Hardware and software solutions have practical limitations. Telephony systems are introduced to
the market at a price point to ensure widespread adoption. As time passes, technology advances,
and businesses advance too. A key system designed to support a 10-person office 5 years ago
may well be stretched to capacity today, as that office may have 30 people. Hardware could be
utilized to the max already. Expansion capability isn’t just hardware. Older hardware solutions
might not be able to support newer features your business requires. Expansion in software can
only be supported to the capability of the hardware.
In your evaluation process, you can’t just look at the requirements of today. The investment of
technology a business makes is a strategic investment. It has to support the operational needs to
the moment, today, as well as growth and evolution to support the strategic needs of the business
for some foreseeable future. For some companies, that might be 7 to 8 years, and for others it
might only be 3 to 4 years. Incorporate your strategic direction plans for the business with your
strategic plans for technology. The technology won’t do you any good if you don’t build it to
support your business.


Setting Proper Expectations
The bottom line for the success of any project is setting proper expectations. The executive
management team may have very clearly defined expectations based on ROI and the size of the
investment. Users may have an entirely different set of expectations based on workflows and
operations processes. If you build your entire ROI case around the technology but implement an
integrated service in a way that increases the labor effort of your staff, the overall result may be
counterproductive. Everyone needs to know what to expect and what is expected of them. The
only way to really be successful in any large project is to clearly define and communicate
expectations. Like the network, the telephony system, and your business as a whole, the life
cycle of an implementation project is a repeating pattern of setting and reviewing expectations.

Management Expectations
The executive management team focuses on key strategic factors. Capital expenditures (CAPEX)
address the investment, and are often tied to ROI. Operational expenditures (OPEX) take into
account the cost of operations and support; they’re tied more to the total cost of ownership
(TCO). Technology can often drive process re-engineering that changes workflows. Technology
is often molded to fit business needs, but just as often, business processes are modified to
leverage some feature of the chosen technology. You must also consider the return on effort and
factor the work effort involved into the total cost of implementation and ownership.
Implementing VoIP or IPT comes with a broad general set of management expectations that vary
widely, yet are all accurate. One view is that IPT will save money. This view generally assumes
a modest CAPEX spread across all users of the service. Using cost recovery as a lens through
which to view CAPEX helps balance the total CAPEX against the true investment and determine
how quickly a solution might pay for itself. The money saving view often sees that OPEX can be
reduced by saving on recurring cost. For many companies, this is based on a shift from a
provider-managed service such as Centrex to a self-managed-IPT service.



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Another view is that VoIP may cost more. One migration can lead to another. Some VoIP
deployments have been plagued with network upgrade upon upgrade, but this is often because a
comprehensive readiness assessment wasn’t completed on the front end. Many business
managers find that in their environment, CAPEX is as high as it would be to continue using
traditional TDM-based PBXs. OPEX can be variable and particularly difficult to measure when
you broaden the scope of the IT staff’s role. The impact of requiring a new key competency or
technical skill in a networking staff can inflate OPEX for the first year or two as the staff climbs
up the learning curve. There still may be tremendous value in integrating strategic business or
multi-media applications.
A third view is that it doesn’t matter. It could be a cost-neutral decision either way. This is why
some companies take cost reduction off the table in their strategic planning of technology. It’s
important to know your core competency and focus on that. The business you’re in is the
business you have to support. The technology choice needs to be driven to support the strategic
direction of the business. Don’t let the technology drive your business. Leverage the technology
to strengthen your business.
Integrating voice and data is a complex project, filled with potential pitfalls. Comprehensive
planning, network readiness assessment, and project oversight is absolutely vital to the success
of an IPT implementation.
Every step should be documented. Every test plan should be thoroughly reviewed by multiple
team members. As you begin the implementation, you’ll face a series of potentially service
impacting events. Circuits might be upgraded. Routers and switches might be replaced, and
firewalls might be reconfigured. At every step along the way, thoroughly test the new
environment. And for each of these steps, document a fallback plan to ensure business operations
aren’t disrupted if unexpected problems arise. Build a comprehensive plan for success with full
management buy-in.

End-User Expectations
It’s essential that end users be included in the process from planning to implementation. They are
invested in the success of unifying communications services. They need to know that although
things will be different, different isn’t bad. During needs analysis, you’ll need to understand their
requirements. As the project progresses, setting expectations about how things work, voice call
quality, and new features are important to them.
Collaborating with users makes them part of the process and part of the project and gives them
an investment in the success. Treat them as advisors and partners whose business needs you’re
supporting. Training is often available through the vendors whose solutions you’ll buy. Take full
advantage of vendor training opportunities. Train the trainer sessions with key users is an
approach that’s proven very effective.
Even the final cutover implementation needs to actively engage end users. They will be your
primary system testers. Use them to help develop test plans, and they’ll feel their needs are being
addressed. Be proactive and responsive. Another very effective technique is to activate a special
cutover Help desk for several days before and after the “go live” date for your new system. This
approach builds confidence and makes for satisfied users. When the users are satisfied that
they’re getting what they need and understand how to use the system, your project is well on its
way to being successful.


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Inventory of Existing Capabilities
As you begin the assessment part of the project, you must perform an audit of the capabilities of
your current telephony system. Such an audit provides key insight into potential problem areas.
What you know about your operating environment provides the foundation on which you build
your assumptions. The more you know, the more accurate your assumptions will be. There are
some specific areas to audit that help increase your chances of success:
   •   Network infrastructure
       • Identify circuits used for wide area networking and Internet connectivity
       • Inventory the network hardware—routers, switches, cabling.
       • Document software such as router OS versions; this is a good time to get your routed
          network all on a common foundation
   •   Current business data applications
       • What servers are in place today?
       • What is changing or evolving in parallel?
       • Identify any linkages between your existing business applications and voice services
   •   Security
       • Inventory/audit your security environment
       • Ensure that firewalls and other security systems will allow VoIP traffic
       • Determine capabilities for moves/adds/changes allowed by your security system
       • Assess legal issues regarding security and encryption for the areas in which you will
          be operating
   •   Management tools
       • What network management tools are being used in your data center?
       • What server/application management tools are you using in your data center?
   •   Current voice services and equipment
       • List all PBXs and key systems in your organization. For each, include its “End Of
          Life” date and/or date at which it will be fully depreciated
       • Document any inbound and outbound call center environments
       • Document expected telephony performance
       • Inventory all existing telephony features, determine their importance, and decide
          whether they will be available in the new telephony environment; this audit is
          covered in the next section
       • Measure your current busy hour calling patterns
       • Intra-office
       • Intra-company (between offices)
       • Offnet for each office (incoming and outgoing calls [local, long distance,
          international]—not to other offices)
       • Fax machines
       • PSTN channels/lines for each office


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Existing Telephony Performance Expectations Audit
Ideally, you should quantify TDM service quality aspects that you want to carry over into IPT,
either to ensure you build a system that meets user expectations or so that you have defined
criteria to which to hold your systems integrator accountable.
Quantify and define requirements around:
   •   What is a reasonable length of time to wait for dial tone? At what point do you start
       jiggling the hookflash or start punching in the number anyway?
   •   Once you’ve finished dialing a number, how long do you wait for a ringback tone before
       hanging up and trying again?
   •   For every 100 calls you make where you correctly dial the number, how many of the 100
       do you expect to succeed?
Based on responses to these questions, define a set of SLA requirements against which the IP
telephony environment will be measured, to evaluate its success and ensure that it meets
customer expectations. These SLAs should include:
   •   Availability expectations—This is usually expressed in uptime percentage of the
       telephony service from an end-user perspective.
   •   Throughput—Stated as successful call completions as a percentage of call attempts. You
       might want to provide different throughput requirements for different call types. For
       example, you might require a 99.9% successful call completion rate for all incoming calls
       to a call center, while requiring that only 97% of all internal calls complete successfully.
   •   Call setup and teardown boundaries; for example:
       o 95% of all call attempts get dial-tone within 2 seconds while the remaining 5% get
         dial-tone within 4 seconds
       o 90% of all call attempts get ringback tone within 6 seconds while the remaining 10%
         get ringback tone or congestion tone within 9 seconds
   •   Voice quality expectations, usually expressed in Mean Opinion Score (MOS) bands of
       0.2 range—For example:
       o 85% of calls must have a MOS score of 4.2 or higher
       o 95% of calls must have a MOS score of 4.0 or higher
       o 4% of calls must have a MOS score of 3.8 or higher
       You might also want to consider defining different voice quality requirements for inter-
       office (that is, calls that cross the WAN) than you do for calls within a local office.




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Existing Telephony Budgeting, Relationships and Tools Audit
Also worth considering as part of planning your IPT rollout are the potential changes in
responsibility for paying phone bills and impacts on business processes:
    •   Prior to IPT, each region/office may have had their own interconnect and telephony
        budget.
    •   After IPT rollout, who will pay the telephone bills? Same as before? Or maybe the NOC
        team charges each department for their telephony service and holds the entire budget for
        telephone bills?
Consider existing service provider relationships—telephony, network, and desktop management.
Also consider existing management tools—event manager, network management, and
server/application management tools. Which (if any) of these existing relationships and tools
should be integrated into your new converged environment?

Existing Capabilities Audit
A logical approach to auditing the capabilities of the existing system would seem to be to go to
the documentation for the existing system, check to see which features have been paid for and/or
implemented, and use that as a guideline for the new system. This approach only gets you part of
the way there. What is really needed, beyond knowing what features are available, is knowing
what features are actually used. This second requirement needs a great deal more research but
can help you ensure that your new IPT system will provide every feature your organization
needs, while avoiding the unnecessary effort, complexity, and cost of acquiring an IPT system
that provides features that your organization simply does not need.
There are two general approaches to determining which phone features are used and which need
to be replicated in any new system. One is network-centric and one is user-centric. The network-
centric approach involves gathering network traffic, such as signaling traffic, to ascertain which
features are being used. Although this approach makes perfect sense to an engineer, it often
neglects features that are local to specific phone sets. Another network-centric approach is to
check the PBX or Centrex configuration information for the features that are available for each
line and to use those as a basis for the new system design. Although this option comes closer to
understanding the user’s needs, it is still, at best, an indirect method. The best way, actually, is to
ask the users! The users will be able to explain not only the features that they use but also the
relative value of each.
One must also consider that certain phone features that may be unimportant and replaceable in
one industry are critical to others. “Wake-Up Call,” for instance, is a feature absent from most
business telephony systems but one that is critical for the hospitality industry. “Skills Based
Routing,” the ability to route a call based on the skills of a particular individual call taker or class
of call-taker, might not be important in a hospital environment but may be of the utmost
importance in an insurance company. These are some of the reasons to ask the users rather than
make assumptions.
The user-centric approach begins with the network-centric approach, but instead of jumping right
to the inventory and analysis, the list of features is translated into a survey tool to be used by the
phone user to indicate features that they employ. Figure 3.2 shows a list of many of the most
common features.

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Figure 3.2: Traditional telephony features.

Once a comprehensive list of potential features is developed, a suitable user survey approach
must be created. Once again, the obvious approach is not the most suitable. The obvious
approach is to provide a list of all possible features and ask the user, at the very least, to check
off which features they use and, at the very best, to rate the features in terms of their importance
on a 1-to-3 or 1-to-5 scale. The problem with this approach is that the users often don’t
understand the technical name for a feature. The obvious solution is to provide descriptions of
the features, but that still leaves the problem that not all users use all features all the time and that
they might leave off an important feature.




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Another approach is to provide some type of card with the features listed so that when a user
uses a particular feature, they can simply “tick off” the feature. Although this approach works
well and provides a more-or-less accurate audit over a period of time, it generates a lot of cards.
The total number of cards can be reduced by having the user check off the feature the first time
they use it and use hash marks to indicate how many times they use the feature during the study
period. This reduces the number of cards, but analysis is still problematic. A further
simplification can be achieved by choosing representative users for each department, job
function, or operational area. Doing so assumes that each executive secretary will use the same
set of functions and features—not always an accurate assumption—and that each call center
operator will use the same set of features and functions—more likely to be true within a job
category. Entry-level operators will likely use the same subset of functions; supervisors will
likely use the same set, and so on.
Once a cross-section of users is identified, one more simplification step can be achieved.
Through the use of a specially designed survey card, which is drilled at specific points around
the periphery to align with audit questions, it is possible to greatly ease the survey and audit
process and quickly analyze results. The creation and use of cards of the kind shown in Figure
3.3 can greatly enhance and speed information collection and analysis.




Figure 3.3: Sample telephony capabilities audit card.




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It is also not required to print capabilities on a card that will not be used by a specific audit
group. “Wake-Up Call,” for instance, is not required on a call center operator’s list of possible
features. Different colored cards should also be used for each functional area to allow common
and unique requirements to be identified.

Synchronizing Existing and New Capabilities
Once the cards have been marked by having the appropriate holes removed, they can be stacked
and jogged to align the holes; then information can be derived from the card deck regarding
features of importance. This is done by inserting a long stick, such as a small diameter chop stick
or knitting needle, into a hole and then shaking the stack gently until all cards with the same hole
punched out drop from the deck.
This process will result in a list of features that are used in the current system and that therefore
must be supported by the new system. Users who only have requirements that are supported by
the new system may be placed into the first group for migration. A list of those requirements will
also need to be assembled. This process will also result in a list of features that were not known
to be requirements of the new system but which must be included in the RFP. Features not
supported in the initial release of a new system but which will be supported in subsequent
releases can result in users needing those features to be placed in a second group for migration.
This process will also result in a list of features that are never intended to be supported by the
new system. The needs of these users must be somehow synchronized with the capabilities of
any new packet telephony system through retraining, reworking of the need for the feature, or a
rethinking of a move to the new packet telephony system until the features are available. In many
cases, it should be possible to extend the lifeline of an existing system through the use of a
feature or gateway or other mechanism to allow it to continue to provide the needed feature for
the foreseeable future.
                                           Voice over Packet
The broad term VoIP has been accepted by the industry and consumers to refer to a new, inexpensive
way of making telephone calls using the “Internet.” Although “insiders” understand that the IP network is
not always the Internet, the common definition is convenient and easy. What VoIP actually means is that
the voice information is sent using proprietary signaling protocols, such as Cisco’s Skinny or H.323 and
increasingly using Session Initiation Protocol (SIP). The term VoP, or Voice over Packet, is inclusive of
VoIP and is a more general term meaning the coding of voice into binary and transmitting over a broader
range of options, including ATM, Frame Relay, and DSL. By considering a wider range of voice over
packet options, not just VoIP, an organization can often maximize use of their current networks, avoid
dealing with regulatory issues in certain countries, and extend the life of existing assets.




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Network Readiness Assessment
The chapter introduction talked about a race car analogy. At this point in the chapter, we have
talked about the reasons to join the race and how to properly assess that the new race cars being
purchased or built, at the very least, have all the capabilities of the race cars being replaced and,
preferably, are more powerful, faster, and have more features. We now turn our attention to the
race track: the existing IP network. Once the decision has been made to move to VoIP or IPT, an
extremely important part of the planning process is to evaluate not just whether the existing IP
network can handle the predicted call volumes while meeting the exacting real-time performance
criteria associated with telephony, but also what enhancements are required to the IP network to
ensure that it can do so. More than 60% of organizations that have already rolled out IPT did
require some level of IP network upgrade and, in a not insignificant number of cases, the cost of
the network upgrade ended up being equivalent to or greater than the cost of the entire new IPT
system.
Many existing networks today evolved from departmental local area networks (LANs) that have
grown over time. In some cases, regular, methodical redesign work has been performed every
few years. Just like the PSTN undergoes constant traffic re-engineering changes, some IP
networks have been finely tuned and redesigned to support new data applications. For others, the
applications in use today may be very different than the original applications were. It’s important
to fully understand the network performance requirements of VoIP services being added and
how they will impact existing network services.
Many companies begin voice integration projects fully understanding that network upgrades will
be required. Network nodes such as routers, switches, and firewalls may already be running at
high CPU utilization. They might not be able to support VoIP services without upgrade or
replacement. Existing wide area network (WAN) circuits may already be overburdened carrying
the existing data applications. Network bandwidth, robustness to single points of failure, and
processing capacity are key factors in deploying any new service like integrated voice.
One of the most common mistakes, and one that has doomed more than a few IPT projects to
certain failure, is skipping the network readiness assessment and jumping directly to
implementation. There are four primary reasons for skipping the network readiness assessment,
and all appear to be viable reasons until the project completely falls apart leaving the
IT/telephony group with no credibility and the organization with no telephony system. The
typical reasons are:
   •   “A network readiness assessment takes time and money. VoIP/IPT is just another IP
       application and we know it will work. Why waste time and money proving what we
       already know?”
   •   “We hooked up a VoIP server and some phones over the weekend and everything worked
       beautifully! We can now implement with confidence.”
   •   “We have enough bandwidth in our network to transmit the entire World Book
       Encyclopedia twice every second. If our network can do that, we can definitely handle
       VoIP and, besides, our current network utilization is below 30%.”
   •   “We can’t upgrade our old TDM system anyway: we have to go to VoIP. Why do an
       assessment? We’ve got to move to VoIP anyway!”


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The following horror stories, from real organizations who did not do network readiness
assessments, will illustrate the inaccuracy, and risk, associated with not performing a
comprehensive network readiness assessment before moving the first live call to the existing IP
network.

   Horror Story #1
   A large US retail chain went straight to pilot without having completed a network assessment. Their
   call quality was terrible, the call setup delays were frequently more than 5 seconds, and down times
   were frequent and protracted. The pilot was deemed a failure, the roadmap to IP telephony was
   discarded, a lot of money was wasted, and they are still using their traditional TDM system.
   Lesson Learned: Network readiness assessment is a critical part of any move to VoIP/IPT.




   Horror Story #2
   A large government agency went straight to live implementation without either assessing their
   network or executing a pilot. Telephony traffic saturated their network, with the result that their call
   quality was consistently unacceptable and all other distributed applications in their network
   experienced long delays resulting in extreme telephony and data user dissatisfaction and low
   productivity. To resolve the problems, the customer had to back out 75% of their IP phones. They are
   now in the process of doing their network assessment on a live network and will have to try to
   redesign and upgrade their network without impacting their live users. Their original ROI forecast has
   been blown away and their current budget analysis is that their IPT rollout will end up costing them
   more than double their original forecast.
   Lesson Learned: Any change to the existing network has far-reaching implications on overall
   organizational operations and productivity. An investment in network readiness assessment ensures
   accurate planning and budgeting, minimum impact on the organization, a dramatically increase in the
   chances of success, and provides a safe fallback position in case things don’t go as planned.

So, our recommendation? Do an IP network readiness assessment—the effort required will
definitely be worth it, as it will help you plan a successful IPT implementation project that:
   •   Meets its target dates, as both the resource effort and time needed to upgrade the IP
       network are factored into the IPT rollout plan
   •   Comes in on or under budget, as the cost of the IP network upgrade will be factored into
       the overall cost of the deployment and budgeted for accordingly
   •   Meets user expectations of telephony performance without impacting on organizational
       productivity




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The Network Readiness Assessment Process
Conducting a successful and valuable network readiness assessment is not as simple as
simulating some calls over your IP network and noting the results. Here are some more horror
stories from real organizations, but this time, they are from organizations who did conduct a
network assessment but who still ended up with serious problems after going live with IPT.

   Horror Story #3
   A medium-sized enterprise set up their automated assessment tools to run overnight so that the
   telephony traffic wouldn’t degrade the network QoS demanded by their users. The results of the
   assessment showed acceptable voice quality for all calls. An initial implementation was then rolled
   out and the voice quality was very low because the voice traffic was now competing with other IP
   applications for bandwidth.
   Lesson Learned: A theoretical network readiness assessment can be accomplished completely
   offline to avoid negative impact during initial phases but real traffic must be simulated during real
   business hours to determine the true impact of the new IPT traffic prior to actual implementation.




   Horror Story #4
   A large distributed IT enterprise ran their own assessment and completed their network upgrade
   based on this information. However, when they completed implementation, they found that a large
   percentage of their off-net call attempts were suffering from very low voice quality. After acquiring a
   management tool, they used it to identify that, during peak hours of the week, there was insufficient
   bandwidth available to their primary PSTN gateway. It turned out that they had forgotten to simulate
   off-net traffic in their assessment.
   Lesson Learned: Use existing call records from telephone carriers, PBXs, and other sources to
   ensure that the network readiness assessment includes all calls, both on-net and off-net, and that call
   volumes can be supported during the busy hours.




   Horror Story #5
   A university was conducting an assessment and based the distribution of their simulators on their
   network diagrams. One campus was shown, in the plans, to have a 10Mb link, so they designed the
   assessment to have 80 calls occurring simultaneously across this link. They found that the voice
   quality for these calls was appalling. They lowered the number of calls being simulated and
   discovered that acceptable call quality could only be achieved if the number of calls across this link
   was limited to 15. This made no sense, so they sent an engineer out to investigate. It turned out they
   did have a 10Mb switch on that link, but only 1 E1 card (that is, a 2Mb link) was connected to it—their
   network diagrams had been incorrect.
   Lesson Learned: Always validate configurations, system capability, equipment inventory, call
   capacity, and so on.




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In a Perfect World
In a perfect world, with an unlimited budget of time and money, the network readiness
assessment would be a carefully applied four-step process. The first step would be to gather
information from the live network. This information would be merged with projected network
traffic loads anticipated from new IP telephony applications, which would then be subjected to
static modeling, which is the process of creating what would, in effect, provide a theoretical
snapshot at a moment in time. The output of the static model would provide the input for
dynamic computer simulation, which would provide the basis for a proof of concept test. Then,
and only then, would an actual field test and field trial be undertaken. Most organizations know
that the investment of time and money in such an exhaustive series of steps is not warranted.
That is, of course, until they either fail in their attempts to migrate voice over to their IP network
or come very close to such a failure. It is only then that they appreciate what they have risked.

In the Real World
Sadly, in the less-than-perfect world of time and budget constraints where most of us live, it is
generally not possible to go through this entire network readiness assessment process. Therefore,
the minimum set of steps recommended to get the best value out of your assessment is to
estimate projected traffic, perform a simple overnight assessment test run and, if successful,
move to a daytime assessment test run, starting with low assessment traffic levels and gradually
building up to simulating the estimated traffic. If this set of procedures is successful you could
then move to a pilot phase. However, by adopting this minimal approach, you still have the
opportunity to stop simulating traffic if the test begins to seriously impact the network and other
applications and users. Unfortunately, many organizations move straight to a pilot, usually with
less than acceptable results.

Understanding the Key Metrics of a Network Assessment
Whether you are conducting your IP network readiness assessment yourself or using an
assessment service provider, it is good to understand the key metrics and factors involved.
The PSTN has evolved over time into a finely tuned network that has been optimized for
delivering voice traffic. In business especially, for years you heard the phrase “toll quality voice”
to describe a service level suitable for business. And of course, you have heard a description of
service good enough to hear a pin drop. The Mean Opinion Score (MOS) described in Chapter 2
has been the accepted benchmark for measuring this quality. This network has been built and
conditioned to do one thing really well—deliver voice conversations. Your data network has
been optimized and honed to support the unique set of data applications your business requires.
Integrating telephony services will place new, sometimes-inflexible demands on the resources of
your network.
Data networks use IP to deliver traffic. IP is described as being unreliable and having no
guarantees. It is a “best efforts” protocol. It was designed to carry sporadic, unpredictable traffic
loads that burst to peak volumes at times. You’ve probably heard the phrase “bursty in nature” to
describe most IP traffic. At the transport layer, above IP, there is Transmission Control Protocol
(TCP) and User Datagram Protocol (UDP). TCP provides the ability to guarantee delivery
through a process of synchronization and acknowledgement messages, but this delivery
guarantee also adds overhead.

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Voice Traffic                                         IP Traffic
Connection-oriented—A dedicated path is               Connectionless—Voice conversations are digitized
established for each telephone call.                  and bundled inside IP packets, then transmitted
                                                      over the best route based on routing protocols.
Calls are long in duration (3 minutes on average).    Although IP telephone calls have the same duration
                                                      as traditional ones, the individual packets are small,
                                                      so conversations are cut up into many packets,
                                                      typically lasting only a few milliseconds.
Delivery is guaranteed once the call path is          Best efforts are made to deliver traffic but there are
established.                                          no guarantees.
Designed to use specific bandwidth. The PSTN          Uses the bandwidth that is available.
uses a 64Kbps voice channel.
Real-time voice traffic is very sensitive to delay.   IP data traffic is often delay-insensitive.

Table 3.1: Voice and data traffic comparison.

What you see is that IP data networks have very different basic characteristics than traditional
voice telephony networks:
    •   Unlike the dedicated circuit connection the PSTN establishes to carry a phone call, IP
        network traffic is composed of packets. Packets tend to be short in duration, although a
        phone call consists of hundreds, even thousands, of packets. Packet data is generally
        bursty in nature. Since packets are small in size, they are of short duration in terms of
        time spent on the network, and they can be routed over many diverse paths and carry
        addressing or routing information inside each packet. IP doesn’t establish circuits, it’s
        connectionless.
    •   IP packets aren’t generally delay-sensitive. Email messages can be delayed 60 seconds,
        or 60 minutes without adverse impact to the “conversation.” IP traffic is historically non-
        real-time traffic. Delays are expected in an IP network. IP is a best efforts protocol with
        no guarantee of delivery. The PSTN is designed to minimize delay and provide near-
        instantaneous delivery of voice conversations.
    •   IP uses all the available bandwidth when it has data to deliver. IP doesn’t require
        dedicated bandwidth.
Today, IP networks are designed, redesigned, and constantly reconfigured to carry any kind of
traffic. With digitization, any media (voice, video, data files, and so on) can be carried inside IP
packets. The most common are data, voice, and video. The challenge you face in unifying
communications is in delivering a business-quality telephony service with all the characteristics
users expect, such as clarity, fidelity, and near-instantaneous delivery.




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To understand the key metrics that will impact on the quality of your IP telephony service, let’s
take another look at the results of the Telephony Performance Expectations audit that was
described earlier in this chapter. User expectations regarding telephony performance are
generally quite consistent, as TDM has trained users to have high expectations. These
expectations can be summarized as:
   •   High availability: A user expects to get a dial-tone every time they lift a phone handset.
   •   Call quality: A user expects that, if they dial the number correctly, then they will get
       either a ringback tone or a busy tone within no more than a few seconds—users become
       impatient quickly if they get no tone at all and get irritable if they consistently get a
       network congestion tone or announcement.
   •   Voice quality: A user expects to be able to hear the other party in the call clearly and does
       not want to hear echoes of their own voice. A user considers it to be unacceptable if they
       have to abandon the call because of poor voice quality.

   It’s important to note here that call quality and voice quality are both critical, but very different aspects
   when you look at quality of service in VoIP solutions. Voice quality focuses on the fidelity and user
   experience from the sound of the call. Call quality includes voice quality as a factor, but also reaches
   more broadly into the total user experience. How quickly was dial tone provided? Did touch tone
   dialing work on the first attempt? Do connections across the networks, between offices (or hops
   across the IP network) process quickly? Do calls complete as expected or do they fail after all digits
   have been dialed? Call setup delays or failures negatively impact the user experience, reducing call
   quality regardless of the actually quality of the voice traffic.

When a user perceives unacceptable call or voice quality, that user is likely to terminate the call.
If this happens frequently, dissatisfaction with the service lingers. This perception is very
difficult to overcome, so you are better off to plan and re-architect your IP network in advance,
to make sure you don’t end up in this situation.

High Availability
It’s important to embrace the telephone network services as a mission-critical component of
daily business operations. The PSTN has been engineered to provide 99.999 percent uptime of
the telephony service, often referred to as “five nines” reliability. This is equivalent to
approximately 5 minutes of downtime per year. Corporate data networks have rarely offered this
level of reliability in the past. Think about your corporate network. Have you experienced more
than 5 minutes of network downtime in the past year? At this phase of the VoIP/IPT project, it’s
crucial to understand that some redesign effort may be necessary to provide suitable network
reliability to meet the voice and call quality requirements associated with a telephony service.




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Five nines reliability is not just about uptime of routers, switches, interfaces and links in the
network—it’s about ensuring that the network can still provide telephony service even when
there has been an outage of any kind, anywhere in the network. The general requirement on an IP
network to deliver five nines reliability is that it can still provide full service to all users under
any single point of failure. This requires that there be redundant paths and fault-tolerant or high
availability equipment to guarantee availability in the event of the failure of a given node or
device. It also requires that, even under any single point of failure, there must be sufficient
bandwidth available for use by the IP telephony service so that it can still meet user expectations.
In enhancing your IP network to support your IP telephony service, it will be crucial to identify
the IP network availability objective that will be committed to the IP telephony application in an
SLA.

Call and Voice Quality
Both call and voice quality require adequate bandwidth, low latency, low jitter and low packet
loss.

Bandwidth
There must always be enough bandwidth in the network to handle both your IP telephony traffic
and all the traffic associated with your other distributed data applications, even under any single
point of IP network failure. You probably already have a good idea of the usual profile of the
bandwidth usage of your existing data applications, but you also need to calculate the bandwidth
requirements for IP telephony. To do this, it’s a good idea to first look at your existing TDM call
traffic patterns, as the telephony habits of your staff are very likely to remain consistent after
you’ve deployed IP telephony. More detailed guidelines for gathering these call traffic patterns
are covered later in this chapter, but the basic idea is to base your bandwidth requirement
calculations for each office around the number of calls placed and received during the busiest
hour of the busiest day of the year for that office.
Once you have an idea of the maximum number of simultaneous calls that will be handled by
each office in your organization, you’ll next need figure out what your policies are going to be
around codec utilization, as this will impact directly on the bandwidth that will be used by these
calls.

Choosing the Right CODEC

To transmit voice over a data network, the voice conversation must be packetized. To transmit
the conversation in packets, you need to digitize it. Voice digitization is not a new technology. In
the 1960’s, the US phone companies began using a digital carrier system called T-carrier while
the rest of the world began using a slightly different system called E-1. Initially T-1s and E-1s
were used as a trunking technology between phone company central offices. Today, the T-1 or
E-1 circuit is a common business connection.




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Earlier, we saw that voice represents an analog signal. To use IP, you must first convert it into
digital format. This conversion is commonly accomplished in the traditional voice services
through a technique known as pulse code modulation (PCM), using a coder and decoder (or
codec). Using PCM, analog voice conversations are sampled 8000 times per second. A technique
called pulse amplitude modulation (PAM) is used to convert each sample into one of 255
possible 8-bit words. Through a process of compressing and expanding (called companding),
noise is reduced to help eliminate background hiss and changes in volume.
There are several voice processing standards that were developed by the International
Telecommunication Union (ITU, formerly the CCITT) to standardize encoding of digital speech.
In the PSTN, the G.711 standard is used worldwide.

   More information about encoding standards is available at the ITU Web site at http://www.itu.int/.


Voice Compression Techniques
Different encoding mechanisms use different levels of compression. Although G.711 encoding is
widely used, it typically generates a 64Kbps voice stream in each direction (that is, 128Kbps in
total for a G.711 call). For some businesses, quality considerations might make a different
encoding scheme more suitable. Greater compression of the voice reduces the bandwidth
requirement, but voice quality may suffer as a result.
Table 3.2 provides a comparison of different encoding schemes that are all widely used. It lists
the Codec types and algorithms used, the bit rate and sample size, and the algorithmic encoding
delay. It also shows the maximum possible Mean Opinion Score (MOS) for these codecs. It is
important to remember that factors such as encoding delays are cumulative with other delay in
the network. If the total delay exceeds 250 milliseconds (ms), voice quality will suffer audibly.

   These algorithms are described in depth in ITU-T standards (http://www.itu.int/) and a wide variety of
   papers.


ITU Codec        Coding Scheme                       Bit Rate       Sample        Encoding     MOS
                                                     Each Way       Size          Delay
G.711            PCM                                 64Kbps         8 bits        <1ms         4.4
G.726            Adaptive Differential Pulse Code    32Kbps         4 bits        1ms          4.2
                 Modulation (ADPCM)
G.728            Low-Delay Code Excited Linear       16Kbps         40 bits       2ms          4.2
                 Predictive (LC-CELP)
G.729            Conjugated Structure Algebraic      6Kbps          80 bits       15ms         4.2
                 Code-Excited Linear Predictive
                 (CS-ACELP)
G.723.1          Algebraic Code-Excited Linear       5.3Kbps        160 bits      37.5ms       3.5
                 Predictive (ACELP)

Table 3.2: Voice encoding schemes




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The following list highlights these encoding schemes:
   •   Pulse Code Modulation (PCM), or G.711, is the approach used in the PSTN today and is
       widely used in many VoIP environments to ensure consistent voice quality. Virtually
       every equipment vendor in the market supports G.711. This is also by far the most
       common codec used in intra-office calls, i.e. where calls are within the LAN.
   •   G.726, known as Adaptive Differential Pulse Code Modulation (ADPCM), could reduce
       the bandwidth requirements by half while only sacrificing 0.2 of a point on perceived
       quality in the MOS.
   •   G.728, or Low-Delay Code Excited Linear Predictive (LC-CELP), coding is widely used
       in voicemail systems for digitizing stored voice messages.
   •   G.729 can deliver an 8-kilobit sample with less than 16ms of processing time. It delivers
       the best tradeoff of all the standardized codecs, giving the best balance between low
       bandwidth utilization with very reasonable voice quality. It is widely used on the WAN
       for inter-office or trunked calls. As an aside, if you’re considering using VoP rather than
       VoIP, the G.729 codec standard has often been used in Voice over Frame Relay (VoFR)
       and is supported by many frame relay equipment vendors.
   •   G.723.1, or Algebraic Code-Excited Linear Predictive (ACELP), creates models of the
       human voice, then predicts what the next sound will be. For efficiency, it encodes the
       difference between the actual sound and the predicted sound, and the difference is
       transmitted to the receiving end. In some implementations of this codec, users
       complained women and children’s voices were not represented accurately, probably due
       to their higher pitch.
Note that the codecs with greater compression rates (G.728, G.729 and G.723.1), while needing
less bandwidth, are generally considerably more resource intensive and therefore generally
require more expensive equipment. So if you’re examining how your codec utilization policy can
save your organization money, keep in mind that the use of these codecs generally requires
greater CAPEX, with potential long-term savings in OPEX.
A much simpler approach to bandwidth management for IP telephony has been to overprovision
the network. This is analogous to the gigabandwidth approach mentioned earlier, but constrained
to within the corporate network. In many cases this means adding bandwidth. Although over-
provisioning might work initially, in the long run, it’s a path to problems. First, it requires
investment. Upgrading the network capacity in terms of connections, switches and routers can be
a very expensive proposition. This approach might work for a time in a small LAN, but in the
metropolitan or WAN or a larger enterprise, it’s often too expensive to be a practical solution.
One thing is certain: If the bandwidth is available, users will fill it. Already existing data
applications can quickly consume the additional capacity, starving voice services.
Overprovisioning also rarely improves call quality issues related to dial tone, call completion
rate, and factors that might degrade service within a single node, such as codec processing or
packetization delays.
You are far better off implementing tangible mechanisms for measuring the QoS and ensuring
that network resources can support the required traffic load without service-affecting
degeneration of call quality.



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QoS
Voice and data are different applications. They each place different demands on the network.
The requirements for any application are often described as a class of service. The unanswered
question may be: How many classes of service are required to meet the complete QoS needs of
all the different business applications on the network?
In order to meet performance-level requirements, the network needs to take into account all these
factors:
   •   Throughput is most often measured in bandwidth. It’s a simple measure to determine how
       much of the data inserted into the network makes it through to the destination.
   •   Error Rate is a measurement of data loss in some form. Remember that IP is a best efforts
       protocol and relies on higher-layer protocols, such as TCP, or the application itself to
       overcome errors and guarantee delivery. Legacy mainframe applications are generally
       very intolerant of high error rates and data loss.
   •   Delay is a fact of life in IP networks. Even the Ethernet network used inside an office is a
       contention-based technology. At its roots, Ethernet is a Carrier Sense Multiple
       Access/Collision Detection (CSMA/CD) technology regardless of speed. At the IP layer,
       routers use statistical multiplexing to process traffic and user data has a very bursty
       characteristic. There will be some transmission delay in any network. Nodal processing at
       each “hop” of the network adds to delay. Delay is also cumulative from one end of the
       transmission path to the other.
   •   Jitter is the variation in delay. Because packets can take different routes across the
       network, delay variation in IP networks is common. Jitter in voice conversations results
       in unintelligible conversations that sound “jerky.” Jitter buffers can be deployed to reduce
       this impact, but buffering can only do so much to overcome impairment.

   New approaches to delay and jitter management include intelligent buffering that inserts this variation
   in delay by inserting it in dead spots in the conversation. Pauses between words or sentences might
   be indistinguishably increased. This is often referred to as delay distribution.

   •   Scalability is another factor. As companies grow and businesses change, the ability of the
       network to grow with evolving business needs is an important consideration.
   •   There have been many studies that demonstrate the most expensive component of
       network services to be management and administration of the network. This is a factor of
       return on effort that can directly impact OPEX. A network that is very labor intensive to
       manage requires more effort, increasing both the OPEX and the TCO.




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Today, conscious effort is made to design networks to meet specific performance criteria
required by business applications. Telephony as a network service is one more application for
which you need to design. The ongoing danger is that every new application potentially creates a
new class of service. Network complexity can spin out of control. To maintain manageability,
most network engineers favor using only a few critical service classes:
   •   Quick delivery supports real-time traffic such as voice and video
   •   Guaranteed delivery supports mission-critical traffic such as mainframe systems
   •   Best efforts delivery is standard IP to support everything else
   •   A management class to ensure the network can always be managed even at peak traffic
       loads
In a well-managed network with adequate bandwidth, QoS is simply a prioritization mechanism.
One approach to QoS might be to aggregate similar traffic types to ensure each type takes the
appropriate network path as a traffic engineering technique. There are several approaches to
providing QoS for IPT. Each adds overhead in some fashion to support this prioritization. You
might think of it as a “traffic cop” function.

Approaches to QoS with IP
The most common techniques for delivering QoS in IP networks fall into three broad categories:
   •   Signaled QoS—Integrated Services (IntServ) introduces a signaling protocol to IP. That’s
       the overhead—another protocol. Using IntServ, the user’s application sends a call setup
       signal to the network. This is basically a request for a set of service delivery parameters
       required to complete a call. Using the approach, the network can check resource
       availability and deliver traffic accordingly. This approach is very similar in process to the
       circuit setup and teardown associated with a telephone call on the PSTN. In the PSTN,
       ISDN (access networks) and SS7/CSS7 transit networks provide this signaling
       mechanism.
   •   Provisioned QoS—A second approach requires advance planning and design.
       Differentiated Services (DiffServ, also called the DiffServ Code Point—DSCP) requires
       specific routes through the network predefined and available for each type or class of
       traffic. Paths might be pre-existing as part of the total network design or involve yet some
       other protocol to set up resources on demand. DiffServ is most often used as an
       aggregation technique to direct similar traffic along the same major network paths.
   •   QoS “Shim” or Bypass—MultiProtocol Label Switching (MPLS) is an approach that
       eliminates the normal hop-by-hop routing IP uses. It adds a “tag” to each packet that
       essentially shortcuts delivery to the best available path for that type of traffic. MPLS is
       compatible with frame relay and ATM networks. It has become a widely used approach
       to delivering QoS in many environments.
There are other approaches used in providing QoS for VoIP call quality. 802.1p/q tagging is
frequently implemented with VLANs at Layer 2 inside parts of the network. IP Type of Service
(TOS) markings and IP precedence approaches have also been popular. There are a variety of
Weighted Fair Queuing techniques used by different vendors to specifically address the issues of
delay and jitter.



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Each of these methods boils down to prioritization, aggregation, or some combination of the two.
These techniques either request network resources or rely on some other protocol or mechanism
to ensure the necessary paths are available through the network. These QoS approaches also
provide traffic management and in some cases traffic shaping capability.
QoS implementation presents a hurdle because there are many options and approaches to choose
from. Different vendors often prefer different approaches and techniques, sometimes making
comparison of solutions challenging. Because QoS is really an approach to resource management
in the network, distinguishing traffic types is a vital part of the process.
These issues not only need to be dealt with inside the LAN, they become even more critical in
the WAN. WAN QoS raises the complexity significantly. If you maintain your own private
WAN, you have direct control over traffic handling. If your WAN consists of circuits purchased
through a network provider, you’ll need to review, and perhaps revise the SLA you have in place
with this provider, to guarantee that your prioritized traffic is properly handled. At the very least,
you should ensure that this SLA addresses the key network performance requirements that you
need in order to be able to meet your own telephony SLAs and ensure the delivery of any real-
time intensive streamed services or applications in a manner that meets your users’ requirements.
An effective SLA that fully documents IP network provider commitments should include:
   •   Availability assurances, usually expressed in uptime percentage of the network from an
       end-to-end perspective
   •   Throughput expressed in guaranteed bandwidth for a given traffic type; this might be
       expressed in kilobits or as a percentage of the network carrying capacity
   •   Delivery guarantees against packet loss expressed as percentage of lost/dropped packets
   •   Tolerable jitter characteristics; this facet of QoS will be addressed in more depth in the
       next chapter
To be effective, the voice service provider, whether they are a third party provider or an internal
department, must monitor the network to ensure that the underlying network-centric SLAs are
being met by the network provider. Chapter 6 will dig deeper into monitoring, and Chapter 7 will
explore ongoing optimization.
WAN providers, the telcos and ISPs, have widely adopted MPLS technology, but you will need
to negotiate how prioritization tagging applied within your network will be treated once it hits
the provider network. This is the only way you can ensure suitable WAN performance. If the
provider remaps MPLS tags to their scheme for their backbone QoS, you’ll need to understand
their approach, and make sure that you’re getting the service delivery guarantees you need for
your business services.




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When testing network performance, especially across a WAN, it’s crucial to test the end-to-end
connection performance, not just node-to-node. Delay is cumulative. So is jitter. Voice is an end-
to-end service and can only be tested effectively by simulating live calling patterns.
Traffic engineering and testing must consider peak traffic loads. Testing in a controlled lab
environment will yield very different results than an implementation in your production network.
When evaluating your QoS approach, use a combination of network testing tools. Traffic
generators need to simulate all traffic types in conditions that match your busiest times.
Analyzers and monitors will help verify that bandwidth, throughput, error rate, delay, and jitter
meet the requirements for IPT voice quality.
The right balance of network performance and QoS can be blended with MPLS and 802.1p/q and
VLANs to enhance QoS. These techniques, when designed methodically, can also improve
security by isolating voice service from other data traffic while enhancing call quality.

Tips and Tricks for A Successful Network Assessment
Now, having covered the key elements of a network assessment, let’s have a look at a few tricks
that will help you avoid the most common mistakes, so that you can get the best value out of
your network assessment.
It’s important that you methodically test the characteristics of the network to ensure that voice
and call quality requirements can be achieved, even at peak telephony call rates, during peak
utilization times on your IP network and during any single point of failure outages in your
network.
It’s also important to maintain a holistic view of network resource management. As a
counterpoint to voice and call quality, it’s crucial that IP telephony traffic doesn’t starve other
mission-critical data applications of network resources, primarily bandwidth. You have to plan
for balance between real-time critical telephony traffic and other traffic (such as bursty email and
Web traffic) that, while less time critical, is equally important to the productivity of your
business.
You need to understand aspects of the data flows on your network such as traffic type, frame
size, any prioritization schemes in use (MPLS or 802.1p/q, for example), utilization (both normal
utilization and peak conditions), latency or delay, jitter, and loss. Measuring these factors will
help in benchmarking service and application performance.




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Simulating Calling Patterns
To ensure that your network assessment accurately measures the impact of telephony on your IP
network, it is crucial that all IP telephony traffic modeling and simulation conducted during the
assessment accurately reflects your organization’s existing calling patterns. Call usage reports
from either the current provider or existing system provide a wealth of data about your existing
inbound and outbound telephony traffic flows.
Whether the current telephone service is delivered through a company-managed PBX or a telco-
provided Centrex-like service, this information is crucial and should be readily available either in
the form of reports or Call Detail Records (CDRs). These reports help to identify important data
elements such as the busy hour of the business day, the busy day of the week and month, and the
peak call volumes that must be supported in the new system. Enterprise business will find that
remote offices might have very different traffic patterns and requirements than the corporate
headquarters location. Remote offices will also vary depending on the role they play in the
enterprise. A sales office will place mostly outbound calls, while a customer service call center
will mostly receive inbound calls. It’s important to be model all these different calling patterns in
your assessment, so that you get an accurate view of the impact of these different calling patterns
on different parts of your network.
Many businesses experience cyclical trends. In agriculture, the growing season may be a factor.
In transportation, summer vacation may put more traffic on the roads. Financial services often
see cycles related to the fiscal year rather than the calendar year. It’s prudent to evaluate these
cycles to identify historical calling patterns for the past several months or even for the past year.
Your assessment needs to model and simulate your peak activity periods, not just day-to-day
business operations.
Simple VoIP calculators such as those available at http://www.voip-calculator.com/calculator/
can help you identify key issues:
   •   Estimating the bandwidth needed to support a known number of phone lines
   •   Calculating the bandwidth required to support busy-hour call volumes
   •   Determining how many paths might be needed across a WAN
Network analyzers, whether commercial or open source, offer performance measurement tools to
help understand the current service levels provided in your network.

Simulating Gateways
Initially there were two types of gateways used for early IPT solutions. A signaling gateway
provided the ability to connect to the SS7 network in the PSTN. A trunking gateway was used to
connect the actual voice paths. This technique of separating functions into “command/control”
and “traffic bearing” components was widely adopted for efficiency and to provide economy of
scale. As multi-media usage has evolved in the past 10 years, we often heard the term “media
gateways.” Although it has remained a common practice to separate the signaling or command
path from the media path, these functions are frequently consolidated into a single gateway.




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There are three types of media gateways most commonly deployed in an IP telephony
environment:
   •   Gateways to PSTN: These are gateways that perform a bridging function between your IP
       telephony environment and the PSTN, to facilitate the transit of outgoing and incoming
       calls from/to your organization. On the PSTN side of a gateway, the most common
       interconnect methods used are ISDN (PRI over T-1/E-1 or BRI over a single twisted-pair
       line) or analog, while the voice compression algorithm is PCM, as previously discussed.
       On the internal side of your organization, a gateway communicates with the rest of your
       IP telephony environment using IP telephony control protocols such as SIP, H.323 or
       MGCP and transits voice streams over the IP network using one of the codecs mentioned
       in the previous section. When someone in your organization makes an external call, the
       call is routed to the most appropriate gateway, which then initiates the call into PSTN and
       takes responsibility for mapping the PCM-encoded voice channel on the PSTN side into
       bi-directional RTP streams over the IP network using the appropriate codec. Similarly,
       for incoming calls to an organization, when the PSTN makes a call to the gateway, the
       gateway passes the call setup request into the IP telephony system, so that it can be
       routed to the appropriate end user device and, once the call is answered, again takes
       responsibility for mapping the PSTN voice channel into bi-directional RTP streams to the
       answering device.
   •   Gateways to traditional PBXs: Generally, an IP telephony migration may take months or
       even years. During the migration, it is necessary to be able to facilitate calls between
       users on traditional PBX handsets and IP telephony users in the IP network. This is
       accomplished by placing gateways between traditional PBXs or key systems and the IP
       telephony environment. The two most common form of interconnect provided by these
       gateways are analog lines and Q.SIG.
   •   Gateways to IP Telephony Service Providers: Many traditional telephony service
       providers are now providing IP telephony interconnects to enterprise and many new small
       IP telephony service providers are springing up every day. It is therefore becoming more
       and more common for enterprises to route incoming and outgoing calls via an IP
       telephony service provider. In this case, the gateway is in fact an IP-to-IP gateway, as it is
       actually routing IP telephony traffic on both sides. The major difference in this case is
       that each side of the gateway is in a different domain—the internal side of the gateway is
       in the enterprise’s domain, whereas the outside of the gateway is in the service provider’s
       domain. As such, these gateways are responsible not only for bridging between telephony
       environments, but also perform a vital telephony firewall function. Session Border
       Controllers (or SBCs) fall into this category of gateway.




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Irrespective of which types of gateways you will be implementing in your environment, in all
cases, they are likely to have specific performance requirements on the enterprise IP network and
on the external (PSTN, traditional PBX or IP Telephony Service Provider) side of the gateway.
This needs to be taken into consideration. If the data network bandwidth and QoS on the path to
a gateway doesn’t complement the trunking requirements of that gateway (i.e. the forecast
number of simultaneous calls that that gateway will be required to support during peak hours),
problems will follow. Overlooking details such as this can result in performance bottlenecks in
your IP network or between the two networks. Don’t forget to model and simulate gateways and
their expected call traffic load and use a broad combination of network analyzers and monitoring
or assessment tools to perform a comprehensive evaluation of your network’s throughput and
performance at these potential bottlenecks.

Plan For The Future
The primary motivator for conducting your network readiness assessment might be to plan
towards an IP telephony deployment, but don’t stop there. Don’t just assess and understand voice
service requirements in conjunction with all existing data services. Look to the future and
evaluate business plans for other new services too. If you’re planning a major CRM system
rollout next year, that evolution has to be factored into the plans for voice integration. Just-in-
time planning might seem effective at first glance, but in the long term, often costs more than a
methodical approach with comprehensive data gathering and information analysis. This
methodical approach will yield a more resilient network design with the capacity to not only
support a successful integrated voice service but also provide a growth path to support all the
unified communications needs of your business.


Assessment Tools
If you opted for the “do-it-yourself” approach, you will have access to sophisticated tools
embedded in the routers, switches, gateways, end systems, and other components of your
networks. However, even if you haven’t chosen the DIY option, there is still an array of tools
you can use to provide “the big picture” as well as pinpoint areas that require a more granular
focus.

Third-Party Tools
Third party tools for modeling, monitoring, measuring, and managing are invaluable. They
provide broad, cross-platform visibility into facets of network operation that far exceed what
humans alone can do without those tools. These third-party tools consist of a broad array of open
source and commercial solutions. The selection of these tools is the subject of the third-party
tools selection report card described in the next section and their application to the life cycle of
the network are described in further detail in subsequent chapters.




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Third-Party Tools Selection “Report Card”
Choosing third-party network modeling, monitoring, measuring, and management tools can be a
daunting task. The job of reviewing any given tool is not so difficult but, rather, the comparison
of multiple tools from multiple systems, the alignment of the various capabilities, and
determining what is important can be staggering and is no peripheral task. The tool or tools in
this group that are selected are critical path, make or break, choices that, to a very large extent,
management might not even consider critical to success, thus, the acquisition of such tools may
have to be justified, fought for, and ultimately won and put to good use.
To assist you in your selection process, I have created the Modeling, Measurement, Monitoring,
and Management Tools Report Card (4MTRC). The report card is a set of six Microsoft Excel
spreadsheets. There are five identical report card spread sheets that may be used to evaluate as
many as five separate 4M tools. The sixth spreadsheet allows you to compare the tools side by
side and support an intelligent selection process. The spreadsheets are locked to avoid
inadvertent changes but a password is provided so that the spreadsheets may be modified by an
organization for their own use. Those knowledgeable in the inner workings of Excel can take a
look and see that there is nothing particularly clever about the coding. The real value is the
knowledge that is embedded in the categories that were chosen for comparison as well as the
weighting factors applied by the author to the different categories. For users with limited
resources these two factors—the comparison list and associated weighting—will be an
invaluable source of assistance in the selection process, while for the more resource-rich
organization, will provide a convenient starting point for building your own customized selection
report card for your organization.
Further, it can also be extremely difficult to justify to your management team the cost of
purchasing network modeling and assessment tools that will only be used once. Fortunately,
there are lots of consultants and systems integrators out there who not only own assessment
tools, but who are also experts in conducting fast and efficient network readiness assessments.
The purchase of a one-off assessment service from one of these experts is usually much easier to
get budget approval for and has the added advantage that these experts are unlikely to overlook
any significant details while conducting the assessment.
If you choose to go this route, use the report cards to assist you in differentiating between the
capabilities of the consultants who bid for the assessment contract, as this will help you to
narrow the field down to those consultants who can provide a comprehensive assessment service.
Also, when you’ve selected an IPT assessment consultant and they come on-site, take advantage
of their presence and assign some of your soon-to-be IPT operations staff to work with them—
this is a great opportunity to get some free hands-on IPT training!
Finally, if you are thinking of using a systems integrator for the design and deployment of your
IPT system, be aware that many systems integrators include IP network readiness assessment as
part of their standard implementation service. However, keep in mind that the assessment service
provided by an SI as part of a full rollout may not be as comprehensive or exacting as one
provided by an independent network assessment specialist.




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Summary: The Road to Success
While “your mileage may vary” in this chapter we have provided some basic steps every
company must take to successfully integrate voice and data network services. IPT is one of many
new technologies being deployed today. As previously discussed in greater detail, you will need
to consider these basics when integrating any new technology:
   •   Requirements analysis—First map out your objectives. Why are you making this change?
       Don’t make the decision to move to IPT and then try to explain why. What business
       needs are you addressing? What factors drive the decision? Document your expectations
       of the benefits you expect to see. The success of a VoIP deployment hinges on the data
       gathering and analysis. The more you understand ahead of time about your requirements,
       the further you are ahead of the game.
   •   SLAs—Formulate SLAs on which the success of the IP telephony deployment will be
       measured. Evaluate the impact of these telephony SLAs on the IP network SLAs that you
       have in place with your WAN service provider. Formulate new IP network SLAs if
       necessary.
   •   Planning—Be methodical and document every facet of the entire IP telephony
       deployment/implementation process. If there is ever a time to be overly thorough in
       documenting plans, this is it.
   •   Assessment testing—After you’ve documented your telephony requirements and mapped
       them into requirements on your IP network, test your network. This is the only way to
       know for certain that your network can support your VoIP requirements. Test and
       document all the related network parameters that influence successful voice delivery.
       Thorough analysis in the beginning guarantees that you know what to test at this point. It
       gives you the assurance of planned testing. Ad hoc testing of things that you guessed at
       won’t increase your chances of success. Integrate your testing to support a fully
       converged network. Test everything, not just VoIP services capability. Your existing
       business applications on the WAN and LAN need to be tested while voice traffic is being
       simulated. It’s also a good time to test your security posture. If you design without
       considerations for security, run all your VoIP traffic through firewalls later on, you’re in
       for surprises. Security elements add overhead and delay. They can impact performance,
       and it’s absolutely critical that you test to your coming real-world as thoroughly as
       possible.
   •   Acquiring the right resources—Resource acquisition means many different things.
       Sending your technical staff to training before you’re halfway into the project is resource
       acquisition. For other companies, bringing in outside help from either a consultant or a
       trusted vendor partner is a tried and true method of acquiring resources in the short term.
       There is no shame in admitting you need expertise from outside, then working with a
       consulting partner. Leverage expertise where you can. Partners whose core competency is
       successful implementation can be a huge asset in deploying new services.




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   •   Looking ahead—Don’t look at integrating voice, whether you’re looking at it as VoIP or
       IPT as the final destination. It’s a step along the road of your constantly changing
       network environment. Evaluate the scalability of your chosen solution against future
       business strategies. Can your network continue to grow to support your business 3 years
       from now? 5 years? Once you can deliver voice services in your environment, can you
       continue to grow at a sustainable rate? How will new business applications affect voice
       service? This is a perfect time to review the overall future capacity of your network in
       conjunction with your strategic direction.
After reading this chapter, hopefully, you appreciate the vehicles as well as the track and have a
deeper understanding of their relationship. This chapter has probed more deeply into the real
value in diligent and thorough assessment of the network prior to beginning implementation. It
has followed the methodical path established in Chapter 1 and has begun the exploration into the
requirements for suitable bandwidth and how you might evaluate QoS approaches to ensure
delivery of acceptable call quality. The next chapter will continue to dive deeper into the design
and deployment life cycle.
The focus has continued to be on providing specific, actionable items that you can perform for
your own situation and, and you can use the provided report card to assist you in the selection of
the proper tool or tools for modeling, measuring, monitoring, and managing your multi-media
network.
Chapter 4 will describe the process to translate the planning and assessment information
developed in this chapter into a design upon which you can do pre-deployment testing. At this
point, you have completed the first step in the IPT life cycle. Chapter 4 will explore the next step
in making your VoP project a reality.




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Chapter 4: Design and Pre-Deployment Testing
Chapter 3 described the first phase in the life cycle of any successful IPT project—planning and
assessment. The information that you gathered and the steps that you took in that phase set the
stage for the next step: design & pre-deployment testing. As Figure 4.1 shows, planning and
assessment, design and pre-deployment testing and implementation are all initial phases. After
these initial phases, you enter the actual cyclical part of the life cycle of your IPT project.




Figure 4.1: IPT project life cycle.

The major areas within Design and Pre-deployment Testing are:
    •    Network Design Considerations
    •    Translating Needs to Service Level Agreements (SLAs)
    •    Validation of Design
    •    Testing      Feedback        Network Modification   Testing Loop
We will discuss each in great depth in this chapter.




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Network Design Considerations
Chapter 3 covered a variety of areas fundamental to network design. Although this chapter will
not completely rehash the discussion, it will pull certain assumptions from the previous chapter
that will provide a basis for the exploration of network design in this chapter.

The Team
At this point in our project life cycle, the company implementing the IPT project may or may not
have engaged, or even considered engaging, outside assistance. Regardless of whether it has
come up or not, this is the time to make some difficult decisions. At this point, the company
would choose a “do-it-yourself,” “managed service,” or turn-key “outside service provider”
approach—or depending upon geographic coverage and other considerations, some hybrid of
these solutions. It is also important to distinguish the team that will provide the network transport
from the team, or part of the team, responsible for the telephony and multimedia systems
themselves. Table 4.1 shows prospective team members, their possible roles in the project, and
the pros and cons of selecting a specific prospective team member. Considerations of budget
must be balanced with the fact that no other single project is likely to have such a profound
impact on the overall operation of the organization as a project involving telephones.
 Prospective Team
                          Role                       Pros                          Cons
 Member
                          Provide bridge between     A lot of knowledge is         May not see the
 Current PBX Vendor       current capabilities and   locked in the current PBX     importance of their role or
 Professional Services    capabilities of new        configs, tables,              may be reluctant to help
 or Consultant(s)         system                     documents, and so on          in the decommissioning
                                                                                   of their equipment
                          Benchmark current          Know current network          Not familiar with voice
                          network                    Already on payroll            May be rivals of voice
                          Assist in testing          No “spin up” time for IP      group
 Current IP Network
 Tech                     Get cross-trained in       networking                    May view voice
                          voice to be multi-tech                                   people/app as simple and
                                                                                   inferior to data
                                                                                   people/app
                          Collaborate on design      Know current procedures,      Not familiar with voice
                          Collaborate on testing     procurement, budgeting,       May be rivals of voice
                          Get cross-trained in       and so on                     group
                          voice                      Know current “players”:       May view voice
 Current IP Network                                  techs, management,            people/app as simple and
 Mgmt                                                suppliers, carriers, and so   inferior to data
                                                     on                            people/app
                                                     Already on payroll
                                                     No “spin up” time for IP
                                                     networking
                          Ensure all voice           Know current voice            Not familiar with IP
                          functionality              needs, apps, and users        May be rivals of IP/data
                          synchronized               Already on payroll            group
 Current Voice Tech       Telephony system           No “spin up” time for
                          benchmarking and           telephony
                          testing
                          Get cross-trained in IP

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Prospective Team
                         Role                       Pros                          Cons
Member
                         Collaborate on design      Know current procedures,      Not familiar with voice
                         Collaborate on testing     procurement, budgeting,       May be rivals of IP/data
                         Get cross-trained in IP    and so on                     group
                                                    Know current “players”:
                                                    techs, management,
Current Voice Mgmt                                  suppliers, carriers, and so
                                                    on
                                                    Already on payroll
                                                    No “spin up” time for
                                                    telephony
                         Assist in install          Temporary                     Not familiar with network
Temp/Contract Multi-     Go away                    Go away after install         Will take “lessons
Tech                                                                              learned” and valuable
                                                                                  knowledge away when
                                                                                  they leave
                         Assist in voice and data   Know current network          More expensive than
                         benchmarking               Already on payroll            IP/data or voice techs
                         Assist in testing          No “spin up” time             Difficult to hire
Current Multi-Tech (if   Ensure all voice           Can provide cross-            Difficult to retain
they exist)              functionality              training to IP/data and
                         synchronized               voice techs
                         Ongoing support of
                         multimedia system
                         Assist in voice and data   Expand tech staff             Aren’t familiar with
                         benchmarking               Mandatory if “do-it-          network and users
                         Assist in testing          yourself”                     More expensive than
                         Ensure all voice                                         IP/data or voice techs
New Multi-Tech
                         functionality                                            Difficult to hire
                         synchronized                                             Difficult to retain
                         Ongoing support of
                         multimedia system
                         Ensure that the            They have intimate            May be difficult to work
                         underlying network         knowledge of the              with if their IPT solution
                         components—switches,       underlying infrastructure     was not chosen
                         routers, firewalls, and    and stand to benefit if
Network Equipment
                         so on—are capable of       upgrades are needed
Vendor
                         handling the load of the
                         new services, providing
                         appropriate quality of
                         service (QoS), etc.
                         Manage voice and data      Know current procedures,      Possible rivalries
                         issues                     procurement, budgeting,
                         Manage budget              etc.
                         Ensure system              Know current “players”:
IT/Network               reliability and security   techs, management,
Management               Coordinate integration     suppliers, carriers, etc.
                         of voice with VPN          Already on payroll
                                                    No “spin up” time
                                                    Data, voice, and video
                                                    responsibility

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Prospective Team
                    Role                        Pros                         Cons
Member

                    Assist in lab testing and   Temporary                    Not familiar with network
                    initial implementation      Go away after install        Does not know voice
                    Go away                     May have Best Practices      requirements in advance
Consultant                                      methodology to leverage      Will take valuable
                                                                             knowledge away when
                                                                             they leave


                    Assist in lab testing and   Deeper understanding of      Not familiar with network
                    initial implementation      equipment and software       Will take valuable
                                                May have Best Practices      knowledge away when
                                                methodology to leverage      they leave
Manufacturer Prof                               Should cross-train/do
Services                                        knowledge transfer
                                                Temporary
                                                Go away after install


                    Network benchmarking        Familiar with network and    May see telephony
                                                infrastructure critical to   service provider as rival
                    Bandwidth adjustments
                                                success
                    QoS Issues
                                                ISP provide infrastructure
                    VPN/tunneling/security
                                                VPN, tunneling, and
Internet Service    SLA-related issues          security are needed
Provider
                                                services and difficult for
                                                “do-it-yourself”
                                                SLA coupling to project
                                                critical


                    Benchmark current           Critical supporting skill    May see ISP and/or voice
                    network                     sets                         systems integrator as
                    Assist in testing           May already know             rivals
Data Systems
Integrator          Assess performance          network and
                                                applications/users
                    impact on data apps
                    Prioritization

                    Benchmark current           Critical supporting skill    May not already know
                    network                     sets                         network, users, and
                    Assist in testing           NOT needed if using          needs

Voice Systems       Get cross-trained in        telephony service
Integrator                                      provider
                    voice to be multi-tech
                    Assess performance
                    impact on data apps
                    Prioritization



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 Prospective Team
                            Role                      Pros                        Cons
 Member
                            Turn-key system           Turn-key                    Probably does not know
                            implementation            Alleviates need to          network or users and
                            Coordinate carriers and   interface to ISP            needs
                            service providers         No need to interface with   May be more expensive
                            Ensure SLA                carriers                    than alternatives
                            performance, telephony    “One-stop shopping”
                            features, and system      May be less expensive
                            reliability               than “do-it-yourself”
 Telephony Service                                    Price may include
 Provider                                             upgrade/technology
                                                      refresh
                                                      May include dedicated
                                                      Help Desk and/or account
                                                      team and/or on-site
                                                      personnel
                                                      No need to keep
                                                      telephony or multi-tech
                                                      skill sets on staff
                            Provide representation    Provide much-needed         Must be in touch with the
                            of the users in the       user viewpoint              agreed upon project
 Voice User
                            implementation process                                objectives or they may
 Representatives
                                                                                  hinder progress and risk
                                                                                  deadlines

Table 4.1: Prospective project team participants.

A big part of the selection of the different elements of the project team is to assign
responsibilities, work out any overlaps or conflicts, and test any assumptions regarding the
responsibilities of each group. And, in keeping with good project management practice, there
should be an actual human responsible for ensuring that each and every task is completed, on
time, on budget, and to the standard of quality set for the task.
Project teams will work together with management and user representatives to decide such things
as the rollout plan, phases, order of implementation, and schedule. Project teams must also
coordinate on such issues as where the IP PBXs (call servers) and gateway(s) will go, call
routing configuration, and specifications for other services such as conference bridges,
voicemail, auto-attendants, automatic call directors, media servers, and other accessories and
appliances. And, if the project is a “do-it-yourself” project, the location, hours of operation,
operations procedures, and staffing for Help Desks and Network Operations Centers (NOCs) will
have to be determined. If, however, Help Desk and/or NOC services will be handled by an
outside provider, the location, staffing, and operating procedures will have to be specified in an
SLA. This issue seems a bit more trivial if the network is to be in a single country than if it is a
multinational operation.




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Based on rollout phases and schedule, the data team builds its own plan for the network design to
ensure that needed bandwidth, network configuration for quality of service (QoS), and
connectivity points are provided for the overlay telephony infrastructure. In parallel, the internal
telephony team or systems integrator must build the solution in the lab for the proof of concept
phase that is a mandatory step prior to implementation. If a “turn-key” system is being used
through a telephony services provider, a very close collaboration must be maintained to ensure
the proper interpretation and implementation of user needs and desires prior to a system test.
These steps are addressed in more detail later in the chapter.

Assumptions
First, let’s assume that your organization has an existing IP infrastructure—either a traditional
intranet or a VPN that uses the Internet—and that any desirable changes to the network have
been made prior to, during or after the assessment phase of the project but before the planning
phase. Changes, such as a shift from a private IP network to a MPLS VPN service, the
implementation of QoS mechanisms in the LAN and WAN to support prioritization of voice
traffic, or possibly the correction of bandwidth deficiencies in the network have already been
accomplished. Let’s assume that your existing IP network is one of the following:
   •   Provided and managed completely by an in-house department over telecommunications
       facilities leased from carriers
   •   Completely provided by an outside carrier or service provider (via a VPN or similar
       service)
   •   A hybrid of these two approaches
In addition, let’s assume that the IP network is designed to be a multi-media network including at
least two of the following media types: data, voice, video. If video is not already accommodated
and in use today, it may be necessary at some time in the future given the commodification of
desktop video and emergence of telepresence and high-definition video systems. Baselines exist
for voice quality metrics, such as MOS and other QoS measurements from the assessment effort,
and it will be your job in the network design to assure that the baselines are met or exceeded.
Finally, let’s assume that user satisfaction is an important component of the network design and,
therefore, you will prioritize voice quality and call quality over bandwidth and processing power
of individual systems such as IP phones and gateways. It is understood, of course, that bandwidth
and CPU capability are important contributors to both cost and overall network performance, but
you will err on the side of the user in this balancing act. Let’s also assume, going into this phase
of the project, that a complete inventory and synchronization of the telephony features needed by
users has been done, as described in depth in Chapter 3 and that all parties understand the needs
of the users in regard to replicating existing functions that are necessary (vs. replicating the
dozens of traditional PBX features that aren’t necessary for your organization) as well as their
expectations for new functions and features they will gain as a result of the implementation of
the new system.




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In terms of the actual implementation, let’s further assume that the network will require
attachment to the Public Switched Telephone Network (PSTN) to allow non-IP and off-net calls
and to ensure 911 calling. For this reason, your design will provide gateway access until such a
time that the entire world is packetized voice, an eventuality that lies several years, if not
decades, into the future. The assumption of interconnection to the PSTN will not only assure
connectivity outside the organization but internal connectivity during any phased
implementation. PSTN connectivity also puts the network into a different category from a legal
and regulatory perspective. In the United States, for instance, interconnecting to the public
network can subject VoIP providers to regulations regarding 9-1-1 calls for public safety as well
as compliance with the Communications Assistance to Law Enforcement Act (CALEA), also
known as the “wire tap law,” which has recently been amended to include packet-based networks
in addition to the circuit-based networks that it has always covered. These regulations may
increase enterprise liability especially in light of other governance regulations and bear
consideration by organizations of any size.
Implementation will be phased by a department, geographical area or some other criteria, to be
determined so that you can have a phased implementation. A phased implementation plan is
crucial to success, allowing tight management of migration with an eye toward minimizing
disruption to business operations and isolation of any problems encountered during migration so
that they do not cascade through the network and negatively impact users.

The Voice/Data/Video “Triple Play” and Unified Communications
The term “triple play” refers to the provisioning of voice, data and video services over a single
network. Although the term triple play originated with residential carriers and service providers,
it is also a convenient shorthand for organizational networking. The primary difference between
the residential triple play and organizational triple play is the fact that in the residential
environment there is a much stronger emphasis on video. Video as a multi-media component has
yet to really take off in most organizations (though video is a far more important element in the
business environment that it has been in the past).
Another often heard term is unified communications. Although the term triple play is a carrier
term that is finding increasing acceptance in the service provider and enterprise communities, the
term unified communications (or UC for short) is an enterprise term that is sweeping through the
service provider and carrier markets like wildfire. The two terms are not synonyms—are not
interchangeable. The term triple play really refers to the basic data, voice, and video services
while UC refers to applications that use one, two, or all three of the media. For example, voice
and data are two media. Integrated voicemail, one example of dozens of very exciting UC
services, can make use of both. For example, IPT is used to deliver a spoken message from the
voicemail server to your telephone if you dial in to the voicemail system. However, if you want
to retrieve the voicemail messages and download them to your PC or handheld devices, the
voicemail files are treated as data files and are downloaded using a data-type session and likely a
traditional data protocol such as File Transfer Protocol (FTP) or Trivial File Transfer Protocol
(TFTP).




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Let’s take a closer look at the three “multis” of multi-media—voice, data and video—and break
each down further into three different types or flavors of service: real-time, near real-time and
background or batch. Real-time and near real-time are distinguished from background or batch
by the fact that real-time almost always has a human involved on at least one end of the
communication and very often on both ends. What this means is that delay and delay variation
are far more important in real-time and near real-time than they are in background or batch.
Networks must, therefore, be optimized to accomplish a higher quality of experience for the real-
time and near real-time end-user applications. By contrast, background/batch applications do not
happen in real-time and, in fact, in many cases are not even attended by a human but rather by a
computerized process.

Real-Time
In the case of voice applications, for instance, Table 4.2 shows two sample applications for
voice: VoIP and IPT. Both are interactive, bi-directional telephony applications and both usually
involve a human on both ends of the connection unless, of course, a call goes to voicemail or an
Interactive Voice Response (IVR) system. Such a system may, in fact, allow navigation of a
menu for an automated response, such as an account balance or airline arrival time, or may end
up being a human-to-human call. VoIP differs, however, from IPT in that VoIP is a simpler
voice communication, often part of a Netmeeting or other conference, and IPT includes phone
numbers, signaling, and all the other necessary elements of traditional telephony.
Real-time data applications include interactive chat or Instant Messaging (IM), whereby one user
types on a computer keyboard or phone keypad and the results of the typing appears on a distant
screen; or whiteboard applications, where one either draws on the screen or on a tablet for the
benefit of viewing of the human at the other end of the connection. A perfect example of a real-
time video application is video conferencing, which is really the same idea as two-way voice but
with a moving picture in addition to speech.

    One important multi-media consideration, which is beyond the scope of this guide, is the
    synchronization of the simultaneous voice and video in real-time multi-media conferencing or
    entertainment video applications.




                           Real-Time                     Near Real-Time         Background/Batch
Voice                      VoIP, IPT                     Pre-recorded audio,    Voicemail Synch
                                                         Voicemail, IVR
Data                       Chat, IM, Whiteboard          Browser, FTP, email,   Database Synch,
                                                         SAN, Telemetry         Archiving
Video                      Video Conferencing            IPTV, Training Video   Video Cacheing
                                                         (One way)

Table 4.2: Voice, data and video application examples.




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Near Real-Time
Near real-time applications are represented by such applications as prerecorded audio, browser,
File Transfer Protocol (FTP), traditional and multi-media electronic mail, storage area networks
(SANs), telemetry, IPTV, and one-way training video. Near real-time applications are similar to
real-time applications in that a human is watching and waiting for the information to arrive. They
are dissimilar in that the timing or pacing of information arrival for near real-time does not need
to approach the parameters of low delay and small delay variation required by interactive human
speech and video conferencing.

Background/Batch
Examples of background/batch are voicemail and database synchronization and archiving, and
video caching. No human is explicitly waiting for the completion of the task, so these
applications can have the lowest priority and, therefore, the worst performance characteristics.

Interrelationship of Multi-Media Services
Although there is certainly an inclination to ask each person or department responsible for each
application—voice, data and video—to take charge of their own particular area of the design for
the new network, this is absolutely the wrong approach. What all parties must realize is that any
successful multi-media network is an ensemble performance. In fact, what should really occur is
that all political issues, traditional departmental loyalties and other issues that will stand in the
way of a successful, combined, multi-media network should be addressed directly and
eliminated. This should be done prior to bringing in any carriers, service providers, integrators or
any other third parties. Because the voice/telephony project is usually a matter of integrating a
new application—voice—or new applications—voice and video—into an existing IP
infrastructure, what should occur is that application ambassadors, or ombudsmen, should be
designated to act in the best interests of the users of the specific applications. But, instead of
pitting one group against another with the outcome being a victory for that specific group, the
idea is for ambassadors or ombudsmen to represent their users with the objective of coming up
with the very best possible multi-media solution, given time constraints, budgets, objectives, and
management and user expectations, keeping in mind that many humans will be users of more
than one of the multi-media services.

Application Availability
Application availability is by far one of the most obvious aspects of system operation and the one
that will be judged most harshly by the user community. Application availability will drive many
other design decisions. In fact, consideration of application availability should make many of the
possible design tradeoffs clear and many of the choices obvious.
Application availability has many components, all of which must be considered very carefully
during the design phase. Figure 4.2, for instance, highlights some of the key design
considerations and decision points. Evaluating each one in terms of application availability will
provide guidance to the decision-making process and increase the likelihood of a correct
outcome.




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Figure 4.2: IPT design decision points.

For instance, what type of device should a user have to make and receive calls and how should
the user be connected to the network? Part of the design process is an inventory of who has what
type of phone today. Typically there are many flavors of analog and digital phones among the
user base and this will lead to what type of IP phones they get (good, better, best) as well as
whether their old analog/digital phone will remain. In one IPT survey, 25% of the respondents
plan to continue using analog/digital phones. Careful consideration of this point is important as
it’s a critical factor in cost. There are many ways to categorize users, but for the sake of this
discussion, let’s say that the user is an office worker. If you allow users to keep their current
telephone devices and use a converter such as an analog terminal adapter to convert their existing
signals to the ones used on the new IPT network, you can probably save money over the cost of
purchasing a new IP phone.
But, with telephony service availability in mind, you would probably choose a new IP phone.
Why? For one thing, a new IP phone will most likely be more reliable than an old phone with an
adapter and will also, most likely, provide easier access to a broader range of the capabilities of
the new IPT application. The reason for this is that both a new IP phone and the analog terminal
adapter that can be used to attach the traditional telephone to the new network can have similar
problems, such as out-of-date software or hardware problems, but the IP phones are typically
more intelligent and have more processing power and better diagnostics. How about connection
to the network?




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There may be some cost benefits to connecting to the network through an existing PC, but what
happens if the PC is out of service, could the phone also be unavailable? VoIP phones are sold in
configurations that allows physical connectivity of the phone without the PC needing to be
turned on as well as those that use software on the PC to supplement some or all the functions of
the telephone. Which does your organization have? Or, what about a situation in which the PC is
connected via the phone? Some connections via the phone allow the PC to connect if the phone
is not working and some don’t. Although providing the same cost and convenience advantages,
there is still the question: If the one device connecting the other to the network is out, are they
both out? And what about mobility? A wireless connection would be more flexible but would it
be as reliable? If telephony service availability were of the utmost importance, the decision
would be made to provide a traditional hardwired desk phone.
Should call servers or softswitches be centralized or decentralized? Should they be on your
premises or managed by a service provider or integrator? What about the application availability
impact of different implementations of PSTN gateways? How do application availability
concerns impact the number and placement of gateways? And what is the impact on the
outsourcing of gateways or use of gateway services? What if your organization presently has a
voice VPN? These and other considerations should be made with application availability, and not
just access line availability or similar traditional elements in mind. Don’t forget that users do not
care why they don’t have dial tone, and shouldn’t.
Table 4.3 shows actual and SLA application availability targets for real-time, near real-time and
background/batch applications. The “actual” number is a realistic target and represents a number
that can be used to set expectations of management and users for when they read reports. In
reality, your performance should be better, as the “actual” values usually represent a lower limit
for individual sites or users and the SLA values show aggregate numbers across all individual
sites or users. The SLA application availability numbers are, of course, below 100% and
represent target values for purposes of SLAs after factoring in outages and other conditions
affecting application availability.

   Crafting the “proper” SLA will be addressed in more detail later in this chapter.




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Application            Application            Delivery        Delay             Delay Variation
                       Availability                           (One way)

Real-Time              Actual/SLA             Actual/SLA
Telephony              99.7/99.99%            >90%/99.0       40-80ms (reg)     1-20ms
                                                              100-250ms
                                                              (global)
Data                   99.7/99.99%            100%/99.0       50-100ms (reg)    1-20ms
                                                              150-350ms
                                                              (global)
Video                  99.7/99.99%            >95%/99.0       40-80ms (reg)     1-20ms
                                                              100-250ms
                                                              (global)
Near Real-Time
Audio                  99.5/99.97%            >97%/99.5       50-100ms (reg)    1-20ms
                                                              150-350ms
                                                              (global)
Data                   99.5/99.97%            100%/99.5       60-150ms (reg)    1-20ms
                                                              150-350ms
                                                              (global)
Video                  99.5/99.97%            >95%/99.5       50-100ms (reg)    1-20ms
                                                              150-350ms
                                                              (global)
Background/Batch
Audio                  99.0/99.95%            100%/99.5       80-200ms (reg)    1-20ms
                                                              200-500ms
                                                              (global)
Data                   99.0/99.95%            100%/99.5       80-200ms (reg)    1-20ms
                                                              200-500ms
                                                              (global)
Video                  99.0/99.95%            100%/99.5       80-200ms (reg)    1-20ms
                                                              200-500ms
                                                              (global)

Table 4.3: Multi-media service characteristics.

You will also note that application availability measurements will actually decline as one goes
from real-time, at 99.7% actual and 99.99% SLA, to background. The reason is that in a
contention situation, where available resources are below normal and there is competition for
bandwidth, buffers and processing capacity within the network, real-time applications should
receive highest priority access, near real-time next priority, and background/batch lowest
priority.




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Delivery/Accuracy/Loss
The second consideration of application quality measurements will be a category that can be
referred to as delivery, accuracy or loss. If you see the glass as half full, delivery will be your
choice of terms as it represents the percentage of packets that were sent from the source that
actually make it to the destination. If you see the glass as half empty, your chosen term is loss, as
you would think in terms of the percentage of transmitted packets that never make it to the
destination. If you really don’t think about glasses as being half full or half empty, you might
consider this value as the accuracy of the network in delivering packets. Let’s take a positive
posture and consider the percentage of packets delivered.
As with application availability, you usually see SLA values for telephony and video that are
greater than the actual expected minimum values, except in the case of data and all varieties of
background and batch services for which expected delivery is 100%. The reason for this is that
the later services make use of the Transmission Control Protocol (TCP) as opposed to the
telephony and video’s use of the User Datagram Protocol (UDP) for sending information across
the network. This distinction is important because of the way in which a transmission error is
handled in the network. Basically, TCP will retransmit if errors are encountered and UDP will
not. The reason why the 100% actual number is less than the SLA value, therefore, is because the
SLA value counts the percentage of packets that are sent that arrive on the first attempt. The
100% value may require additional attempts, up to some retry limit, but the 100% number is used
for planning purposes: 100% of sent packets arrive at the destination, and the retransmission time
for any packets that need to be sent more than once due to errors while traversing the network is
adjusted for in the overall delay time. It is noteworthy and important that IPT uses UDP for voice
transmission and in most implementations for signaling as well; therefore, UDP never
retransmits voice or signaling packets. If they are lost, they are lost.




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                                   Error Handling in Packet Networks
Packet networks are not perfect. There is a variety of reasons why packets may be damaged—that is to
say, have the value of their 1 and 0 bits changed—or for packets to be discarded and never arrive at their
intended destinations. This is as true of data packets as it is of packets carrying voice or video, but the
way of handling the problem is a bit different, depending upon the protocol being used to send and
receive the information.




Bad Packets
To assure accurate transmission of the packets, the sending system does a special calculation on the bits
in the packet. The calculation is usually referred to as a Cyclical Redundancy Check (CRC), block check,
Frame Check Sequence (FCS) or Checksum. These are all variations on the same theme used in
different transmission protocols. The results of the special calculation are either placed at the end of the
packet—in the packet’s trailer following the payload bits being transmitted—or in the control bits at the
front of the packet called the header. In either case, they are intended as a special part of the message
that allows the receiving system to determine whether any 1 bits were changed to 0 bits, or vice versa,
which could drastically change the meaning of the information being transmitted. When the packet arrives
at its destination—or in many cases at an intermediate relaying system, such as a router—a calculation
identical to the original calculation is performed again. The new result is compared with the result of the
original calculation stored in the packet by the sender. In the case that the new result and the original are
the same, it is assumed that no bits were changed and the packet is either forwarded or used by the
system. In the case that the new calculation does not match the original calculation, one of three
outcomes is possible. In some cases, the error checking calculation is so sophisticated that it is possible
for the receiving system to determine which bit or bits were changed, up to two bits, and fix them. In other
cases, the receiving system requests the sending system to resend the packet, up to some number of
retransmission attempts as identified by the retry limit. In still other cases, the receiving system discards
the inaccurate packet and lets some other part of the network worry about the missing information.
Lost or Missing Packets
It is also possible that packets may be discarded or get lost along the way for other reasons. If, for
instance, the intermediate routers between the source and destination become overburdened they might
intentionally make a decision to discard packets. This situation, usually referred to as network congestion,
is not uncommon at times even in well-architected and managed networks because building a network to
handle every possible demand is cost prohibitive. In the case of lost packets, some protocols will provide
a sequence number for transmitted packets so that receiving systems can know whether packets are
missing. Some systems, however, do not. In the case that a sequence number is missing, packets can be
recognized and a request can be made that the missing packets be retransmitted, up to a specified retry
limit.




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Which Protocols Do What?
The Internet Protocol (IP) is a protocol common to all applications in IP networks. IP is always used in
conjunction with a higher-layer protocol, which is virtually always either TCP or UDP. IP has an error-
checking mechanism, but only on its header information, and has no sequence number capability. IP
relies on the higher-layer protocols to provide sequencing, if needed. TCP is a richer, more robust
protocol and has both an error-checking mechanism on its header and a sequence number mechanism,
with the tradeoff of more processing and higher bandwidth utilization than UDP. UDP has neither. For
these reasons, TCP is used when information accuracy is vital, such as paychecks and browser info, and
UDP is used for IPT and similar applications.


Delay
Delay figures are cumulative numbers that include all factors that can impact the time it takes a
packet to travel from sender to receiver. In addition to the delay due to the network distance of a
transmission over physical links from sender to receiver, there are added delays for the time that
packets spend in intermediary relaying systems such as switches and routers, known as nodal
delay. Delay is usually a constant factor for any given communication. Reasonable ranges of
values, for planning purposes, are shown in Table 4.3. Both regional (same continent) and global
(different continent) values are shown. The component of delay due to distance is usually fixed
or so very close to fixed for most connections that it can be considered fixed. In addition to
planning, these numbers are equally valid for SLA purposes, though some service providers may
show more accurate numbers for different continent pairs.

Delay Variation or Jitter
Delay can also have a variable element due to a variety of factors, such as how long the packets
spend processing and queuing in intermediate systems on their way from the source to the
destination. As has been noted earlier, for the 100% delivery applications, delay variation might
also be impacted by the one or more retransmissions needed to achieve the 100% delivery. Delay
variation is usually absorbed for data applications and is usually unnoticeable to the end user,
while delay variation in voice applications can be very noticeable and contributes to user
complaints and dissatisfaction. Delay variation is shown in milliseconds and is rarely less than 1
or more than 20. These values may be used for network design as well as for SLAs.
Fixed delay occurs in traditional telephony systems and can be quite noticeable in transoceanic
cable or satellite systems. Delay variation also occurs in traditional telephony but is usually so
negligible due to the high-speed circuit switches that provide the foundation for older networks
that delay variation due to intermediate switching systems can usually be ignored.




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In all the real-time and near real-time applications that Table 4.3 shows, it is assumed that there
is a human watching or listening for the information to arrive, and therefore, in terms of delay,
near real-time is in a close second place behind real-time. In real-time applications, end-to-end
delay must be very short and with as little variation as possible to prevent disruption to the
timing and flow of the human conversation. It is interesting to note, almost contrary to what one
may think, that very long fixed delays, even exceeding a second and a half, have a less negative
impact on MOS values than do much shorter but inconsistent delays. Variable delay is disruptive
to the pace of the conversation but once the speakers adjust to a fixed delay, such as that
experienced over a satellite link, they are much more comfortable speaking. Near real-time
applications, however, can tolerate greater delay, though not the excessive delays possible in
background and batch applications. In the case of near real-time applications, buffers may be
used to optimize throughput without introducing delay that lowers the QoS perceived by the
human user. This is not to say that buffering information for near real-time applications is always
an acceptable solution.

Differentiation, Prioritization and Queue Management
The key to optimizing all applications is to provide the proper mix of resources and handling
within the underlying IP network, and the first step in this optimization is proper differentiation.
By knowing the specific needs of the specific type of traffic contained in each packet,
appropriate and different handling can be applied, increasing the likelihood that all packets will
be handled optimally.
In general, voice applications are less impacted by loss of information than data applications.
Studies have shown that as much as 10% of voice samples can be discarded randomly from a
voice conversation with little or no loss of understanding, although a change of a single bit in a
data transmission can drastically alter the meaning. Due to differentiation when selective
discarding of packets in an overload situation is required, voice packets can be selected and
discarded, instead of data packets, up to some limit near 10%. Data applications, in contrast, are
less affected by delay and delay variation than voice packets. For this reason, in a situation in
which some packets will be delayed and some will be forwarded, differentiation allows voice
packets to be prioritized and sent with lower delay, while data packets may be delayed to offset
the handling of the voice packets.




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There are several mechanisms that can be used in networks to differentiate one type of packet
and its preferred treatment, but the mechanism common to most applications is provided by the
one protocol common to all applications in IP networks, IP. Figure 4.3 shows the format of the
special control bits present in the header portion of the IP packet. The area of most interest to the
current discussion is the Type of Service (ToS) Byte, pronounced “toss byte” and more recently
described as the Differentiated Services (DiffServ) field. The network may also be configured to
use Explicit Congestion Notification (ECN), which uses two bits in the DiffServ field in the IP
header. The ToS bits provide four types of information: the precedence or general priority of the
packet, and within the precedence, normal or low delay, normal or high throughput, and normal
or high reliability. Precedence has eight levels from “routine,” the lowest priority, to “network
control,” the highest priority, and six classifications in between. By setting these bits, an
application such as FTP or IP telephony, can signal the network as to what special treatment the
packet requires. In network design, it is imperative to assure that the ToS bits are set properly
during modeling and simulation exercises and to be sure that desired values find their way into
the final network implementation. It is also important to note that although ToS bits are defined
relatively clearly in “standards,” there are different interpretations of ToS bits from manufacturer
to manufacturer and even from one product line to another, even though the IEEE 802.1p and
802.1q standards clearly and unambiguously describe how to map the Layer 3 DiffServ Code
Points into Layer 2 Virtual LAN labels, and vice versa. There are, therefore, ramifications for
networks in which packets might encounter different interpretations on their path from end to
end.




Figure 4.3: IP header structure.




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                                       Prioritization in Perspective
In the mid 1990s, a major global telecommunications carrier implemented a new multi-media VPN
network for a large multi-national customer. The carrier had sold the customer on using their multi-media
VPN service as opposed to either building it themselves or using a competitor’s network, based upon the
carrier’s superior ATM platform and sophisticated prioritization mechanisms. The customer paid a
premium for certain service classes. Although the carrier actually offered several gradations of class of
service, the new customer settled on three, which they referred to as First, Business and Coach using an
airline analogy that is fairly commonplace in networking. Each class of service had specific characteristics
associated with it, and each one had a different cost per megabit of information transferred. The user
organization accepted this pricing arrangement reluctantly after being sold on the benefits of class of
service and the positive impact it would have on application performance and the resulting end-user
productivity.
After the network was installed and operating, the customer decided to test the differentiation
characteristics to ensure that they were getting what they paid for. They set the appropriate parameters
on a packet generator and tried the First Class traffic. Indeed, they did get the promised performance and
were very pleased with the outcome of the test. They then set the parameters on the packet generator to
simulate Business Class service. The customer was a bit perplexed, and not particularly happy, when
they found that the traffic got the same high performance characteristics as the First Class traffic. One
might think that they should have been happy to get the same performance for a lower price, but they
were unhappy because they felt that the reason for paying the higher price was bogus. They repeated the
test for the Coach Class traffic and got the same results.
What were they to do? There were some among the group that proposed sending all traffic as Coach
Class, which, as their tests had shown, would give them performance equal to that of First Class but at a
Coach Class price. There were others who proposed sending all traffic as Business Class, in case there
was some trick they had missed. At least they would not completely ruin the performance but could save
some money. The problem was that marking all the traffic as any single class, regardless of the treatment
of that class, defeated the purpose of differentiation and made prioritization unnecessary. In the end, they
called the carrier, explained their test, and asked the carrier to explain the lack of difference between the
classes. The carrier explained that their network backbone was actually over-engineered and that the
differentiation, although always present, and the associated prioritization, while always applied, would
only become evident if there were a competition for resources that would only occur during a network
outage in the backbone.
The customer felt, in effect, that what they were paying for was an insurance policy in the unlikely event of
a service-affecting outage in the carrier’s network. And, they really were correct. The customer was able
to renegotiate their contract with a much smaller price difference between First, Business and Coach
classes and is now very happy with their multi-media VPN. It is also noteworthy that the customer has
never had to make any “claims” against their “insurance policy,” but now that they understand the issues
and the cost of the “premiums” is lower, the customer is very pleased with its situation, especially as they
migrate more IP-based voice onto their network.


Voice QoS
QoS is a measurement of the treatment of the packets traversing a network and includes delay,
delay variation, packet loss and network availability. Classification and marking of packets, in an
attempt to assure certain promised levels of quality, is done at or near the network edge and are
controlled by Class of Service rules. The calculation of instantaneous QoS in the network, while
important, provides only one measure of the user’s quality of experience. With its broader
definition, QoE is rapidly becoming the more important consideration for voice services.




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QoE
QoE is defined as a telephony system user’s perception of the quality of the communication
being experienced from both the technical QoS rating and other aspects of the call. QoE takes
into account the cumulative effect of network characteristics on the human speech being
transmitted and received. QoE was first designated as a methodology for assessing the
satisfaction of users of network video services in the mid 1990s and has been applied to voice
since the late 1990s. QoE is a result of a variety of factors and is best measured by way of a
human MOS, which represents the opinion of a panel of human judges, on a five-point scale with
five being the highest. Due to the loss in translating analog speech waves into digital values and
back, in effect, 5.0 MOS has never been achieved. The best Pulse-code modulation (PCM) scores
are in the 4.2 to 4.4 range, depending upon network conditions, but are consistently 4.4 in ideal
lab conditions.
In this early phase in the network design, it is often desirable to convene a panel of judges from
your user community and perform MOS panels to get user buy-in and to assist in the process of
properly setting expectations. This approach is an important tool for setting proper user
expectations and assuring that the system will meet user needs but becomes impractical after
implementation. After implementation, and to a great extent during design and benchmarking,
automated tools will be used to simulate certain types of calls, estimate what the human MOS
would be and help isolate the factors that are impacting voice and call quality. Even using PCM
in the new IP-based system, the voice will sound different, but the idea is that “different is not
bad and, in fact, a properly tuned IP voice system can actually be better than old analog and
digital voice. Care should be taken in the selection of panelists to choose from the broadest range
of job functions in order to optimize the voice coding selection for your specific network.
However, because it is not possible to empanel human judges whenever an opinion on voice
quality in the actual operating network is desirable, a number of measurements and derivative
calculations have been developed (as described later in this chapter) that will form the basis not
only for design but also for the ongoing network measurement and reporting and will be
specified in the SLA. This function must be automated using tools developed specifically for the
monitoring and reporting of voice and call quality issues. Tools such as those used to monitor
router or switch performance, packet loss, delay and delay variation, and system availability only
provide a very general idea of the effect of network activities on voice and call quality.

QoE vs. QoS
Although QoS is an important element for technical issues such as SLA compliance, meeting
users’ QoE expectations is crucial to a successful IP telephony implementation. QoE can be
affected by many factors:
   •   Human factors
   •   Voice encoding/compression
   •   Network issues
   •   Service/feature support




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Human Factors
The biggest factor impacting QoE is user expectations. The users’ prior experience and use of
telephony will play a large part. Are they used to a highly reliable traditional desk phone and are
being asked to use an IP system that occasionally drops calls or a wireless or cordless phone that
does not work at certain locations in the building? Is there a critical path or lifeline application
such as 9-1-1 emergency calling or do they use the phone for revenue generation, such as an
outbound sales call center? People who use the phone for casual conversation or other less stress-
inducing purposes are likely to be less harsh critics of any new system.
Speaker and listener age and gender also impact QoE. Age has been shown, for instance, to
impact MOS. Older listeners tend to give higher MOS while younger listeners’ MOS values tend
to be lower, which is almost counter-intuitive. One would think that those who have been
listening to phone systems longer would be tougher critics than younger folks, but research
shows that older listeners have more highly developed linguistic ability and can get more
information from less communication; therefore, they are more satisfied with the same signal
than younger people.
Gender also impacts MOS. Males tend to give higher MOS and females give generally lower
MOS. The primary reason for this seems to be that higher frequency sounds are more important
to female communication, and these signals are often lost or distorted in the analog- to-digital
conversion process.
Speaker and listener familiarity also plays a role in QoE. The more familiarity between the
speakers, such as a mom and child, the lower the MOS; the less familiarity, such as a first time
caller to a call center, the higher the MOS.
Native language of the speaker is also an important consideration. Although PCM-based systems
get fairly consistent MOS regardless of the native language of the speaker, due to PCM’s analog-
to-digital conversion process, systems based on Code Excited Linear Predictive (CELP) are often
optimized and get lower MOS values if not optimized to the unique sounds inherent to the native
language of speaker. This is because CELP-based systems are voice-optimized and use code
books containing sounds that are matched to codes for transmission. In order for a sound to be
transmitted, it must have a corresponding code.
These considerations are often very difficult to factor into a design, but are critical to user
acceptance and satisfaction and must, at least, be considered. Use of PCM for voice coding and
reducing the number of recoding at gateways and other points in the network will improve the
MOS and the user’s perceived QoE.




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Voice Encoding/Compression
One of the underlying assumptions going into this network design effort was that in situations of
bandwidth versus voice quality, voice quality will prevail. For this reason, IPT systems should be
implemented using the G.711 coding scheme, known to all as PCM. PCM is the standard for
voice coding worldwide in traditional telephony networks and although it has not been
implemented quite as widely in IPT networks, there are a number of desirable characteristics to
recommend it for the job (see Table 4.4).
ITU-T             Coding           Bit Rate        Sample Size      Encoding         Mean Opinion
Standard          Scheme                           (Bits)           Delay (Time)     Score (MOS)
G.711             PCM              64K CBR         8 bits           <1 ms            4.4
G.726             ADPCM            32/24/16K       4/3/2 bits       1 ms             4.2, 3.8, 3.2
                                   CBR
G.728             LC-CELP          16K VBR         40 bits          2 ms             4.2
G.729             CS-ACELP         8K VBR          80 bits          15 ms            4.2
G.723.1           ACELP            5.3K VBR        160 bits         37.5 ms          3.5

Table 4.4: Voice coding options.

First and foremost, the CELP systems are optimized for voice transmission and don’t perform as
well across a wide range of demands. Use of PCM in IPT systems will also sound more natural
to users and result in less user resistance. Another reason is that when voice transits traditional
gateways, as well as other IPT systems, it will not need to be recoded but rather can stay in
native PCM; therefore, it will be less subject to transcoding loss.
The problem in using G.711, however, is with bandwidth. As Table 4.5 shows, with the addition
of packet overhead, G.711 PCM well exceeds the 64K channel rate of traditional telephony. In
the worst case, G.729A at 50 packets per second over ATM uses only 43kbps, only 40% of the
bandwidth of G.729A over ATM, for example with a MOS of 4.2 versus a 4.4 score for PCM.
What is not taken into account in this comparison, however, is degradation of G.729A over
multiple encodings through multiple gateways, which can be substantial in large intra-national
enterprise networks.
Bandwidth             802.1Q       PPP             MLP              Frame-Relay      ATM
Consumption           Ethernet
G.711 at 50pps        93kbps       84kbps          86kbps           84kbps           106kbps
G.711 at 33pps        83kbps       77kbps          78kbps           77kbps           84kbps
G.729A at 50pps       37kbps       28kbps          30kbps           28kbps           43kbps
G.729A at 33pps       27kbps       21kbps          22kbps           21kbps           28kbps

Table 4.5: Sample bandwidth consumption of PCM and ELP codecs.




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Voice Compression

Another consideration is the use of voice compression. It has been shown in actual networks that
voice compression has minimal benefit in packet-based voice systems (IP, Frame Relay, ATM).
Voice compression algorithms usually add nodal processing delay that is unacceptable in very
long distance calls when delay budgets are already near their limits. This is not to say, however,
that packet voice systems should not use protocol compression, such as Van Jacobson header
compression on IP or PPP packets. They should, but for compression to be effective, it must
occur on the digital form of the voice after packetization.
                               Data Compression vs. Voice Compression
Voice compression and data compression vary in some important ways. Voice “compression” is usually
performed on audio in its analog form and often effectively ‘speeds up’ the voice at one end and slows it
down at the other end. This type of voice compression is unsuitable for packet voice applications.
Compression of voice in its digital form is also unsuitable for packet voice, though it is seen in many
implementations, usually as an option. The reason why it is unsuitable is that it works on voice after it is in
its digital form, and there are very few repetitive patterns that can be compressed. What does work with
voice over packet, especially on bandwidth-limited links, is traditional data compression that works on the
underlying protocols that transport the voice content. One example is called Van Jacobson Compression.
Van Jacobson compression can be applied to TCP, IP, PPP and other protocol headers and is described
in RFC 1144. For example, Van Jacobson compression can reduce the normal 40-byte IP packet headers
down to 3 to 4 bytes for the average case. It does so by saving the state of IP connections at both ends of
a link and only sending the differences in the header fields.


Silence Suppression

Silence suppression saves bandwidth, and, when properly tuned for a given network, will yield
any where from 2:1 to 4:1 savings. That is to say that with silence suppression—which removes
silence at the sending end of the voice connection and reinserts it at the receiving end—two to
four more calls can be made using the same bandwidth required for one call without silence
suppression. However, the silence suppression algorithm may negatively impact voice quality if
not properly tuned.

Echo Cancellation

Another consideration, well understood in the traditional voice network but often not even
considered in IPT networks, is echo cancellation. Regardless of what one may hear, echo
cancellation is mandatory in IPT systems. Echo cancellation is not needed in IP phone to IP
phone configurations over an all-IP backbone with no gateways, but this configuration is highly
unlikely. Echo cancellation should occur as close to the endpoints of the call as possible,
preferably in the IP handset, softphone, analog terminal adapter and gateway where outbound IP
calls jump onto the PSTN and inbound calls jump onto the IP network. It would be wise to
ensure that all call servers and gateways that interconnect telephony end systems, be they Analog
Terminal Adapters or IP phones, that do not provide their own echo cancellation can, at least
optionally, provide echo cancellation.




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Tandem Hops/Multiple Encoding

Special attention should be paid in the design to ensure that voice is not recoded more often than
needed. Ideally, communication occurs only between IP phones but, this will be a long time
coming. In the meantime, the number of gateways should be reduced as should the number of
times voice is recoded. Most IPT implementations should consider using PCM on all devices for
voice coding and reducing the number of gateways (except in situations in which bandwidth is
extremely expensive or delays are very long). This will result in the best possible voice and call
quality. This is an issue of importance in both the carrier/service provider environment and in
enterprise implementations and has a profound impact on overall perceived quality of
communication.

Network Issues
Network issues have already been covered earlier in this guide, so let’s simply review the
important aspects of network issues that impact QoE and which, therefore, must be considered in
your design effort:
   •   Delay budget is important and includes all sources of fixed delay. Delay less than 150ms
       one-way is ideal; >300ms will impact the users’ QoE. Delay should be specified in the
       SLA.
   •   Delay variation (aka jitter) is a bigger QoE problem than delay. Delay is fixed and easier
       for the listener to adjust to. Delay variation is difficult to adjust for and causes anxiety
       and stress on the part of listener. Jitter less than 20ms one-way is ideal and should be
       specified in the SLA.
   •   Packet loss >1.5 to 3% is where many human listeners can begin to perceive a
       degradation in voice quality even though the human ear can adjust to packet loss up to
       about 10%, or more in certain circumstances, such as where the speakers are known to
       each other or there is a known call context. Packet loss impact can be reduced by
       shortened voice samples and fewer samples per RTP packet, which is something that the
       organization responsible for programming the IP phones or adapters has control over.
       The end-user or end-user organization may or may not exert control, but this should also
       be addressed in the SLA.
   •   Network availability directly impacts QoE. No dial tone is a BIG problem and causes loss
       of confidence in the system. This needs to be a key component of any SLA.




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Service/Feature Support
In migrating from an old system to a new one, it is critical to ensure that the new system
performs all needed functions of the system being replaced and, to the extent possible, that users
are comfortable with and accept any changes to work procedures required by the new system.
This may seem like an obvious and simple point, but user reluctance to change is one of the
biggest reasons for the failure of IPT systems.
A number of valuable considerations can be made during the network design phase. For instance,
is Voice Band Data (VBD), such as analog modems and fax machines, to be supported in the
new network? If so, it will be necessary to test modems at all speeds and verify all fax groups
work without down-speeding. As simplistic as it may sound, it is necessary to test Dual Tone
Multi-Frequency (DTMF—aka Touch Tone) systems. DTMF is critical for many applications
and not all voice coding algorithms support all digits. An inventory of all calling features, as
described in Chapter 3, must be performed prior to the network design phase to ensure that all-
needed calling features are supported. Supervisory tones and intervention, signals that allow the
network to perform special functions, such as allowing a person to break out of a voicemail
system or make another credit card call, must be supported as well as all call forwarding, routing
and blocking features and any other calling features identified in the inventory. Billing and call
accounting must not be ignored, abandoned or otherwise neglected during the network design
phase. Many organizations use call accounting and billing for a variety of reasons, such as
customer billing, verification, tracking of calls, and allocation of internal costs.

   This topic is covered in more depth later in this chapter.


Voice Metrics and Measurements
There are two basic types of voice quality testing, non-intrusive, often called passive, and
intrusive, often called active. Non-intrusive does not require special test calls and is based on
actual conversations. Intrusive voice quality testing requires a special test call and is based on a
comparison of source signal with signal after transmission.
The most desirable situation is to assemble a panel of judges and let them listen to a call and give
the call a score from 1 to 5, with 5 being the best. After initial selection of products and
technologies that will be put to use in the network, however, this approach is impractical;
therefore, a number of calculated values are generally used by automated testing tools, including
MOS, J-MOS, PSQM/PSQM+, PESQ/PESQ-LQ and R factor. Non-intrusive methods include
MOS, derived MOS, E-model and R value/R factor. Intrusive methods include Perceptual
Analysis Measurement System (PAMS), Perceptual Speech Quality Measure (PSQM) and
Perceptual Evaluation of Speech Quality (PESQ). The following brief review will allow you to
understand their relative importance and value in the network design process.




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MOS
MOS is adopted from traditional telephony. The traditional “opinion” of voice quality in the U.S.
was from a panel of 16 human judges from Central Illinois consisting of 8 men and 8 women
who judged voice quality on a scale from 1 (lowest) to 5 (highest). This type of human-
originated MOS is useful for human user QoE analysis but is useless for SLA development and
network monitoring. Today, MOS is often calculated by a management tool using the QoS
indicators for loss, delay and jitter.

PAMS
PAMS was developed by British Telecom and does not replace MOS, though it was a good first
attempt at an automated MOS estimate. PAMS compares original analog voice waves with
reproduced/transmitted speech using a complex weighting method intended to take into account
characteristics important to the human ear. The scale is from 0 to 6.5, with 0 being “perfect” (no
difference between waves). PAMS values estimate MOS +/- 10 to 20% and therefore do not
provide enough accuracy for use in IPT networks though PAMS values are excellent for
benchmarks and comparisons when both comparison values are calculated using PAMS.

PSQM/PSQM+
PSQM is defined in ITU-T Recommendation P.861. Like PAMS, PSQM and PSQM+ do not
replace MOS, though PSQM more closely estimates MOS than PAMS at +/- 10%. Like PAMS,
PSQM and PSQM+ are excellent for benchmarks and comparisons and represent improvements
on the PAMS algorithm. Like PAMS, the scale is from 0 to 6.5, with 0 being “perfect” (no
difference between waves).

PESQ/PESQ-LQ
PESQ is described in ITU-T Recommendation P.862. Like its predecessors, it does not replace
MOS (+/- 10%), is excellent for benchmarks and comparisons, and is an improvement on the
PSQM algorithm with the same scale from 0 to 6.5, with 0 being “perfect” (no difference
between waves).

E-model/Design Tool
E-model, described in the ITU G.107 standard, is a predictive tool that is excellent for use in
modeling voice quality in a packet network. The use of design tools incorporating the E-model
algorithm combined with network performance data gathered from the live, base network is
strongly recommended. E-model predicts average voice quality using a sophisticated and mature
mathematical model that takes into account delay, delay variation (jitter), packet loss and codec
performance.
The result of the E-model calculation is an R factor/R value that predicts voice quality on a range
from 0 to 100, where 0 indicates lowest voice quality and 100 indicates best quality. R factor E-
model scores are most useful when based upon measured, rather than hypothetical, parameters.
Analysis should, when possible, include Real-Time Protocol (RTP) streams, source and
destination addresses, sequence numbers and jitter profile in order to most accurately predict the
impact of the IP bearer on MOS.
Figure 4.4 shows the relationship between the MOS value and the R value resulting from the E-
model calculation with a description of the user satisfaction level that they represent. A G.107
default value of 94 corresponds with the MOS value of 4.4, which is a target value for many
network design efforts.

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Figure 4.4: R Value compared with MOS.

Call Integrity/Privacy/Security
Migrating voice calls from traditional telephony networks that are inherently secure, particularly
within the backbone of the network, to shared IP networks that are notorious for their insecurity
requires a new level of diligence in call integrity, privacy and security not previously seen in
telephony. The idea that IPT is just another application is one that has been taken to heart by
hackers and cyber-criminals and the archives are beginning to fill with hacker exploits against
VoIP, IPT and VoP. The key points to keep in mind during the design phase are that IP phones
are not traditional phones with a keypad and a ringer. They are special-purpose computers in a
telephone form. They have processors, memory and protocols and can run programs—
increasingly programs written in Java or formatted using eXtensible Markup Language (XML)
and similar languages that are familiar enough for the hacker to compromise and utilize. In all
phases of the network design phase, physical and human security must be considered and, like
any IP networking project, the IPT services and the systems that support them should be subject
to periodic security audits and penetration tests.

Impact on Business Processes
In the best possible case, the migration to an IP-based voice platform will have an immediate and
positive impact on operations, costs may be lower, and users will be ecstatic with the new
applications and Unified Communications tools provided by their wonderful employer. In fact,
the new system will give them bragging rights beyond their wildest dreams. IP phones on their
desk and calls over WiFi phones that attach to their hip or fit in their purse. It can’t get much
better than this! But, alas, the truth is probably a bit more mundane: this, after all, is dial tone we
are talking about. Your reality will, of course, probably be somewhere between these two
extremes, and it will be just as important to capture positive impacts and socialize them within
the organization as it will to identify negatives and quash them.




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Standardizing Positive Impacts
Standardizing positive impacts can be accomplished through two channels: formal training and a
grass-roots effort. Formal training can be via classroom, webinar or written and can be conducted
all at once or phased over time with practice in between. The grass-roots approach can be as
simple as featuring employees who have put the new system to good, company-approved uses to
demonstrate their skills at informal “lunch and learn” sessions or in the context of other
meetings.
Many organizations really do take the “it’s just dial tone, what could be so complicated?”
approach and neglect training, but they do so at their own peril. Absent official training, users
will develop their own skills and habits and may or may not make a positive contribution to
getting the most productivity out of the new investment. “Over-the-cube wall” help should not be
a fall-back to official training.

Avoiding Negative Impacts
Negative impacts can be avoided by having a simple, clear system for reporting problems and a
feedback mechanism to let the users know that the problem is being investigated or has been
resolved. Most organizations choose to use their existing Help Desk and trouble reporting and
tracking system. Any organization considering this approach should be sure that their Help Desk
personnel are properly trained and that their systems are capable of handing the volume and
issues.


Translating Needs to SLAs
At this point, it does not matter whether the organization’s new IPT services will be implemented
in-house, outsourced or some combination. In any event, an SLA will be needed. The SLA will
become, effectively, the voice user’s Bill of Rights and will represent what the user has been
promised, how compliance is measured, and any penalties that might exist for non-compliance.
In a growing number of cases, users are also held accountable for their actions in terms of
number and types of calls, destination, use of gateways into traditional telephony systems and
other actions, and are monitored under the agreement.




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The “Proper” SLA
There are many different types of SLAs in the wide world of networking and telephony, and
many of them are “proper.” In order to be a “proper” SLA, the SLA must not only spell out the
requirements of the carrier or service provider (or internal IT organization) and the attendant
penalties but also specify similar requirements and penalties for the customer organization (or
business unit). It is also common to specify certain benefits or bonuses for exceeding the
requirements set forth in the SLA. These provisions acknowledge the fact that in certain cases
resources are limited and encourage the carrier, service provider or IT organization to apply
those limited resources to the problems of a specific customer or business unit.
An SLA is often an addendum to a service contract or a contract in its own right, and this should
be considered from the outset. Because the document may at some time in its future be evaluated
and interpreted by a judge or arbitrator, attorneys’ technical terms, acronyms and jargon must be
clearly defined or eliminated entirely where doing so does not hurt the meaning of the document.
Nothing should be left to interpretation, diagrams should be provided, and, where formulas are
used, examples should be provided as to their proper application. Terminology should be clear,
crisp and unambiguous, and nothing should be left to chance. If it is understood, for instance,
calculations of service availability assume a maintenance window from midnight Saturday to
2:00 am Sunday, that understanding should be explicitly stated and an example of the service
availability calculation should be provided.

Metrics
The proper SLA may include any number of items, but at a minimum should include specific
details for application availability, packet delivery, delay and delay variation. For numerous
guidelines on specific measurement details, refer back to Table 4.3. Chapter 3 also provided
some guidelines for other possible aspects for a proper SLA, including throughput, call setup and
teardown boundaries, and MOS values.
Availability should include any distinctions for on-net, off-net, wireless access or other
variations as well as allowances for time zone differences, maintenance windows and other
periods of unavailability that are known in advance and should not be included in down-time
calculations. Availability that differs per application should also be noted, as well whether
availability is for a single user, class of user, department, geographic region or network-wide.
Where possible, the availability should be designated for a site too. In very large networks, it is
possible to have a single user or site down for a very long time while hundreds of sites or
individuals are up and running.
Packet delivery should include any special considerations for retransmission protocols, variations
in requirements by application or class of user, geographic area, wireless vs. wireline and so on.
Special consideration may also be given to packet delivery during certain network outages or
other special circumstances.
Delay and delay variation targets in SLAs can be very general but should really be at least at a
continental and inter-continental level for all appropriate continent pairs to which service will be
provided. This type of tiered SLA for delay and delay variation provides more accuracy, better
planning and a clearer understanding of voice impairments when they are reported as well as a
better understanding for remedies under the SLA in case of non-compliance.



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Other items that merit consideration for inclusion in the proper SLA are response times and
Mean Time To Repair (MTTR) commitments—not just “mean time to respond”—for systems
and users under contract and those not under contract and for those within certain distances of
service centers or service personnel. Consideration should also be given for customer-provided
on-site stocking of spare components or use of hot-standby components to lower costs, increase
uptime, and alleviate some of the pressure on service staff for the health of the network.
Time windows for upgrading existing service and for adding new sites to the network should
also be spelled out clearly in the SLA. In all cases, measurements and penalties should be clearly
stated and examples shown so that a “Did you comply with section…”-type questions can be
answered with an unambiguous “yes” or “no” and the veracity of the answer can be known.

Measurement
Measurement of all compliance metrics, to the extent possible, should be automated and have
thresholds for automatic exception reporting that match those found in the actual SLA. Exception
reporting is highly desirable as it cuts down on the total number of alarms and alerts and allows
you to focus only on those issues that represent non-compliance with the SLA. Care should be
taken to align reporting with local time zones and units of measure, as well, to allow the fastest
understanding of SLA compliance reports and most intuitive and localized approach to problem
resolution. Where possible, SLA penalties should also be automatically calculated and credits
applied, as appropriate.
Measurement, specifically for SLA compliance, is a very important area to apply IPT
management tools. The reasons for this are numerous. From a business standpoint, costs are
reduced and consistently can be assured by using an automated tool that is aligned to the
specific, agreed-upon metrics of the SLA. Any manual approach or one based on lower-layer
network statistics from which guesstimates of the impact on voice and call quality are calculated
lack efficiency and consistency. From a technical standpoint, the sheer volume, range and
interaction of measurements needed to develop meaningful reporting and to highlight areas for
problem isolation and troubleshooting are daunting and could easily overwhelm any semi-
automated or manual system. Combining voice-aware or multimedia-aware IPT management
solutions with an exception reporting approach and trend analysis to anticipate problems and
eliminate them before they occur, as opposed to simply trying to mitigate them when they do
occur, should be the objective of every IPT network design and implementation. It is far easier to
design the proper IPT management into the system at the beginning than it is to go back and
retrofit a solution later.

Establishing Feedback Loops and Review Cycles
Feedback loops allow operational data to be gathered from the live network and compared with
SLA requirements. Adjustments then can be made to the network. Early in the roll-out of a
network, after final system test and acceptance, review cycles should be performed frequently,
possibly daily, and a conference call or meeting involving operational personnel and
management should be held. After the network settles down, weekly then monthly meetings
should be conducted. The review meetings are a part of the ongoing network operations and
optimization cycle. Review meetings should be held at least monthly to ensure that the network
is operating as desired.



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Validation of Design
Many organizations prefer to choose a network design and validation tool in the early stages of a
project and to use the input forms and screens of the tool to provide guidance as to the
information that needs to be collected. Others prefer to determine the important characteristics of
their network and then choose a tool that fits their needs. This guide takes the approach that the
choice of the tool should be dictated by the need; therefore, it is at this point in the process to
select the design and simulation tools that will be used in this design phase.
A report card similar to the one provided in Chapter 3 for assessment tools is provided with this
chapter to assist in the choice of network design and validation tools. Very often, the same
provider will have tools for assessment and for design and validation that use the same inputs
and produce results with little additional effort. The version of the Modeling, Measurement,
Monitoring and Management Tools Report Card (4MTRC) for network design and validation
will assist you in the selection process of these very important tools. The report card is a set of
five Microsoft Excel spreadsheets. There are five identical report card spreadsheets that may be
used to evaluate up to five separate 4M tools. The sixth spreadsheet allows you to compare the
tools side by side and support an intelligent selection process. The spreadsheets are locked to
avoid inadvertent changes to cells containing formulas and labels, but a password is provided so
that the spreadsheets may be modified by an organization for its own use. Those knowledgeable
in the inner workings of Excel can take a look and see that there is nothing particularly clever
about the coding. The real value is the knowledge that is embedded in the categories that were
chosen for comparison as well as the weighting factors applied to the different categories. For
users with limited resources, these two factors—the comparison list and associated weighting—
will be an invaluable source of assistance in the selection process; for the more resource-rich
organization, the spreadsheets will provide a convenient starting point for building customized
selection report cards for your organization.
The first project will be to gather together all the Excel spreadsheets, Microsoft Word
documents, Microsoft PowerPoint presentations, handwritten notes, and the appropriate sections
of the SLA that were generated during the early project specification stages and put them into the
network design tool. At this point, the design team can create an operational baseline, validate
the baseline against real-world observations, then commence the process of network tuning. The
cycle will include modifications to the design and service order, the SLA and, possibly,
management and user expectations. It should also be possible to begin tracking costs in real-time
as network elements are acquired.

Network Tuning
After establishing the baselines and validating the model or simulation results against
observations of the real network results, it is possible to perform network tuning, which is an
optimization process in which specific characteristics of the network are systematically modified
to minimize or maximize certain characteristics. The most important measurement in the
network tuning phase will be the E-model R value. Every effort should be made to maximize the
R value while keeping cost in balance, both in terms of network costs and resource utilization.
Before beginning any serious modeling effort, engineers should familiarize themselves with the
tools and just “play around” to gain familiarity, posing hypothetical questions and answering
them, and modifying parameters at will. After the actual network tuning exercise begins, each
step should be documented and only one change should be made to the model at a time.


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Test Bed Architecture
The full network, including all other IP traffic and systems, should be modeled during the
simulation phase. If it is impractical to do so due to the gargantuan size of your network or other
restrictions, at least model all the elements of a sub-network of your network. After the modeling
or simulation on the computer, it will be necessary to build a small test bed in a controlled
laboratory and, after appropriate testing, to move the test to a real environment.
                                        Modeling vs. Simulation
Modeling and simulation are closely aligned and the terms are very often used interchangeably, but while
the objectives are very similar and the results may look the same on the surface, the processes
underlying modeling and simulation are very different. Modeling and simulation are both attempts to
predict performance of network-based services given some set of inputs. Inputs can range from wild,
“seat of the pants” guesses to sophisticated traffic studies. The fundamental difference between modeling
and simulation is that modeling provides a snapshot at a moment in time while simulation is a process
over time with changes to network traffic patterns, call volumes, Class of Service (CoS) requirements, and
QoS and QoE results. The budget-minded very often use an Excel spreadsheet to develop reasonably
accurate snapshots, are emboldened by their results, and use their informal models for actual network
design, validation and optimization. This approach appears savvy and does save costs in the short term,
but even small variances, which are not even really mistakes or miscalculations, can magnify in the real
network and can be very costly, often costing many times the price of a good simulation tool, training and
some consulting to get the project kick-started.

There are several issues that need to be considered and tested based upon your environment.
Considerations and parameters that are common to most IPT projects include:
    •   Network impairments
    •   Timing/synchronization issues
    •   Buffer management
    •   Static/dynamic/adaptive jitter buffers
    •   Broadband bandwidth measurement and calculation
    •   Gateway performance
    •   IP device performance
    •   Feature support and end-to-end operation
    •   Scalability

Network Impairments
Network impairments should be introduced in the test bed architecture to simulate characteristics
that will be encountered in a live network environment. This will be accomplished via software
during the design validation phase and via hardware simulators in the lab and limited live beta
testing. Impairments that should be considered are testing of extremely long distances, excessive
packet loss, excessive delay and delay variation, both individually and combined. All codecs that
may be used in the network should be tested as well as all IPT devices, including gateways and
phones and adapters.




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Timing/Synchronization Issues
Although IP networks are inherently asynchronous systems, sending packets of any length when
the packets are ready to be sent, success with voice applications lies in ensuring that packet play-
out at the end systems closely mimics the synchronous delivery of serial voice streams from
older telephony systems. This steady voice communication can be simulated very effectively,
even in a highly asynchronous IP environment, using buffers and, even better, using the Real
Time Control Protocol (RTCP) along with the Real Time Protocol (RTP). Doing so provides a
closed feedback loop between the receiver and sender based on information about the arrival
statistics of multi-media information being sent. RTCP is not widely used today by enterprises,
or even many service providers and carriers. Many organizations are simply struggling to get IPT
up and running in any condition, but RTCP will be used increasingly as the focus turns to QoS
and QoE. The good news is that most manufacturers are ensuring that RTCP is implemented in
their products and that RTCP will be available when organizations are ready to exploit it.
                                             RTP and RTCP
Multi-media information, specifically voice and video over IP, represent digital information or digitized
analog information that must get across the IP network in a manner and with performance characteristics
that are much more sensitive to delay and delay variation than most of their data counterparts. Multi-
media sessions are established using H.323 or, more predominantly in today’s carrier and service-
provider networks, Session Initiation Protocol (SIP), and use RTP to actually carry the information. The
problem is that RTP alone is blindly sending information into the network with no awareness whatsoever
of its fate. Did the packet arrive? If so, what about timing and delivery? What was the delay? How about
delay variation? If packets are discarded, is it possible to dynamically modify the transmission, including
packet size and content fill, to optimize the connection? If a less-desirable codec is being used for
bandwidth savings, is it possible to use a higher-quality codec in the presence of greater bandwidth
resources? These issues and more are of high value in optimizing the multi-media experience. Though it
is not yet utilized in most networks, there is a solution: RTCP. RTCP is used in conjunction with RTP to
provide feedback on the status of the packets as they arrive at the destination, and there are versions of
RTCP being developed to provide information about the state of the intermediate network and its
performance, as well. Network implementers of IPT should tie RTP statistics back to the SLA.

Buffers can be thought of as “holding tanks” for information as it traverses the network in the
form of packets. Buffers are good in that they provide a staging area in the case of momentary
network overload to keep packets and reduce loss. Buffers are also bad in that improperly
managed buffers can add unnecessary delay to connections, especially time-sensitive multi-
media traffic. During the network design validation and testing phases, it is important to manage
buffers in end systems such as IP phones and gateways as well as in intermediate systems such
as routers. Most routers are not optimized by default for multi-media traffic and often require
software upgrades to accommodate true multi-media traffic. The likelihood of a successful
implementation of IPT is greatly increased if this optimization is done in an earlier, rather than
later, phase.

Buffer Management
Most simulation tools will allow some input of information relating to buffer management, such
as buffer management algorithms, ingress and egress buffer sizes, and internal switching
approaches. Buffers are large areas of memory for storing multiple packets as they transit the
network. In some cases, it is valuable to get buffer management performance results from
manufacturer-specific simulation tools for input into third-party simulation tools.




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Static/Dynamic/Adaptive Jitter Buffers
In addition to the large memory buffers, there are also jitter buffers, which are often as small as
three bits in size, that are used to compensate for the timing variations between the circuit and
the receiving device. There are three basic “flavors” of jitter buffers: static, dynamic, and
adaptive. Static are the least effective for multi-media. They use a single, fixed and
predetermined size of buffer, usually optimized for fragmented data packets for all network
traffic. Static buffers are very common in older routers, especially the lower-end small
office/home office devices. Dynamic buffers are better than static buffers because they
dynamically calculate an optimum buffer size based on the first n packets received. Although
this approach is preferred over static jitter buffers, it is not an optimum solution. Adaptive jitter
buffers represent the state of the art as they can adapt to changing network conditions. Not only
should IPT equipment that employs adaptive jitter buffers be chosen for use in the network, the
use of adaptive jitter buffers should be simulated to the extent allowed by the chosen testing and
validation software tools.

Broadband Bandwidth Measurement and Calculation
Table 4.5 only hints at the wide range of bandwidth utilization between two codecs of very
similar MOS performance, but Tables 4.6 and 4.7 strike closer to the heart of the matter. The
difference between Tables 4.6 and 4.7 is simply the packetization interval. Table 4.6 shows
packets sent every 10ms, resulting in more packets but a smoother flow of voice that is less
susceptible to packet loss. Table 4.7 shows a packetization interval of 30ms. With a 30ms
packetization interval, bandwidth is consumed less quickly because more voice samples are
placed in each packet, thereby reducing the number of overhead bits per sample. However, the
connection is impacted more heavily by packet loss, as the loss of a single packet will result in
the loss of three times as many voice samples.




Table 4.6: IPT bandwidth estimate with 10ms packetization interval (Source: Matthew Michels, Nortel
Networks).




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Table 4.7: IPT bandwidth estimate with 30ms packetization interval (Source: Matthew Michels, Nortel
Networks).

Results of this kind can be used to play “what if games” and to develop realistic bandwidth
estimates for different types of connections.

Gateway, Softswitch, Session Border Controller and IP Device Capacity and
Performance
Gateway, softswitch/call server, session border controller, and IPT device performance and
capacity must also be factored into the mix, especially if the IP devices are not dedicated IP
phones. In addition to performance metrics, capacity of the devices must be evaluated and
simulated. Metrics of importance for this IPT equipment include not only the number of steady-
state connections that are possible to support simultaneously but also the number of Busy Hour
Call Attempts (BHCAs) and resulting Busy Hour Call Connects (BHCCs). IPT systems are just
now beginning to approach the performance of traditional switching systems and very often it is
demand—especially at centralized locations such as PSTN-to-IP gateways and traditional and
IP-enabled call centers—that overload systems resulting in lost calls and potentially lost revenue.




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Testing, Feedback, Network Modification, Testing Loop
Everybody agrees that testing in a laboratory environment, followed by testing in a controlled
semi-live “beta” environment, followed by controlled release and then general release, is a good
idea. It is remarkable how few organizations actually budget the time and personnel needed to do
a thorough job of testing prior to actual live operation. Many organizations simply underestimate
the complexity of voice itself—whether packetized or not—and don’t budget enough time or
resources to fully understand the system until it is under live load and much more difficult to
learn and modify. Organizations will hopefully be willing to learn from the failure of others and
budget appropriate resources for the testing cycle as shown in Figure 4.5.




Figure 4.5: Testing, feedback, network modification, testing loop.




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Testing/Proof of Concept Objectives
There are two main objectives for testing and they will work if the foregoing design effort is
done properly:
   •   Does the theoretical design meet the actual objective?
   •   Do chosen products and vendors meet criteria?
And, in case there are either major or minor flaws in the underlying design, this process is set up
in a loop to allow modifications to be accomplished at this step that will ensure project success.

Feature Support and End-to-End Operation
Although many of the capabilities and systems referred to earlier in this chapter can reasonably
be tested in isolated environments, all features must be tested end to end. This will begin in the
“closed lab” environment, then move into the limited production environment, and then on to the
live environment as testing moves closer to a realistic environment.
The actual feature set that will be used in your environment will vary, so it is very desirable to
establish profiles that include the full set of features a given user profile requires. It is also
desirable to have an actual member of the user community who represents other users with the
same profile validate the feature set and help test the features. This relationship should have been
established well before this point and at this point it should be simply a matter of putting the user
to work in helping with final testing. In fact, it is in final testing where the actual users should be
put to work. The voice or multi-tech engineers should ensure that features are working correctly
and to specifications before actual users are brought in—actual users should be used for
validation and certification and not for initial testing and repair. Do not forget that they are the
liaison with the other members of the user community—the ultimate judges of the system—and
that their opinions on the readiness or non-readiness of the system will filter back quickly.
It might be possible to establish as few as four user profiles that are common to all departments,
though it is more likely that there will be far more profiles for any given situation. Let’s look at
six basic profiles, just to get an idea of the type of feature sets they might need. Although
nowhere near the comprehensive list needed by most organizations, this will provide an example
with some points of comparison and differentiation.




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Profile                              Possible Job Functions             Telephony Features
                                     Office worker, general staff,      Dial tone
                                     supervisor, low-level manager      In-house (extension) dialing
                                                                        Local dialing
Basic Telephony User                                                    Long distance dialing
                                                                        Basic voicemail
                                                                        Call waiting
                                                                        Caller ID
                                     Secretary, telemarketer, inside    Basic telephony user PLUS:
                                     sales person, Help Desk worker     3-way conference calls
Intermediate Telephony User
                                                                        Skills-based call routing
                                                                        Call forwarding
                                     Senior secretary, group admin,     Intermediate user PLUS:
                                     senior telemarketer, senior Help   Enhanced voicemail (add fax
                                     Desk worker                        storage/retrieval and advanced
Advanced Telephony User                                                 messaging and retrieval features)
                                                                        Enhanced call forwarding
                                                                        Multi-party conference calls
                                     Administrative assistant, senior   Advanced user PLUS:
                                     sales person, senior manager       Multimedia conferences (Web
Power Telephony User                                                    plus audio)
                                                                        Enhanced call forwarding with
                                                                        remote access
                                     Senior manager, technician,        Most of the above PLUS:
                                     developer, product marketing       Call routing management and
                                     manager, senior sales manager      follow-me features
Multi-media User                                                        Voice messaging
                                                                        Instant messaging
                                                                        Multi-media conferencing
                                                                        Collaboration
                                     Sales executive, account           Remote telephony
                                     manager, senior sales manager,     Remote data
Road Warrior                         sales VP, sales director,          Remote video
                                     marketing executive
                                                                        Advanced call routing and follow-
                                                                        me features

Table 4.8: Sample user profiles and telephony features.

It is clear that many organizations will have additional requirements, such as various skill levels
of call takers, call taker supervisors, operators, executives, administrative assistants, and similar
skill groups but this basic mapping of user profiles to features will show a way of simplifying
both the testing and implementation.




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Closed Lab Environment
Initial testing is performed in a closed lab environment following the process described in Figure
4.5. The process begins with testing to ensure compliance with the SLA. At this point, the SLA
should be considered unalterable and should be changed only under the most extreme
circumstances as it has become the blueprint that has been agreed to by all parties.
Test results are compared with the SLA. If they match within defined parameters, it is time for
the next step. If not, the network is modified and the testing loop is repeated.

After Hours/Off Hours Familiarization and Benchmarking
The next step is to perform the same set of steps as in the lab environment in the real network
during non-business/non-busy hours. A good practice is to perform this phase of the testing using
a staging area. The staging area is clearly marked out with empty tables surrounded by yellow
tape on the floor. Everything that is taken into the staging area is inventoried so you will know
what tools, software, connectors, test equipment, network equipment, etc. was needed to make
the after-hours test a success. From the list of tools developed in this step, it is possible to put
together an optimized toolkit for actual deployment and to write up a series of implementation
guidelines and field service notes.

Busy Hour Testing in a Live Network
After success in the non-busy hour testing, it is possible to reproduce the testing during the busy
hours in the live network. Many of the principles addressed in Chapter 3 about Network
Assessment apply to the live-network testing phase; it may be useful to revisit the “Tips and
Tricks for A Successful Network Assessment” section in Chapter 3 at this stage. After successful
testing in this controlled environment, it is possible to go to the next step.

Evaluation of Results and Ensuring End-User Acceptance
Results from all prior steps are now evaluated and a “go/no-go” decision is made. If staging and
testing results are not conclusive or compelling changes must be made to the network, testing
and validation should be repeated. Otherwise, it is possible to move on to the next step—
implementation and migration, which will be covered in the next chapter.
One requirement that has been referred to frequently during the first part of this guide as a key to
success is end-user acceptance. This point has been discussed peripherally, but it is time to dig in
deeper and understand what end-user acceptance really means before you can move forward with
the implementation of the telephony project.
Prior chapters discussed the importance of keeping the users involved in the project as it
progresses and in final testing and acceptance. Chapter 3 went into a great deal of detail about
how to inventory the functions of the traditional telephony system to ensure that they are
addressed in the new IPT system. And, by “addressed,” not every function of the traditional
telephony system must be replicated, but rather, any discrepancy must be dealt with and the end
user’s satisfaction must be weighed against other factors.




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Let’s assume the laundry list of features has been implemented and checked by the test
technicians. At this point, it is time for representative users to be brought in for a final
acceptance test: what will they hear, what will they experience and how will they judge the
system?
The very first new thing with which the user/tester will have to deal is the new phone device. If
the new phone is substantially different than the old phone—if, for instance, the new phone has a
screen and softkeys or is a PC-based softphone—quick training and familiarization will be
needed. The familiarization is best done by identifying the way in which traditional functions are
performed on the new phone and then showing any new features that are important to the
specific user’s job. What this means is that if someone uses a phone to only make calls, they do
not need to be shown how to program the phone with XML.
The next step will be for the user/tester to activate the phone. At this point, the user will judge
the phone not only on the time it takes for the dial tone to be heard but also on the actual dial
tone. Although most people who have never traveled outside their own country don’t realize it,
there are actually different dial tones in different parts of the world. The first step to a user’s
maximum comfort level and ultimate acceptance of a new system is for the new system to
operate in key ways like the old system; the first point of comparison will be the dial tone. In
implementations of a new system in a single country, it is important that all new phones present a
dial tone that sounds correct for that country. In multi-national implementations, it is highly
desirable for all phones to present a dial tone that is consistent with what is expected in each
country.
The next issue is the time it takes dial tone to be presented to the caller. This is, quite
conveniently, called delay-to-dial tone or time-to-dial tone and is one of the most critical
elements of a user’s comfort in using the system.
For instance, if the user lifts the receiver and waits too long for the dial tone, they will often,
almost instinctively, put the handset back on the hook and raise it again to make the call. This
time, they get dial tone almost immediately, but it is not a new dial tone. It is the delayed dial
tone from their first call attempt. This will happen because dial tone is not automatic, and it is
generally not a function of the local phone—dial tone is an audible signal to the caller that they
may begin dialing and comes as a result of checking with the switch to determine whether a call
would possibly go through if it were dialed. An alternative would be to play an “all circuits are
busy now” or other similar announcement.
Consider another case. Consider the case in which a caller lifts the receiver and, without waiting
for dial tone, begins to press digits. If they pressed 1-555-984-5800, they might get a message
that says “a one is required before this number” or even “55-984-5800 is an invalid number,
please try again.” What happened? In this case, it is possible that the caller began pressing digits
before the PBX or switch was ready, and the PBX or switch only got some of the digits. Both of
these problems are related to delay-to-dial tone.




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In terms of system design, the time-to-dial tone issue can be exacerbated if an IPT device must
wait on signaling from a remote call server or softswitch before issuing a dial tone, which they
often do. If the call server is located locally and connected by a high-speed LAN with low
utilization, the issue is not much different from a traditional telco arrangement with a local PBX
or switch. However, if the new IPT solution uses a centralized call server that is located in a
distant office, especially across a congested network and one that is prone to discarding packets,
then the Time-to-Dial Tone might be too long, resulting in user dissatisfaction with the new
system.
There is a subtle difference on how and where dial tone is actually produced in some IPT
systems. For example, using a SIP-based system the phone itself may actually generate a dial
tone whenever the receiver is lifted or the speaker button is pushed. This initial dial tone is
considered a ‘comfort’ feature in that the user expects to hear dial tone. Only after the user has
entered some recognized pattern of digits, or pressed a ‘send’ button, does the call server actually
get involved in the process of call setup. Equivalent to ‘delay to dial tone’, in this instance it
would be ‘delay to call (or session) setup’. The impact on the user is the same; delay is unwanted
and is a major factor in call quality and QoE.
After the dial tone is received and dialing is underway, the next issue will be any messages and
tones that a caller will receive while placing the calls. All messages must be in the proper
language and tones must be consistent with what the user expects. This includes any trunk/all
circuits are busy and other progress/proceeding tones or network announcements. The most
important of these will be the busy or ringing/ringback tone that is received. An issue with the
busy or ringing tone involves whether they come from the distant end (remote ringing/busy) or is
generated locally (local ringing/busy).
In the first case, it is assured that the caller will always hear the appropriate busy or ringing tone
for the location being called. This is not always desirable, however, for two reasons. The first
one is that even a busy line or a line that is not answered will use network resources to packetize
the ringing or busy tone and send it to the caller. The second issue is that the caller is not always
familiar with the common tones for all locations they are calling. In this case, the caller may
misinterpret the tones. Both problems are addressed by generating the busy and ringing tones
locally, at the caller’s end. In this case, bandwidth is used only for the signaling packets that tell
the local end that the phone is ringing or busy, and not for the entire duration of the ringing or
busy signal. The second problem is addressed because the locally generated ringing or busy tone
is always the same—it is always what the caller expects and does not matter to where the call is
being placed.
Beyond actually setting up the call is the conversation itself followed by call termination. The
quality of the call will be judged by comparison with what users are used to. It is often a good
idea to establish a set of criteria in advance and to emphasize that “different is not bad” and that
the objective is to provide a working, quality telephony implementation but not, necessarily, to
replicate the old system in every way. Generally speaking, having the user rate voice quality on
the Mean Opinion Score (MOS) scale, with 0 being the worst and 5 the best, similar to the scale
shown in Figure 4.4, is the best idea. An E-model R value of 94, which corresponds to an MOS
of 4.4, will result in about the same level of user satisfaction as the traditional G.107 Pulse-code
modulation. Using MOS and R value also allows correlation of the actual user’s opinion to the
metrics used to model the network at which time judgment can be passed on the suitability of the
modeling tools being realized.


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Fall-Back and Contingency Planning
One point that is often overlooked is that success in one phase or step does not absolutely ensure
success in the next phase or step; it only makes success more likely. Fall-back and contingency
plans should always be a part of the Network Design and Testing phases to ensure that older
systems can still be brought back online in the case of a failure. This does not always mean true
parallel operation but does mean that nothing permanently fatal should be done to the older
system before the operation of the new system has been verified against formal acceptance
criteria that have been signed off on by all parties.

Use of Third-Party Tools
Multimedia networks supporting voice applications are complex organisms requiring specialized
monitoring intelligence. It is also obvious that manufacturer-provided monitoring tools are a part
of the solution but do not take into account the nuances and subtleties of the overall, usually
multi-vendor, environment. However, generic monitoring and management tools are
insufficient—to be truly effective, third-party tools must be constructed with some knowledge of
the individual manufacturer’s systems being maintained in order to provide a system-wide, end-
to-end view of system availability, quality and performance. What is needed is an approach that
makes best use of key aspects of each.
The key areas that monitoring and management tools must cover are:
   •   Pre-deployment simulation and modeling
   •   Manufacturer-specific monitoring and management
   •   Real-time business views
       •       Call detail records
       •       Calls in progress
       •       Delay-to-dial-tone rates
       •       Busy Hour Call Attempts (BHCAs)
       •       Busy Hour Call Completions (BHCCs)
       •       Gateway channel configuration, utilization and loading
   •   Real-time call monitoring
       •       Phone and multi-media device availability and monitoring
       •       Successful vs. failed call completion rates
       •       Delay and delay variation
       •       Packet loss
       •       MOS
       •       Poorly performing components
       •       Service level breaches and SLA compliance



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       •       Real-time interface to Manager of Managers (such as HP OpenView, EMC
               Smarts, IBM Tivoli NetCool)
   •   Summary and exception reporting
       •       Utilization trends over time
       •       Managed devices by company, department and location
   •   Asset monitoring and tracking
       •       Phone registrations and de-registrations
   •   Capacity planning
       •       Incoming and outgoing calls
       •       Loading by dial plan, routing rules and gateway
       •       Bandwidth utilization
       •       Delay and delay variation
       •       Packet loss
       •       Route patterns, utilization and availability
Use of appropriate, qualified, sophisticated third-party tools will allow documentation of results
with consistency and reproducibility and a lack of vendor bias.


Summary
This chapter provided plenty of food for thought as you enter the network design and pre-
deployment testing phase of your IPT project. It included considerations both from the IP-packet
realm and from the realm of traditional telephony. In addition, another report card has been
provided to assist you in the selection of design and validation software tools. This chapter has
enabled you to put to good use a lot of the very critical foundational knowledge introduced in
prior chapters.
Chapter 5 will go to the next phase of the project: implementation of the system you have
benchmarked, validated and optimized, and then, having established a firm foundation, will
begin the task of migrating actual users to the new network.




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Chapter 5: Implementation and Migration
You have reached the phase in the IP Telephony project in which you will learn for the first time
whether your combination of expectation setting, design, and project management is going to be
a success. And, because you are working toward a successful deployment, this chapter will point
out ways that your project could potentially go astray so that you can avoid those possible
pitfalls. To ensure success during the implementation and migration, this chapter will revisit the
global enterprise case study mentioned in Chapter 3 and take a closer look at how those involved
actually implemented their global IP Telephony project, giving you a blueprint for your own
implementation.


External Issues
External issues are those factors over which your project team is unlikely to exert much
influence, as opposed to internal issues, which will be explored later. It remains to be seen in any
given situation which issues will have a more profound positive or negative impact on the overall
project, but both will play a role in the project’s success.

Reducing Negative Impact of Implementation/Migration
There is a recurring theme that must remain in the foreground of your thinking throughout the
project—your project, like any other successful endeavor, is being performed for reasons that are
acceptable to management with the aim of getting results that somehow improve the business of
the organization. This must be true of any project: there are business benefits and goals to be
achieved and when you can compare the list of accomplishments to the list of goals and the
accomplishments meet or exceed your goals, then, and only then, have you completed a
successful project.
End users are a group that will often interfere with, or completely disrupt, a project’s success.
They can range from administrative staff to executives to network power users whose opinions
matter to management. Previous chapters have discussed the importance of getting user
participation and “buy in” early in the project—this consideration must continue through every
step of the process.




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Two of the primary areas for user uprising and resistance during IP Telephony projects are voice
“quality” and the new telephone instrument. Both of these issues can be addressed early in the
project by setting proper expectation levels and then, in the migration and implementation
phases, by getting the user’s acceptance, or sign-off, on the implementation, usually after
training. By telling the user what to expect and handling or circumventing disagreements early in
the project, project management—using criteria they establish, often in conjunction with user
delegates—can truthfully say, “We delivered what we promised.”
In the area of voice quality, the issue is rarely “quality” in any way that an end user can actually
measure, but, rather, the sound of the voice is usually just different from the old phone system or
other phone systems they might use, such as their cellular system. If proper expectations are set
early in the project, using tools such as voice quality demos, this should become a minor issue
and, in fact, might raise the confidence level of users about things they can’t see, or hear. Much
of the value of setting expectations comes from the confidence derived when you actually meet
the expectations.
The difference between the old, reliable, beloved phone instrument and the new-fangled
contraption that is now sitting on the desk after installation and migration is a potential
showstopper for IP Telephony implementations. It is possible to ease the transition through
training, allowing the end users to become enamored with their new phone device, to allow their
opinions to be heard and their needs to be addressed, before they are forced to use the new
device. Do not forget that the phone is the basic business instrument of most workers, and
problems with the phone, perceived or real, can have a huge detrimental impact on
organizational productivity and effectiveness. Absence of a hard “HOLD” button, which is
replaced by a soft “HOLD” button that is displayed only when a call may be put on hold, for
instance, creates a barrier to use. Disappearance of the buttons that light up on a traditional multi-
line phone, a different sounding ringer, and similar issues will all contribute to user
dissatisfaction and the potential derailing of an otherwise well-designed next-generation phone
system.

Business Cycles
Businesses have natural cycles, which should be well understood and respected. This is an area
in which it is useful to consult employees, especially from the IT or traditional telephony staff,
regarding past failures and successes. Some things are more obvious than others. For instance,
most individuals would be aware enough not to replace a phone system for a United States’-
based tax accounting firm during the first two weeks of April because the U.S. Internal Revenue
Service (IRS) requires individual income tax returns to be filed by April 15th. But, would a
similarly aware individual be smart enough to take similar precautions for a tax accounting firm
in Mumbai? They would if they understood the business of the Indian firm and realized that this
company handled a lot of the tax preparation work from the U.S., which is a point that might not
be so obvious.




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Holidays
Holidays often seem ideal for certain upgrades and cutovers because the workers who use the
data and telephone systems are absent. However, it may be equally true that support staff and
Help desks are understaffed during the same timeframe, and working on holidays can make for
unhappy project staff, which may translate into a higher number of mistakes than might be seen
during other times. In addition, in international or multinational support situations, there may be
a holiday in the country housing the Network Operations Center (NOC) while it is business as
usual in the country where the installation or upgrade is being performed.

Coordination
A widely known Vice President for a major insurance company once proclaimed, from hard won
personal knowledge, that any large IT project required the same skill set as a cat herder. This
statement is very true of IT broadly, but especially true for next-generation telephony projects
specifically. Like cats, each individual and department that may be involved in the IP Telephony
project has their own agenda, likes and dislikes, priorities, and motivations. It is the true
professional project manager who can align all individuals and get the desired outcome. Some of
the “tips and tricks” are simple in theory but very difficult in practice. Among the tips and tricks
are understanding that the IP Telephony project is only one of many activities in which
participants will be engaged. Gaining positive participation involves selling the individual on the
benefits they will derive by participating, identifying their role as clearly as possible, and
limiting their involvement to the very minimum investment of time and effort. The list of
potential participants that must be successfully coordinated includes, but is not limited to:
   •   User groups/departments
   •   IT/telephony department(s)
   •   Carrier(s)
   •   Internet Service Provider (ISP)
   •   Managed Service Provider(s)
   •   Management/executive sponsor
Another important consideration involves the extent to which the different entities need to be
coordinated. Must an install tech from the carrier and a security escort from the client
organization appear in exactly the same place at exactly the same time and stay together the
entire time? The answer to this one is likely to be yes. Must the cable installer pull a pair of fiber,
terminate, and test the connection before a new gateway can be installed? Most likely yes, but
there is often some leeway as to how other tasks must be performed, and that flexibility must be
built into the project plan.




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Plans
Several inter-related plans are going to have to be put into place, desk-checked, and tested in the
real world. Each of these plans will be subject to a variety of external and internal influences, as
discussed throughout this chapter. As previously mentioned, internal influences are those that
are, or should be, under the control of the project management team—these are easier to identify
and control than external influences.
For smaller organizations, there will likely be only one variation of each plan—but there should
be at least one of each plan. For larger organizations, there will likely be multiple versions of
each plan that will incorporate variations for a host of factors such as organization, security level,
country, and so on.
To be truly successful in your IP Telephony project, each variation of each plan must have its
own ambassador—an ombudsman to speak on its behalf to ensure that it is getting its fair share
of resources and to ensure its timely and on-budget completion. Each of the plans can be thought
of as a part of the bigger organizational plan and, for that reason, all plans must be successfully
completed for the overall project to be a success.
                                                 Plan Validation
Although it is impossible to see and avoid every problem, it is remarkably easy to identify, plan for, and
avoid the major showstoppers. An important step in the creation of any plan is plan validation.
I utilize five tools to help in the plan validation process. Although it is not necessary to use all tools in
every circumstance, it is useful to have these tools at your disposal and become adept at knowing when
and where to apply each. The tools don’t have formal names or a rigid set of rules, so I will refer to them
as Visualization, Role Play, Naysayer Validation, Non-Expert Validation, and Red Teams. The tools
should be applied in the order in which they are listed, and earlier tools will be applied more often than
later tools in the list. Visualization, for instance, will be applied very often, informally, at the desk or sitting
on the airplane, while a true Red Team is very expensive and time consuming but very often is the best
way to ensure success.
Visualization
The first, and possibly the most valuable tool, is visualization. Visualization involves closing one’s eyes
and stepping through a process visually and thinking about each step. People who use visualization
extensively find that it becomes automatic—and even subconscious—and must keep a pad and pen near
the bed so that they can write down things that they come up with during the night.
You don’t need to sleep at your desk—visualization can be a waking exercise and forms an important part
of plan validation. If you are unknowingly visualizing and coming up with useful and meaningful project
tidbits, your life will be plagued with a daunting collection of those little yellow sticky notes. The value of
your visualization, however, can be greatly enhanced by keeping a project notebook, or even a voice
recorder, handy to be sure that all your knowledge is captured and can be put to work.
The value of project personnel who actually know the practical side of the business, as opposed to simply
the theoretical, is also clear. Practical experience can be increased and the value of designers and
theoretical workers can be improved by assigning them to some lab and, preferably, field responsibilities
as a helper or even an installer.
Role Play
A role play is really a visualization exercise involving two or more persons. In a role play for plan
validation, for instance, visualization is performed and the steps spoken out loud, often following a plan
flowchart or document. The role play is followed through until all procedures are either validated or
corrected as best as it is possible to do at a desk.



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Naysayer Validation
Any organization of any size has one or more naysayers—those individuals who can, and do, find fault
with just about anything. Although often considered an annoyance that must be endured for purposes of
intramural harmony in the organization, naysayers can represent a huge asset to the project team, as
long as they are not the project manager.
After doing everything possible to ensure that there are no flaws in a plan, possibly after visualization and
role play to eliminate the obvious glitches, it is time to submit the plan to naysayer validation. Invite the
naysayer to identify every single possible problem scenario, regardless how far-fetched, document the
problem list and possible outcomes—financial loss, user hostility, lower customer satisfaction, physical
injury, security exposure—and assess the potential cost of each problem and likelihood that it will occur
relative to the cost to fix or avoid the problem. This is a basic risk mitigation exercise that is a crucial part
of any project but is rarely ever done as a formal part of the technical project management of an IP
Telephony project.
Don’t have a naysayer of your own? Go out and borrow one from another department or hire an outside
naysayer/consultant. The good news is that you can return them when you are done gathering their input.
Non-Expert Validation
A desirable next step in plan validation is non-expert validation. Non-expert validation works because
non-experts do not have the same biases or frames of reference that you have relative to your project.
They are in a much better position to evaluate the project from a clear, unbiased perspective. A second
benefit of non-expert validation is that if your non-expert can explain a concept back to you or clearly
present your idea, it must be simple enough to be understood by management and the user community
and will be less likely to be misunderstood. In many cases, the non-expert might be a temporary worker or
a member of the target user community. Regardless of who provides the feedback, non-expert validation
offers substantial value for a very small investment of time and money.
Red Teams
A Red Team is an approach in which you perform a ‘dry run’ of highly complex and mission-critical
presentations, plans, and proposals. This idea originated in the aerospace industry in the late 1950s and
early 1960s. Whenever a project team was going to make a critical presentation or proposal to
management or to their client, they would assemble an impartial Red Team from among other project
teams within the company. In this way, they could get the best possible practice with the least negative
exposure. The team that is being evaluated is known as the Blue Team.
As project teams became less single project/single client-focused during the 1970s and 1980s, the Red
Team approach all but died away and was almost entirely forgotten. That is, until the mega-projects and
customer-focused teams within the telecommunications industry began to emerge in the early 1990s. The
Red Team idea was revived and, although not used as widely now as it was in the 1970s and 1980s, still
represents a very valuable tool for plan validation.




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Implementation Plan
An overall implementation plan must be created for the project. There will be a number of
subsidiary plans that must be put into place, as well, such as the training plan, test/acceptance
plan, migration plan, ongoing operations plan, business/system continuity and contingency plan,
and escalation procedures—all of which become part of the overall implementation plan.

Training Plan
One of the first questions often heard during implementation planning is, “Training for a
phone?,” which is usually followed by the aside, “You pick it up, you get dial tone, and you
dial.” Although user, and management, skepticism of the need for training is widespread,
training is mandatory if your organization is to get the desired benefits from the new system; this
is one of the elements that should be addressed in setting proper expectations early in the project.
The training plan must be divided into two parts: operations training and user training.
Operations training involves training on the actual guts of the system and what is needed to keep
it running. Training is often under-appreciated and under-budgeted because point-and-click
Graphical User Interfaces (GUIs) make a system look like anyone off the street could use
them—anything but the truth. In fact, GUIs alleviate the need to learn complex and arcane
commands but still require operator intelligence and knowledge to keep a system up and running.
In the case that the sophisticated, and often finicky, GUI system is down, operations training will
ensure that staff can use an old-fashioned dial-in modem and a command line; it is always
advisable to have this skillset on staff and readily available.
In situations where operations are outsourced to a carrier, ISP, or Managed Service Provider,
training is still needed, but the focus is somewhat different. In a “do-it-yourself” situation, the
organization must understand trouble reporting, troubleshooting, and problem resolution for
upwards of 90 percent of problems they are likely to encounter. The organization’s personnel
must be at least as capable in operation, upgrade, and repair of the system as the vendor or
manufacturer’s field personnel.
In an outsourced (or partially outsourced) situation, an organization’s staff, though smaller and
more streamlined, must be experts in problem diagnosis and trouble reporting. Although there
will be an initial inclination on the part of the service provider’s staff to waste a lot of time
retracing the steps of the organization’s own people and redoing tests that have already been
done, after the outsource organization is comfortable and confident, time will be saved and
problems will be solved more quickly because the early diagnostic steps of the customer
organization will be relied upon.




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Test/Acceptance Plan
Based upon a combination of the list of end-user and management requirements for the new
phone system and work done in the lab and in early field trials, it will be possible to put together
a series of test and user acceptance plans. Regardless of the actual content, each plan, or segment
of the plan must be written in such a way that a person of sufficient authority can sign off on the
plan. More than one critical delivery has been botched because the person in the office at the
time of the delivery said “I don’t have the authority to sign for that.” This situation must be
avoided.
In large, distributed organizations, one approach that works well is to have special training,
possibly via a Webinar or online training session, for the individuals authorized to sign-off on the
test plans. If the new voice system is going into a branch bank and the branch manager is the
designated person to accept the system, an alternate person should be designated and both
persons should receive training to assure that someone is trained on the acceptance criteria and
procedures. Training a primary and backup person will increase the likelihood of a timely
acceptance of the system and avoid delays in the schedule.
 The training, very simply, might include an overview of the project: why the bank is doing it
and the objectives. Although overall organizational benefits are important, any training or
briefing should also address benefits for the branch bank. Possible problems, fallback
procedures, and contingency plans should be clearly documented, as should the test acceptance
criteria, which should be as easy to verify as possible. Although a bank branch manager can be
asked to ensure that a SIP Invite packet is issued to the SIP server, it is a far better idea to make
“dial tone” and “ability to dial in-branch, main branch, and outside local calls” as four separate
acceptance criteria.
It is also possible to do remote acceptance testing, such as calling to a certain central location
from the remote branch to validate a certain set of procedures. In this case, the branch manager’s
role might simply be to sign off on the amount of time the tech was on site, or there may not be a
role for the branch manager. Experience has shown, however, that project success is greatly
increased if local personnel, especially non-technical management, are involved in acceptance.
Besides being good manners, it can often create a knowledgeable on-site contact at no additional
cost.

Migration Plan
After the design has been modified and tested prior to deployment, a migration plan must be put
into place and be rolled out with discipline and precision. Rarely is a new technology rollout as
visible, or subject to such across-the-board resistance, as with telephony. The migration plan is a
combination of technical changes, user expectation setting, and user satisfaction exercises and
must be executed flawlessly to avoid a negative impact on overall organizational effectiveness.




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Ongoing Operations Plan
After the new system is implemented, and after the users have been migrated, it is time to put
ongoing operational plans into place; however, the plans should be developed, refined, and
validated well before they are needed. Ongoing operations plans should take into account all
aspects of the life cycle of the system. For instance, assume as a starting point that the system has
been properly installed and accepted, the following questions might arise:
   •   How will the operational status quo be maintained?
   •   How will the IP phones be updated and upgraded?
   •   Can new software revisions be downloaded transparently? Automatically?
   •   What if a service or feature does not work? Is there a back-rev plan?
   •   Are there problems in a network with mixed versions of software?
   •   What about hardware refresh?
   •   How often and by whose request or approval may IP phones or VoIP over WiFi devices
       be replaced or upgraded to newer models?
   •   How is bandwidth utilization monitored and managed for applications whose usage is
       growing, such as video?
   •   How do ongoing operations plans tie back to the Service Level Agreement (SLA)?
   •   What software tools are available to automatically monitor SLA compliance?
   •   When SLA breaches occur, are there software tools available to provide alerting,
       troubleshooting, optimization, utilization measurement and capacity planning?
   •   How are SLA breaches handled within the organization’s Help desk and reported to
       management?
All of these topics and more must be addressed in the ongoing operations plan.

Contingency and Business/System Continuity Plan
A contingency plan as well as business and system continuity plans should always be a part of
any project—especially one of the visibility and potential impact on an organization as a IP
Telephony project. Having these plans in place, even if they are never put into play, will impress
management and users with the scope of planning and give them confidence that all possibilities
have been considered. In addition, the plans will be available when and if needed. Periodic drills
and reviews should be performed to ensure that the plans will work if put into action.




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Contingency Plan
Retreat, though not a savory choice, must always be kept as a viable option. A well-planned
retreat has saved many a project and rescued the very valuable “political capital” needed for
credibility within an organization. A contingency plan for terrestrial wired telephony might be as
simple as using cell phones as a fallback if the new IP phones aren’t working, but the procedures
should be clearly spelled out, including details such as access to voicemail during the
transition—from the cell phones if needed.
A fall-back plan for IP Telephony should also include a lab-tested and field-verified approach for
recovering the system to the last-known good configuration as well as a timetable for making a
go/no-go decision for various milestones within the rollout. If, for instance, a system is to be
operational at 8:00 am on Monday morning and the rollback procedure, conservatively, takes 2
hours to execute, a go/no-go decision point should be scheduled no later than 4:00 am on
Monday morning if the new or upgraded system has not yet passed acceptance testing.

    Note that the 4:00 am decision point is actually 4 hours, not 2 hours, earlier than the 8:00 am
    operational time. The extra time allows for some assessment to be done at 4:00 am and extra time in
    case the estimated time for the rollback is off, and to validate proper operation after the rollback
    procedure. In any case, an operational phone system is online at 8:00 am.


Business/System Continuity Plan
Overall business and system continuity plans are inextricably interwoven with the business and
system continuity plans for specific systems or subsystems. An organization, for instance, may
have a plan, and hardware, in place to ensure that a building has backup electrical power via a
generator, for instance, in case of loss of primary power. This alleviates the IP Telephony project
team from responsibility for purchasing and maintaining such a system, but this does not
alleviate responsibility for ensuring access to such a system and for participation in periodic
testing. As an example, power outlets that have Uninterruptible Power Supplies (UPS) and/or
generator backup are often orange. In this example, IP Telephony project procedures should
ensure that all IP phones, or equipment racks providing POE, are plugged into the orange outlets.
Beyond that, procedures should ensure that no additional, non-essential equipment, such as
personal radios or space heaters, are plugged into the designated outlets. The IP Telephony
project team must also ensure that the backup power system can accommodate the load put on
the system by their equipment and participate in periodic testing.
                                         The Value of Reliability
What is the actual cost of an outage or other problem, per event, regardless of duration? What is the
actual cost related to the length of the outage or problem? Are there thresholds at which the cost rises?
And, what are the other impacts, both actual and intangible, that result from an outage?
Every outage, regardless of duration, has a cost simply because it happened. The cost may be employee
confidence in the system, customers who are lost because they switch to a new service, or stock prices
being negatively influenced by accompanying bad press. In addition to one-time costs, the cost may
accumulate over time. As an example, having to pay overtime to get time-sensitive transactions entered
after a computer network is back “on the air” or having to pay overtime, or miss the window of opportunity,
for making telemarketing calls to homeowners during the early evening hours. Or, as is often the case in
financial transactions, mind-boggling sums might be lost if a transaction cannot be performed. Numbers
into the millions of dollars per minute are not uncommon for certain financial systems. Only by knowing
the true cost of a service-impacting outage can you assess the true cost and true value of reliability.


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Escalation Procedure
To ensure that escalation is applied in a positive way to the IP Telephony project, you must be
certain that escalation timetables are clearly documented, locked in, formal, and automatic. That
is to say that the Tech I who is onsite installing the soft switch knows that if he or she is not done
within 4 hours of starting, or by 6:00 pm, or whatever, they are to immediately call and notify the
Tech Manager and to get reinforcements. It is also incumbent upon the organization to ensure
that the reinforcements are available and are aware of their trigger points for automatic
escalation. An underlying consideration is to set a bigger penalty for not performing an
escalation than for performing one, and that the blame game be played only after numerous
similar failures on the part of the individual. And, even then, only in a positive manner aimed at
solving the shortcoming, even though the solution might involve training, retraining, re-
assignment, or termination.


Internal Issues
Now that you have a hint as to the range of potentially project-impacting issues over which you
will have limited, or no, control, let’s turn attention to those items over which you should be able
to exert more control. We can then dive into our case study for a greater appreciation of how
these elements can impact an actual implementation.

Creation of Baseline and User Profiles
During the process of determining the needs and objectives of management and users, a set of
criteria emerged that were the basis for system development and validation and, most likely,
have been incorporated at this point into the SLA. From those requirements, additional input
from management, and a review of any applicable security and/or acceptable use policies, a
baseline user profile and specific profiles for classes of users must be developed. The specifics
will differ based upon your exact system but will include a range of capabilities:
   •   Permission to use outbound long-distance and international calling
   •   Applications that are accessible from different platforms
   •   Amount of system memory and processor capability
   •   Number of calls that may be stored in voicemail
   •   Delivery of voicemail via email
   •   Whether voicemails should be retained on the server




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Bringing All Users to Baseline
After the organizational, or departmental, baselines have been established, it is important to
upgrade any users who do not meet the baseline profile and train them to be sure they are getting
the most benefits from the new service. In some instances, system users may have capabilities or
administrative rights that they will not retain when moving into the new system. The
housecleaning aspect of a move to a new system is important, as users and their privileges can
morph over time, usually gaining capabilities rather than losing them. One thing that should not
be done is to migrate users wholesale while maintaining users’ old rights and privileges. For a
variety of reasons, mostly operational and security related, there should be a minimum number of
possible user profiles and very, very few exceptions. The politics may be tricky and the battles
might go to the highest levels, which is why a policy of minimum number of user profiles with
users receiving the minimum number of rights that will allow them to accomplish their jobs,
should be established from the very beginning.

Additional Capabilities by User Profile
Once baselines are established, additional user capabilities and permissions may be granted for
specific user profiles. A common user profile that meets the needs of a certain class of user is
always preferable to personal profiles that must be maintained and administered resulting in a
nightmarish and unmanageable maize of profiles and permissions.

Security Issues
We must also consider the security of our system and not forget that attackers will find
combinations of capabilities and permissions that give them access and allow them to
compromise the system. For this reason, profiles should be established and reviewed before
being implemented and should be audited and reviewed periodically.
Security of IP Telephony systems is particularly difficult because IP Telephony systems are
susceptible to all of the security vulnerabilities of the systems upon which they run as well as
new vulnerabilities related to IP Telephony generally and manufacturer’s implementations
specifically. For instance, a Denial of Service (DoS) attack that exploits the SYN exchange that
begins each TCP session and is aimed at a server running IP Telephony is not a IP Telephony
security problem, per se, but will, albeit indirectly put IP Telephony out of service. On the other
hand, IP Telephony may be attacked directly, for instance, by exploiting weaknesses in the way
that Uniform Resource Identifiers are parsed, which will compromise the IP Telephony service
but will not impact underlying protocol layers or other services running on the same server.
Security is so important that it should be carefully considered at each step of planning and
deployment in conjunction with the internal security department and, as needed, outside security
professionals and consultants.




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Liaison with Security Department
It is important early-on to establish a liaison with the internal and/or service provider security
departments, to seek their input, guidance, and collaboration and to keep them aware of what you
are doing. Very often, folks from the security department have as much knowledge about your
needs and applications as you do about their responsibilities. For this reason, it is a good idea to
start any meetings with a briefing about what you are trying to accomplish and why and to
request and expect a similar briefing from the security people. When working with the security
people, you can increase the likelihood of success (that is, that your efforts will not be blocked or
thwarted in any meaningful way) by gently reminding the security folks charged with your
protection that they are in the role of providing a valuable service to you and that you are the
customer, not the other way around. In many cases, the very best security people have spent at
least a part of their careers in situations in which running the show has had real and serious
consequences in terms of financial loss or national security. It is a bit different in the commercial
world, but the security viewpoint should always be considered and, in the case of a tie, a debate
can be held to allow an upper-level manager to make a policy decision.

Firewalls and Proxy Server Issues
Firewalls, proxy servers, intrusion detection systems (IDSs), and similar tools are the stock-in-
trade of the security business, and the value of properly implemented security systems cannot be
overstated. However, two issues come into play when discussing security as it relates to your IP
Telephony project. The first issue is improperly implemented systems that degrade, disrupt, or
outright stop VoIP services. The second issue is with properly implemented systems that do not
allow VoIP.
In the case of firewalls, for instance, most firewalls today do not make simple “pass/no pass”
decisions based upon simple criteria such as source and destination IP addresses. Today’s
firewalls often perform sophisticated analyses called heuristics, which protect against as-yet-
unknown problems, or stateful inspection, which does sophisticated analysis of the interplay of
multiple levels, or layers, of protocols. In the best case, this process only adds some delay, but in
the worst case, these systems might cause sessions to be dropped or blocked entirely.
Proxy servers are another type of system that can often get in the way of VoIP. A proxy server
acts on behalf of a user. To protect a user from attack and maintain anonymity, a user’s browser,
for instance, might make a request of a proxy server to do some action on its behalf, to retrieve a
Web page, for instance. In this case, the user connects to the proxy server, and the proxy server
connects to the Web server. Each of the Web sessions requires additional resources on the proxy
server, and most proxy servers are sized and configured based upon the number of simultaneous
sessions. If, in your new VoIP service, VoIP calls were to replace browser sessions on a 1-to-1
basis, you would probably be OK, proxy-server wise. But it is likely that the sum of
simultaneous browser and VoIP sessions will be greater than browser sessions only before
implementation of the new system; thus, you run the risk, especially during the busiest times, of
overloading the proxy servers. Security systems—be they firewalls, proxy servers, or IDSs—
should always be a part of system testing.




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IP Addresses and Port Numbers
Most prevalent applications on IP networks, such as browsers, file transfer, and email, use well-
known port numbers to communicate and, therefore, may be readily identified by firewalls and
other security systems. IDSs and stateful inspection services can also determine fairly easily
whether the behaviors of these applications are proper. A problem with VoIP is that it does not
use well-known port numbers, but rather, registered port numbers, and requires special
configurations on firewalls and other security systems to handle different “flavors” of VoIP. By
different flavors, I mean VoIP originated from different manufacturers. This is especially
problematic in environments in which VoIP from multiple manufacturers might be employed,
such as an environment where employees choose the VoIP system they will use, or a system that
is shared with clients. This is far less problematic in a controlled environment where the
organization controls all the parameters, but it is possible to have both situations.
                            IP Addresses, Port Numbers, Sockets, and VoIP
IP addresses plus Layer 4 (TCP/UDP) port numbers equal socket numbers. Sockets uniquely identify the
endpoint of a connection in the network at a moment in time. If you think about a street addressing
analogy, IP addresses get you to the building (that is, server or other IP host, such as a user PC), and the
port number actually gets you to the office in the building (the application). All port numbers fall into one of
three categories: well-known, registered, or ethereal, which means temporary. Most widely used
applications have well-known port numbers, which makes life easier for administrative and security use. If
a widely used application is authorized for the system, its well-known port identified by its well-known port
number is either left “open” or opened provisionally for information packets to flow through. If a given
application is not authorized, or not authorized for the IP addresses, or some other criteria, then it is
closed and any attempt to go through that port is seen as a potential security violation. The problem with
VoIP is that it was not originally included in the list of common, prevalent applications and, therefore, does
not have a well-known port number. It is too late to retrofit it into IPv4, the current version of IP being used
most widely in the US today. VoIP applications, therefore, must use registered port numbers, which vary
by manufacturer, protocol, and software publisher.

Consider, for instance, a situation in which a large hotel chain provides an IP network that is
used by both guests and employees. In this case, the IP network should be segregated for security
and performance reasons because employees will use the carefully chosen and implemented
VoIP solution internally and hotel guests will use anything they want in an ad hoc fashion and
may, or may not, get the voice quality they need in the open, “best effort” world of the
dynamically shared private intranet/public Internet connection.




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NAT and Internal and External IP Addresses
IP addresses come in two basic flavors—registered and unregistered. The registration process
ensures that no two user organizations will have the same IP addresses on the Internet because
the registration process provides unique sets, or blocks, of IP addresses. This does not mean that
organizations cannot use their own IP address assignments on their own, private IP networks:
they can. In addition, there is not a problem with this situation as long as the private IP network,
with its own set of IP addresses, is not connected to the Internet. If it is desirable to connect the
two networks with conflicting IP addresses, this must be accomplished using a device, such as a
router, that performs a special translation function of the internal, potentially duplicate, IP
addresses to external, unique IP addresses. This function is called Network Address Translation
(NAT). In many cases, internal users do not need access to the Internet, and there are many more
internal users than those crossing the NAT boundary. The importance of NAT is clear, but the
reason why it deserves special consideration in VoIP systems is that the use of NAT often causes
a disconnect in the mapping of the underlying IP address to the higher-level phone number or
User Resource Identifier (URI) used by VoIP.
The solution to the NAT problem lies in either ensuring that all “internal” IP addresses are
registered, or using a separate IP network internally for VoIP. Assuring that all addresses are
registered eliminates NAT, which is not practical for many organizations; Using a separate IP
network internally for VoIP simplifies the issues of managing VoIP but raises costs and network
complexity and completely kills the multimedia, triple play network concept. It is also possible
to place servers, such as VoIP or SIP proxy servers, strategically on the Internet/registered side
of the NAT function so that remapping is done after the relationship between the IP address and
phone number or URI is not crucial. Another approach is to “bury” the IP address relationships
within lower-layer transport systems, such as Virtual LANS (VLANS) or Virtual Private
Networks (VPNs).

Internal and External Phone Number Mapping
The foregoing discussion of the use of internal IP addresses that were assigned without reference
to standards or registered assignment and the potential to encounter duplicate numbers is not
limited to IP addresses. Many organizations, especially the world’s largest, have the same issue
with traditional telephone numbers. The problem is solved in much the same way as IP
addresses, where internal phone numbers are used internally but are mapped to external numbers
either through PBXs, IP PBXs, or some other gateway function, on their way into the public
switched telephone network (PSTN). This is a large issue and should be part of a serious
discussion prior to implementation.




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Flash Cut vs. Parallel Operation
The next consideration of migration is whether a system should be flash cut—always, of course,
with a rollback plan in place to return to prior operational levels should a disaster of some type
occur during the cutover—or allowed to operate in parallel for some period of time. The flash cut
can be a bit of a bold move and disconcerting for the end users, but operationally, it is usually a
lot cleaner than having two systems in place for any period of time. A happy middle ground is to
do a phased operation, possibly in two or three steps, in which an individual phone user is flash
cut, gaining their new IP-based phone and losing their old phone all in one fluid motion, but
there will still be old phones in the office area for some period of time until the final set of users
are cut over. If users are properly prepared and trained, this type of phased implementation can
provide a psychological safety net for the users but keep costs and the complexity of the interim
network operations under control.

Migration Priority and Order
The order in which migration will occur and the priority of users within the list should be the
cause of much debate. Invariably, management will either want the highest ranking people to go
first because they are the most important and should get these shiny new tools before the masses,
or they will want the highest ranking people to go last because any problems in implementing the
new technology will have the biggest negative impact on them. There is rarely a middle ground.
What might make more sense is to migrate users who have been involved in system selection and
implementation in the first wave. These “friendlies” are more likely to provide insightful
commentary and useful observations in the early phases as opposed to creating political firefights
that must be fought before migration progresses.
The earliest migrations should be as close geographically to the source of technical support as
possible and, furthermore, capable of receiving overnight packages. This may mean the
headquarters office, but it does not mean that the CEO’s office should be migrated first.

Involvement of End Users in Acceptance
The importance of involvement of the end users in the implementation and migration of the IP
Telephony system has been stressed throughout the project. It is no more or less important at the
implementation and migration phase. Continue to work with the end-user community, generate
good “word of mouth,” work to address all issues quickly and thoroughly, and be aware that the
performance of your network very well might begin to change as more and more packet
telephony users are added to the network. For this reason, keep your feelers out in the user
community, get early insight into possible problems, and use your friendlies to detect problems
early, before they are fatal.




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IP Telephony Implementation and Migration Case Study
At this point, you know a lot. When you combine the previous four chapters with this chapter
and your own broad personal knowledge gained from reading, learning, and, quite possibly, your
own experiences at designing, implementing, and using packet voice, it adds up to a lot of
knowledge. The question becomes: “How can you customize your knowledge to your specific
needs to allow you to directly apply what you know to ensure the success of your IP Telephony
initiative?”. The reality is that level of customization requires a lot of interaction and
collaboration that is not possible in the current medium. So, let’s do the next best thing—explore
how the many hundreds of different factors that have already been discussed were combined to
create an implementation and migration architecture—a blueprint—for a global company. The
hope is that you will see patterns and approaches that will be helpful to you and that in this way,
you can start down the road of applying the theoretical to the practical. Hopefully, this case study
will give you more insight and food for thought.

    To fully understand the case study, take a look at the Request for Proposal (RFP) used by the
    company that is the subject of this case study. You might even find the document useful as a starting
    point from which you may add, change, and delete to create your own RFP. A generalized version of
    the RFP, which has been modified to be more general as well as to hide the identity of the company
    is available from at http://nexus.realtimepublishers.com/content/DGSDVIP_RFP.doc.

The following case study will have many elements that can be applied to you, or at least, using
your imagination, you can put yourself in the action. If you are not a global giant, you might not
need to consider the legal and regulatory environment in many countries—simply in your own
country. If you are a carrier or service provider, maybe you can forego the selection of the
transport network and use your own, or maybe you should consider using the transport services
of another company for your off-net calls. Maybe you don’t want to manage your old gear as a
part of your plan; then don’t, but at least take a look and see how things came together in this
real-life scenario.
                                            The Three Tracks
There is an oft-told story about three blind people and their description of an elephant. The story goes
something like this: Three blind people go to the zoo and are given the opportunity to touch an elephant.
Having never seen an elephant, as all of them were blind from birth, they didn’t really know what to
expect, but having felt many other things, they thought they did have a basis for their observations of the
elephant. After they had all had a chance to touch the elephant, they were asked to describe the beast.
One person, having felt the trunk, said that an elephant was much like a snake, only much bigger. The
second person, having grasped the tail, described the elephant as being like a huge broom, swishing this
way and that. The third impression was as a result of handling the elephant’s leg; the third person
described an elephant as being much like a large tree.
The reason for pausing at this point to share this story is that it illustrates a common situation with IP
Telephony projects—although most folks have never encountered this IP Telephony animal before, they
all bring prior conceptions, often misconceptions, about it, and they all tend to view it from their own
perspective. The participants in your project will often see what they see based upon their experience and
job status. It will be the job of the project manager to make this specialization work for you, as was the
case in the following case study.




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Three different groups are involved in the implementation of the global IP Telephony project:
they are the business group, the technical group, and the legal and regulatory group. The business
group sees the business benefits and pitfalls of the global IP Telephony project; they confirm the
Return on Investment (ROI) calculation and the tax ramifications of equipment depreciation and
worry about factors such as Moore’s Law. The technical group is concerned about the protocols,
devices, and deployment. They concern themselves with considerations such as Quality of
Service (QoS) in a multimedia network, network security, and survivability of systems in often
harsh field conditions. The legal department is worried about all things legal and regulatory:
contracts, local regulation, corporate governance, and similar worries.
Instead of re-educating everyone in new ways of doing things, the decision was made early-on to
harness the natural order of things and design the project blueprint around three tracks: business,
technical, and legal/regulatory, and to closely manage the points of interaction between the
tracks. Although it is the job of the overall project managers to see the 50,000-foot view, the
workers are best focused on their own area of specialty. Let’s explore the project based upon
these three tracks as identified in Figure 5.1.




Figure 5.1: IP Telephony project roadmap.




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A kickoff meeting was held with representatives of all groups present to allow project
participants to meet each other and to get a sense of the “big picture,” which was, literally, a big
picture. Each group received a master E-size copy of their organization’s master project map (see
Figure 5.1), measuring nearly 3 feet by 4 feet. It was agreed that the master blueprint would
remain unchanged, but it was the job of each track to define the specific, detailed, approach for
each of the boxes represented on the master project map. The project map, and all subsequent
detailed drawings from the tracks, are done in Visio and each track was allowed to use any other
tools of their own choice for managing their part of the project, but Visio was the common way
of representing project relationships, eventually yielding several hundred detailed diagrams.

Business
The initial job of the business team was to assess management’s and user’s needs and desires for
the new system—both in terms of actual functionality and business impact. In many
organizations, this task would have been relegated to the technical team, but management in this
case did not want their new telephony initiative to be based upon what was technically possible
but rather wanted to apply available and immediately emerging technology to specific business
needs and business outcomes.
The management and user capabilities audit determined all current telephony capabilities that
were mandatory and those that were desirable in the next-generation telephony project. Audits
included user surveys, Postal Telephone & Telegraph (PTT), carrier and service provider record
audits and billing reviews, and switch and PBX audits.
One noteworthy outcome of the audit process was a realization of how much attrition of
expertise about the older systems there had been. In some cases, retired employees were brought
back as consultants to gather configuration details and call detail record information from older
vintage switches and Private Branch Exchanges (PBXs)—functions rarely performed on some of
the older gear in the network. This realization, in and of its self, was a wake up call for
management.

ROI and TCO
An important consideration in the management review was the financial aspects of the project.
Although the details are confidential, it is possible to share some broad observations. For
instance, since the mid-1990s, the company had been realizing approximately 20 percent year-
over-year savings in their voice and data networking areas with increasing costs in their
telemetry and video areas. The savings represents two factors: the first factor is riding an aging
investment in voice and data infrastructure on the back side of its depreciation curve and the
second factor is demanding cost concessions from vendors and service providers until there was
almost no margin left. It was clear that the new IP Telephony initiative was going to put voice
and data infrastructure into the increasing cost area, at least for the foreseeable future, and that
vendors and service providers could give little more in discounts.




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In the first case, all benefits of the new network needed to be clearly identified to upper
management and metrics devised to show progress. This was accomplished by a senior telecom
and networking management taskforce who approached the task as if their survival depended
upon it—and it did. Spending more without producing tangible results was out of the question. In
fact, the senior vice president in charge of the taskforce commented that “there is no such thing
as strategic in our world: it is purely tactical. Tactical means that you can show hard cost savings
or other benefits and strategic means that you can’t. We have become purely tactical.”
The second area was addressed by taking a different approach to vendors and service providers.
The relationships were redefined to foster a more collaborative engagement, which allowed the
company to get more value from a relationship with fewer key vendors, carriers, and service
providers, and to pay a bit more money to each. This approach also leveraged the relationships to
reduce headcount, especially in very remote areas, in what might be called a very closely
managed near-outsourcing arrangement.
In terms of actual ROI and Total Cost of Ownership (TCO) targets, the ROI and TCO in this
case study represent nearly unrealistic goals for most companies. Most companies cannot
purchase products and services on the scale of the case study company. One of the reasons that
scale matters is that very high costs in certain areas can be averaged out by lower costs in other
areas. This averaging effect is easier with tens of thousands of users. On the other side of the
coin, however, even slightly higher across the board costs will soon spiral out of control if not
contained.
The typical ROI for the company is 12 months, but because of the scale of the changes that
needed to occur, the ROI was calculated at nearly 14 months—considered good by many
organizations. TCO, not including incidental expenses incurred on an individual basis such as
Internet access service charges at hotels for traveling employees, is around $38/person/month for
the VoIP portion of the network only. The network includes other IP Telephony technologies,
such as Voice over ATM and Voice over Frame Relay, with higher TCOs. The company is
migrating away from aging traditional and non-VoIP systems as quickly as possible, convinced
of the financial and operational benefits of a single IP infrastructure company-wide.

Establish Site Profiles
The broad user and management capabilities audits were combined for management review and
the result was a master set of capabilities and target business objectives that were compiled into
MS Excel spreadsheets. From the spreadsheets, user and management representatives from each
of several thousand sites and locations, such as small office/home office and mobile workers,
were able to identify their specific needs. The spreadsheet results were used to develop a
minimum number of profiles.
One of the inputs to the process of developing the client site profiles was the client country
legal/regulatory profile. For instance, in certain countries in which the company operates, it was
against regulations, or in some cases flat-out illegal, to do VoIP. In other cases VoIP is allowed,
but there are restrictions on encryption of information, which for this company, is a very
desirable aspect of VoIP that will keep information more secure than over traditional telephone
lines. Although this step will not be needed for many IP Telephony projects, it is at least worthy
of consideration. Output from the site profile creation was used as input to the development of
private network, IP VPN, and completely outsourced Layer 2 VPN strategies in the technical
track.

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Rough-Order-of-Magnitude Pricing
The next step in the business track was to take configuration inputs, bandwidth requirements, and
maintenance and support information from the technical track for three different network
strategies: a completely private multimedia IP network, with the company basically acting as its
own internal carrier; an IP VPN strategy, which completely outsourced the IP VPN with very
strict SLA requirements; and a completely outsourced global Layer 2 VPN network, with
similarly strict SLA requirements. Comprehensive cost and pricing models were developed for
each option and compared with target ROI and TCO objectives.

Management Review and Site Prioritization
Various scenarios were presented to management based upon which areas of the company, and
even specific offices and locations, would achieve the greatest benefits from a new telephony
system. The final result of this step is that “green field” locations, that is to say new business
locations coming online would, in most situations, be the first recipients of the new system, if a
business need existed. If a clear business need did not exist, the green field sites would receive
older telephony equipment that was displaced from existing sites that were upgraded.
Older equipment displaced as a result of a migration would be returned to a surplus equipment
inventory and would be used to extend the life of offices that had older telephony gear in place
and did not need or did not yet need the new packet voice system or were in a country with legal
or regulatory restrictions on the use of VoIP. The surplus equipment management process will be
described in more depth shortly.

Initial Migration/Green Field Plan and Renegotiation of Favorable Contracts
The next step was to translate management’s needs into an actionable plan for the sites, order of
migration, and action identified by management. This process resulted in the initial migration
and green field plans with green field opportunities taking precedence over migration where
there was a conflict for resources.
Armed with actual requirements and funding approval for the new sites and a clear indication of
the needs of existing sites for the new systems, vendors and bandwidth providers were engaged
in a renegotiation of existing contracts or negotiation of new contracts for green field sites. This
process was part of a feedback loop that allowed management to reprioritize based upon
favorable or unfavorable outcomes from the renegotiation process.

Joint Migration/Green Field Planning Effort
With a clear set of tasks, based upon business imperatives and management direction, it was now
time for a joint meeting between the business team and the technical team to determine specific
actions, dates, times, and responsibilities. The technical team had prepared templates for the
actions needed for each site profile and could combine them, clear up any discrepancies, and
create a task list, time line, and other project management tools, and integrate the operation into
the overall schedule.




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Contracts and Delivery
Contracts and delivery, although processed by the legal team, were managed jointly by the
technical team and business team and were a key output from the joint migration/green field
planning effort. Contract specifications had to align with key project dates and delivery of goods
and turn ups of services had to be identified and provided back to the legal team to go to
purchasing fulfillment and accounts payable. Although these details are often not treated
seriously or taken into account, there are possible operational circumstances; consider, for
instance, a location whose ISP contract is not paid and they lose their service, or the impact on
corporate profitability if fixed assets are not received and accounted for properly. Although
seemingly financial functions of little interest to the technical team, the reverse is actually true:
they are of the greatest importance and should be taken into account at each step of the project.
One of the outcomes of the business track was the surplus equipment management effort.

Inventory and Fixed Asset Accounting and Disposition of Old Equipment
One of the primary reasons for a migration to a next generation of telephony system—any next-
generation system—for many of the company sites had nothing to do with advanced new
capabilities. The real reason was manufacturer discontinuation of older systems and models and
the diminishing availability and rising cost of components for maintenance, repair, and
expansion of their existing, older vintage, telephony systems. A solution to this that extended the
lifetime and created bottom line benefits of the older systems was to take equipment that had
been decommissioned during the implementation of the new system and send it to one of three
regional warehouses for refurbishing and placement in service, either for repair or upgrades, in
other locations that were still using the older equipment.
This approach took what might otherwise have constituted garbage and repurposed it in a unique
way that helped the company achieve its overall business objectives. The inventory management
and fixed asset accounting for each of the components moving into and out of the regional
refurbishment and warehouse centers was managed in the same way as all other assets of the
company and will be until their ultimate disposal.

Implementation of Changes to Business Operations
There are other important considerations in the business track that are noteworthy and interesting
that appear in subsequent detailed diagrams but that are not specifically enumerated in the
project map. They are:
   •   Training
   •   Dealing with user reluctance
   •   Standardization
   •   Impact assessment
   •   Changes to processes
   •   Staff implications




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Training of users was accomplished via a multi-tiered approach. Representatives from each
office, plant, remote location, department, and so on were trained on the system operation and
any special capabilities specific to their business function. These representative were the same
people who had been involved in the needs assessment, design and implementation/migration
planning all along—being, in effect, ambassadors for the new system and, therefore, duly
enthusiastic about its success. The representatives trained other users in both formal and informal
settings and were available as liaisons to customer support, creating a frontline of support that
was able to answer many of the basic questions and avoid too big an additional demand on the
Help desk.
There was, of course, initial user reluctance, in many cases being somewhat strenuous. Most
users eventually became familiar with the new systems, though many never completely got over
their anxiety and were likely to blame the new phone system for any mishap.
Standardization of business systems and practices is somewhat different than the standardization
discussed in the technical track. In the case of standardization of business practices, use of a new,
single, consistent IP Telephony system organization-wide, regardless of the individual features
or enhancements for a given operational area, allowed for a simplification of work processes,
better transportability of employees between offices and job functions, and an overall
organization-wide focus on a single system—all of which resulted in both tangible productivity
improvements and intangible user satisfaction with the system.
Impact assessment was accomplished via a series of formal user surveys, getting higher marks
from users over time, and productivity tools capable of measuring everything from “time to dial
tone” and “connect time” and the overall impact of the project was judged to be positive and in
most cases met management objectives or changes were put in place to get the system in line via
changes to processes.
One last area of concern, and an area where management was upset somewhat in their initial
thinking, was that of the staff implications of a move to a IP Telephony system. In management’s
initial view, after the migration was complete, it would be possible to lay off the entire old
telephony staff and translate the savings in salaries and costs directly to the bottom line. Their
strong belief in this potential benefit was rooted in the belief that “VoIP is just another IP
application. Get some IP phones, plug them into your existing IP network, and you’re ‘good to
go.”
Several things conspired to frustrate them in these efforts, not the least of which that this belief is
absolutely false—as previous chapters have ably demonstrated. Management was warned in very
clear terms to avoid this situation. What management actually found is that they were replacing
individual telephone, data, and video engineers with more valuable, less available multimedia
“triple play” engineers. This replacement process was often through hiring but, more often,
through training. The company was able to let the lowest-tier of least productive employees in
each category go and to keep the new breed of triple play engineers. Much of the training was
formal from outside sources, but some of the training was internal and on the job. The lesson in
this for organizations implementing IP Telephony is to never neglect the value of the body of
knowledge that traditional telco people bring to the project. The transport has shifted to IP, but
the basic application—humans speaking with each other—remains unchanged.




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Technical
The technical track runs in parallel with, and occasionally overlaps, the business and
legal/regulatory tracks. It begins with a review of standards’ status and maturity.

Review of Standards’ Status and Maturity
At the time of the case study, the review of standards’ status and maturity was a much more
important element than it is today. Today, for instance, systems based upon the Session Initiation
Protocol (SIP) are a given, while at the time of the case study, H.323, SIP, and ETSI TIPHON-
based systems were all being considered. Standards, and even Internet Requests for Comment
(RFCs) have come a long way both in terms of their refinement and the way in which they will
be implemented in the marketplace.
Such a review as this today will start by identifying the underlying assumptions, or givens, and
then delve into what may be less accurately known; this analysis will often go into the lower
layers of the network, as well. Only passing consideration was given to Multi-Protocol Label
Switching (MPLS), for instance, in this case study, while Ethernet MANs/WANs, MPLS, Virtual
Private LAN Service (VPLS), and Layer 2 Pseudo-wire VPNs are all options for the type of
QoS-assured networks needed for success in packet voice generally and VoIP specifically.

Develop Private Net, IP VPN, and Carrier/PTT Strategies
Following adoption of a set of standards, separate strategies were developed for using a
completely private network, an outsourced IP VPN based on MPLS, and a Layer 2 carrier/PTT-
provided VPN. What eventually occurred was the development of a compromise plan for a
hybrid network that leveraged the best aspects of one or all of the approaches depending upon a
fairly strict set of selection guidelines developed jointly by the business team to ensure that
business requirements such as ROI, TCO, availability, and reliability were met; the technical
team to ensure that technical elements were properly addressed; and the legal/regulatory team to
ensure that a specific set of characteristics was contractually possible.
For instance, some locations, especially in areas in which the company has a strong presence, are
sufficiently dense to be serviced by a company-provided private IP network; in some very
remote locations, the company operates their private IP network over PTT-provided leased lines
or over global carrier Ethernet WANs. In other areas, sites are served by global IP VPN
connections over a service-provider MPLS cloud. The actual selection criteria were put into an
Excel spreadsheet to allow a multi-column menu approach in selecting what to provide to a
given site based upon a variety of criteria, including application mix, QoS criteria, total site
bandwidth, location, traditional vs. IP telephony, and other similar qualifiers.




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Private Backbone Design/Upsize for Voice Traffic and Backbone Re-engineering
As an outcome of the design process, the company instigated the steps required to prepare their
existing private backbone to handle both the new voice traffic that would be originating and
terminating on their network as well as transit traffic that would be crossing their backbone. The
company used a set of network assessment tools in this phase that were invaluable in gathering
live data about the operation of the existing network and predicting the possible outcomes, and
resulting impact on, VoIP voice quality of certain design changes. The company found that the
outcomes were not always what they thought they would be and certain, often less expensive,
architectural and operational changes yielded better overall improvements in forecasted voice
quality than more expensive changes.

Establish Common Standards and Test Plan
After a careful evaluation of the three architectural choices, and the adoption of a hybrid, best of
breed approach, a set of standards and test plans for the voice application were developed. By
establishing a set of standards and testing methodologies for the overriding voice application, as
opposed to basing testing on the underlying technologies, the company came closer to
measurements, testing, and problem resolution techniques, that would have a more direct impact
on their users than they would have otherwise. Delay, for instance, was calculated “mouth to ear”
as opposed to from network edge to network edge or site to site. Mean Opinion Scores (MOSs)
were estimated and R-Values calculated for purposes of comparison, and SLA compliance and a
single, comprehensive, set of metrics was developed regardless of where in the world the voice
system user was connected.

Establish Private Net, IP VPN, and Carrier/PTT Demo/Test Capabilities
To fully test functionality and validate the network design, full test-bed networks were created to
match the private net, IP VPN, and Carrier/PTT systems. These test beds also allowed hybrid
configurations to be developed quickly for purposes of hybrid testing of real-world
configurations. Engineers, both from the internal company staff and from outside sources
including service providers, carriers, PTT authorities and consultants who had worked on the test
systems, were designated to either the support team or migration team. Support team members
would support the migration and ongoing operations efforts and would be given a permanent
staff position. Members of the migration team were temporary and would not remain on staff
after the migration was completed. This distinction was not completely enforced, however, and
some support team members left or were terminated and some migration team members were
given permanent positions, based upon their performance during the migration and other factors.

Training/Staging for Migration Teams
Most of the training for the migration teams occurred in the actual staging of equipment that
would be taken to the field. Systems were installed and initial testing and configuration
performed before the systems were taken or shipped to the field. Migration team members
received a minimum of training because an investment in their training would not yield a return
in the long run and they would be supported centrally by support team members.




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Training for Support Teams
Because the support teams were to be the ongoing multimedia support for the network, the
interim period prior to and in the beginning phases of implementation were a period of near-
constant formal and informal training. The support team members were to be the elite engineers,
and a substantial investment was made in their training and at each step along the way the
specific training they received was tracked back to the business benefits of the skills and a sort of
informal Return on Training Investment (ROTI) was calculated.

Review of Planning and New Telephony Project Rollouts
After a review of planning, and appropriate modifications for specific circumstances, the new
telephony project rollout was begun. If your network is to go no further than the borders of the
contiguous 48 states, your legal and regulatory issues will be minimal. If you use a service
provider, rather than taking a do-it-yourself approach, the issues will be even less complicated.
However, if you will be operating in multiple countries, using a variety of providers or some
other complicating factors, you will be spending more time on legal and regulatory areas. At the
very least, use the following items from the case study as a checklist to determine whether you
need to be concerned about the specific areas.

Legal
The legal track is executed primarily by attorney’s and includes contracts, compliance with
specific legal requirements, and the resolution of disputes with areas outside the company or
legal complaints arising within the company related to human resources issues.

Closing Contracts for Old Telephony System
One of the most important legal areas, from a standpoint of financial impact, is in closing out
contracts for the old telephony systems and related services. Ordinarily, audits of bills are
conducted within the business track with input and guidance from the technical track and, to the
extent that any moves, adds, or changes comply with the existing contracts can all be handled
without legal intervention. The reason the legal department is involved in this area, more-so than
in other situations, is that the changes to contracts, and contract cancellations, related to the new
packet voice project do not always track the dates and conditions foreseen in contracts. It is also
often the case that contracts can be renegotiated, often at more favorable rates or conditions, as a
part of the project.

New Contracts
In addition to disposition of existing contracts, new contracts to replace old contracts or with
completely new vendors need to be negotiated. In this case, the legal team collaborates with the
business and technical teams to ensure that the contracted goods or services meet the stated
business needs and that any accompanying SLA is properly constructed and administered.




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Regulatory
Regulatory issues within the U.S. could be the topic of another guide entirely; the good news is
that regulatory issues generally do not directly impact private organizations that are constructing
their own private packet voice systems—though they will impact the companies providing goods
or services. This is not the case in many other countries around the globe.

Compliance with National Law and Telco/PTT Regulation
Any company implementing a packet voice capability must be aware of local regulation and law
governing voice systems. Some countries are very open on the topic while others prop up corrupt
regimes, or those seen as corrupt by the U.S. government, using the hard currencies they receive
in settlement compensation when the money collected for calls to their country exceeds the
money collected for calls from their country. In other situations, the issues might be less
regulatory and more contractual, where a contract for voice services through a carrier or PTT
authority includes minimum minutes or dollar volumes.
In the case of the case study subject company, a database was created with all these factors and
more, and that database was used as an input into the business and technical tracks. Some of the
other areas tracked in the database are highlighted in the following sections.

Security and Privacy Issues
The extent to which voice communications, and often data communications, can be made private
is often strictly controlled in certain countries or under certain conditions. Although it is very
desirable for certain organizations to use sophisticated encryption techniques to keep their
information secure from competitors, governments, and even non-authorized employees, the
tools to do so are often prohibited. This is a problem in general for fixed location operations, but
for the mobile employee who may be traveling from country to country, knowledge of these
restrictions is very important as possession of certain hardware and software may be cause for
detention at the borders or by law enforcement authorities.

Carrier/PTT IP Bypass and Local Tail Drop-Off
One of the cost saving elements of VoIP is that it allows bypass of the traditional carrier/PTT
systems, allowing those older systems to be less heavily loaded and eventually removed. This
capability is not allowed in many global jurisdictions nor is local tail drop-off, which allows
delivery of a local call on a traditional circuit that has originated elsewhere in the network. This
restriction is often related to calls carried between company locations for individuals outside the
company but can restrict intra-company systems use as well.

PBX and Gateway Issues
Many of the restrictions on VoIP can be avoided by careful placement and creative use of
traditional PBXs, VoIP gateways, and other systems. If, for instance, a country does not allow
VoIP calls, placement of a VoIP gateway in an adjacent country that does allow VoIP calls will
allow calls to be turned into traditional TDM calls before crossing the border into the restrictive
country. Regionalization of gateways also allows concentration of support staff and may result in
overall higher system availability and lower support costs.




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Contact Centers/Call Centers
One final area of the utmost importance to many organizations, but not addressed explicitly in
the IP Telephony project map, is contact or call centers. Many organizations, be they
commercial, academic, or government, very survival rests on their contact centers and the ability
to do outbound calling to solicit customers, while many other organizations use contact centers
for internal support or to provide aid to customers and prospects. Either way, an organizational
shift to VoIP opens the doors to many contact/call center capabilities that are too difficult, too
expensive, or not technically feasible using older technology.

Special Considerations and Contingency Planning
There are many special considerations in shifting call center operations to a VoIP platform.
These include all-VoIP or hybrid traditional/VoIP, off-shoring the call center to another country,
and on-shoring (which is the outsourcing of the call center to in-country providers and, possibly,
distributed call centers that allow operators to take calls from their cubicles or their home). The
distribution of operators over such a broad geographic area as is facilitated by call takers
working at home also alleviates much of the contingency planning that must occur with a single
call center where all call center functions are concentrated in one place.

Phased Implementation and Call-Taker Training
Part of the migration of call centers to VoIP should involve a safe, secure, phased
implementation to the eventual hybrid or all-IP network. Training can often be accomplished
online using network-based training and collaboration tools, further increasing the financial and
operational benefits of a move to IP Telephony. Planning for migration of a call center to VoIP
can best be achieved as a completely separate project from the rest of the migration.


Summary
This chapter, though it couldn’t provide every implementation and migration detail applicable to
every environment, has hopefully provided a foundation on which you can build a successful
implementation. Combine the information provided in this chapter with your own experience and
knowledge to synthesize new ideas, new approaches, and new ways to be successful in your IP
Telephony project.
The next chapter will focus on how to keep things running smoothly after the migration and
implementation. Much time will be spent on the “tools of the trade” that allow you to monitor
your networks and ensure that they remain within the guidelines established within the SLA.




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Chapter 6: Ongoing Operations
Most, if not all, successful start-up companies are founded by creative, energetic, entrepreneurial
individuals whose 16-hour days, eating at the desk or on the run, decisions on-the-fly and out-of-
the-box thinking that could not possibly be sustained over the long run, are replaced by a new
team. After the “start up” phase, top management positions are occupied by less creative, more
down-to-earth, and more operations-focused personnel whose job it is to standardize operations
and run the company going forward. Companies with foresight often move the founders into key
roles of business development or product improvement. Those companies that do not do so often
stagnate without the enthusiasm and commitment of the original team.
This transition happens as often in Voice over IP (VoIP) projects as it does in entrepreneurial
start-ups and for many of the same reasons. The job of creating a project, selling it to
management and users, and making it happen—in any size organization—is different to the job
of running the system day-to-day. It is the rare individual who can fill both roles. If your job is to
take the reins, create harmony from chaos, and run the new system day-to-day, this chapter is for
you.


Network Operation
The first step as you emerge from the implementation phase and enter the operations phase is to
go back to the beginning and review the original plans, goals, and objectives and to ask several
questions:
   •   Are the original goals and objectives still valid?
   •   Have technologies changed significantly?
   •   Are standards more mature, and does it matter?
   •   Have laws or regulations in any of the areas in which you operate changed in any
       important ways?
   •   Have your operating procedures or business environments evolved in a way that impacts
       your new telephony system?
It may have been several months or longer since the VoIP project was first conceived and
planned. Things may have changed: the business direction, the services your users need, and
costs, technologies, and products. This review step should also include input from management
and users to ensure that all current views are considered.




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Once you have validated the original goals, or modified them to reflect any changes, it is time to
review the Service Level Agreement (SLA), or agreements, to ensure that they reflect the
original goals and objectives and to make adjustments to the SLAs, if needed, in a similar
manner to what was done to the project objectives. Once this administrative step is completed, it
is time to schedule a review cycle to periodically assess the validity of the project goals and set
new goals based upon the network optimization process that will be ongoing in parallel with the
operations cycle.
The important question is, “How often should operations be reviewed?” Some organizations are
fairly static and have, in fact, been doing much the same job in much the same way for many
years. Those organizations might very well be able to have a comprehensive review scheduled
annually if one is not triggered sooner by the optimization process. Other organizations that are
more dynamic and are undergoing more rapid change—or growth, moving into new markets,
evolution of processes and procedures, or similar changes—may be at the other end of the review
spectrum and might, at least for the first year or so, want to conduct a formal review monthly.
The timing is really up to the organization and their needs, but the key is to get it on the calendar
and move on to operations management. Using the SLA as a guideline, take a closer look at
network capacity, reliability, security, voice quality and call quality, growth management, and
how to measure and monitor the network.

Capacity, Performance, and the Broadband Flip
Anyone who has been in networking before the start of the low-cost bandwidth era is very
focused on capacity issues. The industry veteran most likely has a genetic predisposition to
focusing on optimizing bandwidth for purposes of capacity and cost savings. The reason for this
is very simple: prior to the “broadband era” bandwidth was outrageously expensive and
bandwidth demand was calculated and optimized for capacity in much the same way conveyor
belts in a factory are sized: getting the most done with the least resources by filling the conveyor
system up as full as possible. But, for several reasons, this is no longer the case and you must
now do the “broadband flip”
For lack of a better label, I will refer to what came before the broadband era as the narrowband
era. The narrowband era was characterized by individual circuits, or channels, that were
dedicated to the use of a single communication, be it a stream of voice, data, or video bits or
packets.




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Figure 6.1: Narrowband bandwidth calculation.

In narrowband, a general rule of thumb, the two-thirds rule, or a similar approach is often
applied. When utilization grows to about two-thirds of the capacity of the circuit, or channel, it is
time to request a larger circuit. If a network manager budgets properly, tracks the rate of growth,
and knows the installation lead time for a new or upgraded circuit, the management of this
process can be fairly straightforward. For instance, as Figure 6.1 shows, if a remote site is:
    •   Connected to the central data center by a 9.6Kpbs circuit that was averaging a utilization
        of 5.2 kbps in January.
    •   Utilization was growing at a rate of 10% per month.
    •   It takes 3 months to order, install, and test a new 19.2Kbps circuit.
In this case, simple math shows that the new circuit should be ordered no later than April 1st, and
probably sooner, to be up and running by August 1st, before demand exceeds the capacity of the
9.6 kbps circuit. Lacking unforeseen growth or upgrades or changes unreported by the user
groups, the system worked. Ahhh, life was so much simpler back then…but not necessarily
better.
To get from “back then” to “right now,” you must do the broadband flip. Stop thinking about
bandwidth for capacity and begin thinking about bandwidth for performance. Consider a
traditional, channelized, narrowband T1: it has 24 individual channels each capable of carrying
exactly 64,000 bits every second. In fact, the type of sharing, or multiplexing, as the engineers
prefer to call it, that the narrowband T1 uses—Time Division Multiplexing (TDM)—gives each
of up to 24 simultaneous users the impression that they each have a connection that is dedicated
to their own use and, in fact, they really do.
Figure 6.2 represents the narrowband T1. Let’s take a closer look and ponder some possible
scenarios and how they might play out in the channelized narrowband world. In the hypothetical
T1 that Figure 6.2 shows, the darkest of the 24 channels, such as the topmost channel, are both
configured/reserved and are in use at the current moment. The lighter colored channels, such as
the bottom-most channel, are configured/reserved but are not currently being used. The white
channels are neither configured, reserved nor in use.




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What if the top channel needed more than 64 kbps of capacity? Could it share the channel below
if that was already configured/reserved? No, it could not. Could it use any of the idle channels?
No, not dynamically. To do so would require a rather lengthy reconfiguration process. Is it
possible to use more or less than 64 kbps for a single connection? Yes, but not dynamically: a
lengthy reconfiguration and provisioning process is required. The point is, narrowband
connections are “nailed up” and not capable of changing dynamically.




Figure 6.2: Channelized Narrowband T1.

And, what about capacity? How is capacity adjusted? Capacity is increased in groups of the basic
64 kbps channels, called DS0s, up to large capacities, as shown in Table 6.1. When the demand
exceeds 24 individual channels in a single group, called a T1, the T1 can be replaced with a T2,
which can accommodate up to 96 simultaneous 64 kbps connections. It is rare, however, to
encounter a T2, as it is more customary to make the jump in most cases directly to a T3, which is
capable of supporting as many as 672 simultaneous connections, or a fractional T3 comprising
some number of T1s or T2s. The next step before the Synchronous Optical Network (SONET) is
a metallic T4, which can accommodate up to 2016 simultaneous connections. The system was
built to accommodate progressively larger numbers of the same size, 64 kbps, connections, not to
allow for greater bandwidth, as broadband does.

   SONET, as defined by GR 253-CORE from Telcordia, is a method of communicating digital
   information using lasers or LEDs over optical fiber




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Table 6.1: North American (Narrowband) digital hierarchy.

Figure 6.3 shows the narrowband T1 from Figure 6.2 reconfigured for broadband use. Instead of
being carved into 24 identical small channels the non-channelized broadband T1 is a single large
channel that can be shared by multiple connections, possibly more than 24, but the key point is
that the full single channel is dedicated to a single connection at a moment in time. How is this
accomplished? In the narrowband T1, we could associate specific bits with specific channels by
their positions. The first group of eight bits read from the wire belongs to the first connection; the
second group of eight bits belongs to the second connection, and so on up to 24. Any idle or non
configured channels have filler bits to maintain the positioning and this process is repeated 8000
times per second to deliver 64 thousand (8 x 8000) bits per second per channel.




Figure 6.3: Non-channelized Broadband T1.




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In the case of the broadband T1 shown in Figure 6.3, bits are not associated with their
connections by their position but rather by additional protocol bits that are used to create frames,
packets, or cells, which are three different types of electronic containers for transporting
broadband bits. Broadband is actually not as efficient as narrowband for transporting multiple
identical channels of information, but that is rarely the objective anymore. Broadband is good at
dynamically allocating bandwidth resources to the rapidly varying demands of multimedia such
as voice, data, and video.
In Figure 6.3, it is also possible to observe that idle capacity is not trapped in individual
connections that cannot be dynamically shared, as it is in narrowband. In broadband, all capacity
can be used dynamically and idle bandwidth only occurs when all information has been
transmitted for all connections.
And what about capacity? Broadband capacity is enhanced by increasing the size of the “pipe,” not
by increasing the number of same-size pipes in some sort of bundle as is the case with narrowband.
It is possible to configure narrowband connections—such as T1, T2, T3, and T4—for broadband
access, but it is also possible to use Digital Subscriber Line (DSL), cable modem connections,
metro and wide area Ethernet, and other broadband services to allow you to get as close as possible
to the right size “pipe” for both capacity and performance needs. This is where you must make the
big leap from considering bandwidth for capacity—though capacity is still a consideration—to
considering bandwidth for real performance and doing the broadband flip.
As Figure 6.4 shows, increasing the capacity of a narrowband circuit increases the aggregate
capacity but does nothing for transmission performance. Regardless of the number of 64 kbps
connections bundled together, the individual channel capacity does not change. Figure 6.5 shows
that increasing the size of the actual “pipe” and utilizing it in a broadband fashion have an
important impact on performance per connection.

    Transmission Time for 64K bits
             (Narrowband Example)                 Transmission Time for 64K bits
 1 DS0 Channel                     = 1 second               (Broadband Example)
 1 DS0 Channel in a T1             = 1 second   Broadband T1                  = 1/24 second
 1 DS0 Channel in a T2             = 1 second   Broadband T2                  = 1/24/4 second
 1 DS0 Channel in a T3             = 1 second   Broadband T3                  = 1/24/4/7 second
 1 DS0 Channel in an OC-3          = 1 second   OC-3c                         = 1/24/4/7/3 second
 1 DS0 Channel in an OC-12         = 1 second   OC-12c                        = 1/24/4/7/3/4 second
 1 DS0 Channel in an OC-48         = 1 second   OC-48c                        = 1/24/4/7/3/4/4 second

Figure 6.4: Narrowband transmission times.      Figure 6.5: Broadband transmission times.




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Table 6.2 and Table 6.3 show the wide range of possible bandwidth needs, per connection, for a
variety of combinations of Voice over Packet and transmission protocols. The difference
between Table 6.2 and Table 6.3 is the packetization interval. Table 6.2 shows the impact of
creating and sending packets every 10 ms, resulting in more packets but a smoother flow of
voice that is less susceptible to packet loss; Table 6.3 uses a packetization interval of 30 ms.
With a 30 ms packetization interval, bandwidth is consumed less quickly because more voice
samples are placed in each packet, thereby reducing the number of overhead bits per sample;
however, the connection is impacted more heavily by packet loss as the loss of a single packet
will result in the loss of three times as many voice samples.




Table 6.2: VoIP bandwidth estimate with 10 ms packetization interval.

Source: Matthew Michels, Nortel Networks, used with permission.




Table 6.3: VoIP bandwidth estimate with 30 ms packetization interval.

Source: Matthew Michels, Nortel Networks, used with permission.



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These types of bandwidth utilization models can also be used to determine the impact of voice
activity detection (VAD), which eliminate packets that are carrying silence. By eliminating
silence packets, it is possible to get additional benefits with a negligible impact on quality
assuming the VAD system is properly configured and tuned. Proper configuration involves the
treatment of the absence of silence packets at the destination and how silence packets are chosen
for non-sending at the source. If in the absence of voice packets, no sound is played into the
listeners’ ear, as in the earliest implementations, VAD can be very disruptive to the call and
perceived call quality, causing the talker, upon hearing nothing from the listener, to pause and
ask “are you still there?” Insertion of comfort noise, also known as Comfort Noise Generation
(CNG), is vital. At the source, if packets are not sent that are predominantly silence, chopping or
clipping of syllables can occur. Not setting stringent enough criteria for packet non-sending will
lessen the value of VAD, which can be as much as 4:1, meaning that an actual call using VAD
will require 25% of the bandwidth required by the same call not using VAD.
Results of this kind can be used to play “what-if games” and to develop realistic bandwidth
estimates for different types of connections and, thereby, be able to establish baselines for the
operation of your specific network. It is likely that you will end up, with at the very least, two
profiles: inside calls and outside calls. But you may very well have different bandwidth
requirements for a much more specific set of user requirements, much along the lines of the user
profiles developed in Chapter 4. It might be possible to establish as few as four user profiles that
are common to all departments, though it is more likely that there will be far more profiles for
any given situation. Let’s look at four basic profiles, just to give an idea of the type of feature
sets they might need.




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               Profile                     Possible Job Functions                 Telephony Features
                                     Office worker, inside sales person,  Dial Tone
                                     general staff, supervisor, low level In-house (extension) dialing
                                     manager, help desk worker            Local dialing
Basic Telephony User                                                      Long Distance dialing
                                                                          Basic Voice Mail
                                                                          Call Waiting
                                                                          Caller ID
                                     Secretary, senior secretary,        Basic Telephony User PLUS:
                                     telemarketer, senior help desk       3-way Conference calls
Intermediate Telephony User          worker                               Skills-Based Call Routing
                                                                          Call Forwarding
                                     Administrative assistant, group     Intermediate User PLUS:
                                     admin, senior telemarketer           Enhanced Voice Mail (add FAX
                                                                           storage/retrieval and advanced
Advanced Telephony User
                                                                           messaging and retrieval features)
                                                                          Enhanced Call Forwarding
                                                                          Multi-party conference calls
                                     Senior sales person, senior manager Advanced User PLUS:
                                                                          Multimedia conferences (web plus
Power Telephony User                                                       audio)
                                                                          Enhanced Call Forwarding with
                                                                           Remote Access
                                     Senior manager, technician,          Telephony
                                     developer, product marketing         Call Routing Management and
                                     manager, senior sales manager         Follow-me Features
Multimedia User                                                           Voice Messaging
                                                                          Instant Messaging
                                                                          Multimedia Conferencing
                                                                          Collaboration
                                     Sales executive, account manager,    Remote Telephony
                                     senior sales manager, sales VP,      Remote Data
Road Warrior                         sales director                       Remote Video
                                                                          Advanced Call Routing and Follow-
                                                                           Me Features

Table 6.4: Sample user profiles and telephony features.

It is clear that many organizations will have additional requirements, such as various skill levels
of Call Takers, Call Taker Supervisors, operators, executives, administrative assistants, and so
on, but this basic mapping of user profiles to features will show a way of simplifying both the
testing and overall implementation issues.
So what should the operations personnel do with all this information? Using these
considerations, baselines should be established, growth should be observed, and system capacity
tracked. Additionally alternative routes should be established in case of failure, possibly with
multiple carriers and other back-up plans put into place, as described in the following section on
reliability.
There are many modeling, simulation, and tracking tools available to help with this task, as even
the smallest networks are too complex to be tracked manually. Tools to help with this function
are described at the end of this chapter.



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Reliability
Establishment of “reliability” baselines and system monitoring to ensure that all components are
operating within those parameters is very important to the proper ongoing operation of the
network. Reliability must be carefully considered from the standpoint of the impact on the end
user. Reliability issues may be considered, and prioritized, in terms of “service impacting” and
“non-service impacting” outages. To implement a reasonable approach to reliability, one must
consider the percentage of time a system is needed, that the system is actually required; versus
the total amount of time the system is available.
Beyond the service impacting and non-service impacting categories, reliability must be
monitored and addressed in terms of “root cause analysis,” and this must be a key part of training
and standard operating procedures. Root cause analysis, in the terminology of the medical
profession, allows you to treat the illness, not the symptom. If we stay for a moment with the
medical analogy, we would describe the visit of a patient to the doctor. The patient complains
about a temperature. Does the doctor tell the patient to lie down in a bath of ice water to lower
their temperature or does the doctor ask further questions to ascertain why the patient has a
temperature? Remarkably, many call centers and help desks are designed to get the “patient” to
fill a bath with cold water. These help desks are usually characterized by setting goals for the call
takers such as “closed tickets per hour” or a similar metric. The call centers and help desks that
will go to the next step are usually setting goals for their call takers based upon user satisfaction
and solving user problems. They will ask the next diagnostic questions and consider the systemic
issues of any problem—and that is what must be done.
Reliability is related to the “end-to-end view”—the closer a problem is to the end user, the fewer
users that problem will impact. The other side of that example is that the closer a problem is to
the core of the network, the more users it will impact. For example, a problem with an individual
user’s SIP phone will only impact that user’s ability to make and receive calls. Their voicemail
will still be working, but they will not be able to engage in interactive conversations. Although
this does represent a problem, the scope of the problem and the set of users a problem impacts
are limited. However, if a problem affects the VoIP call server, it will impact all users of that
server. These considerations should drive the investment of time, money, and effort to prevent
problems and your investment in mitigation when problems do occur.

Security
An important part of ongoing operations is the security of the entire system generally and
specifically; in this case, the Voice over Packet services. There are three distinct areas of concern
relating to security of voice, specific security threats and vulnerabilities within each area. In
hybrid systems, combining traditional systems and Voice over Packet, the number of distinct
areas of concern increases substantially. As Figure 6.6 shows, the three areas are the phone itself,
the network, and the server. Within the network, there are also three areas of vulnerability: the
phone access to the network, the network backbone, and the server access to the network.




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Figure 6.6: Areas of vulnerability for IP-based voice systems.

Let’s start a discussion of security from the phone end of the connection. What many people
forget is that Voice over Packet-capable phones, regardless of connection type, are actually
computers running specific applications. Very often, the phones are programmable in some
generally available programming or mark-up language, such as the eXtended Markup Language
(XML), and can be reprogrammed as easily as they can be programmed. The types of
vulnerabilities, therefore, are very similar to vulnerabilities in the server with the exception that
security issues with individual phones may impact one or a small subset of users while server
breaches impact all users of that server. Phone and servers are susceptible to three types of
attacks: malicious software, or malware-enabled attacks, as they have come to be known; content
compromise; and Denial of Service (DoS). An additional consideration, even though most
organizations consider VoIP and its related services to be “free,” (organizational telephony
services are about as free today as 800 service was free on traditional telephone networks)—is
that it’s not. Someone has to pay for the bandwidth and, maybe more importantly, the time of the
person using the service. Many organizations are shocked when they do an audit of the protocols
running on their network and find a large percentage of their bandwidth is used by personal
Skype, unauthorized NetMeeting, and other non-approved system uses. These are often the silent
destroyers of response times and performance that chokes off organizational productivity.
Malicious software, or malware, is software that is surreptitiously downloaded to a phone or
server that does things of which the user is unaware and would probably not approve if they were
aware. Malware can be as simple as compromising dialed phone numbers or calling directories
by sending them to an unauthorized third party or disrupting normal operation of the phone, such
as not ringing or selectively not ringing, or allowing unauthorized use of the phone. Content
compromise can allow access to the voice communication by unauthorized third parties and can
possibly overcome encryption methods by allowing the unauthorized third parties access to
encryption keys. DoS attacks are the simplest type of attack and simply block the user from
establishing all calls or specific calls or from using certain system features. DoS attacks are the
simplest but the most common.
Within the “network” portion of the call, the vulnerabilities are typically of the eavesdropping or
DoS varieties. There is greater vulnerability, typically, in the user/phone access than there is in
the server access and backbone. The backbone is usually the least vulnerable to ad hoc attacks,
especially if the backbone is a Virtual Private Network (VPN) provided by a trusted third party.




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Another aspect of security that must be managed operationally is systems that are put into place
to enhance security—such as firewalls, stateful inspection engines, and Network Address
Translation (NAT)—but that often collide with the Voice over Packet systems and prevent their
proper operation. With packet voice services playing an increasingly important role in
organizational communications, these are considerations that must be made prior to
implementation of the security systems or upgrades and enhancements to the capabilities of the
existing systems.




Figure 6.7: Voice over Packet system implemented via a gateway.

A second consideration is packet voice systems that are implemented over traditional telephony
systems. They share the basic issues with the systems implemented via IP-enabled phones except
that the target for malicious software will be the gateway device that connects the traditional
telephone devices to the IP network. This arrangement has not proven to be more or less
vulnerable to hacking than the situation in which the gateway functionality is embedded in the
telephone device itself.
There is yet a third situation, which Figure 6.8 shows that must be considered from a security
standpoint: the situation whereby a traditional telco network, either public or private, and an IP
voice network are interconnected. This is a very common situation that, the call at some point in
its duration will be subjected to the security issues discussed earlier relative to IP-based voice
and to all of the security issues associated with traditional telephony. Although traditional
telephony security issues do overlap with some IP-related issues, the traditional telephony part of
the call has some unique security vulnerabilities as well. The issues that overlap have to do with
unauthorized use of services, and the ones that are unique are related to these issues. The cost per
packet voice call is usually very low compared with the cost of traditional calls, so, therefore, as
mentioned earlier, the largest component of the true cost is usually productivity and time costs,
as opposed to per minute charges, referred to in old telephone network parlance as “toll
charges”—the same term that gives us the word “toll fraud” for the theft of services with those
charges associated.




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Figure 6.8: Voice over Packet and traditional telephony hybrid.

The toll fraud issue in a hybrid network is actually a double-edged sword: the IP network can
provide new, often unsecured access to traditional telephone networks that have toll charges
associated, and the traditional telephony networks can provide unchecked access to the IP
networks. In addition to newly minted exploits, the old ones still work and are still an issue:
Mailboxes that are set up on corporate systems and resold as part of a service package or used by
terrorists or for untraceable or difficult-to-trace drug transactions. Sale of long distance, often
international, calls via corporate PBXs; and other similar loopholes and backdoors cost
organizations billions of dollars a year. This will continue as long as the IP and traditional
networks are interconnected, which in many cases will be for some number of decades.
What can an organization do? The best bet is to work with your security team, whether they be
an inside team or a part of your service provider, to monitor activity and to thoroughly check out
any activity that appears to be abnormal—or an unusually high volume of normal activity.
Eccentricities and anomalies that should be monitored are unusual protocols, unusually high or
low call volumes, appearance or disappearance of encryption, time of day shifts in usage,
unusually high or low volumes for a certain phone number or user, and similar patterns that can
mean abuse. Restoring software to IP phones and servers, PBXs, and gateways periodically is
also a good idea as is using secure versions of the protocols and systems used to install, maintain,
and operate them. Using Secure Shell (SSH), for instance, to maintain gateways, as opposed to
the common Telnet protocol that is available to every hacker, is a good idea as is using secure
versions of SNMP, OSPF routing protocol, and other fundamental workhorse protocols used to
run the networks that carry the increasingly important Voice over Packet traffic. These protocols
have many benefits when compared with their insecure variants in that they employ encryption,
tunneling, and other security capabilities to ensure greater security of the applications.




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Call Quality and Voice Quality
We have previously, and vigorously, defined call quality and voice quality and described the
difference between the two. Nowhere is this distinction more important than in the operations
phase of the Voice over Packet network’s life cycle. Call quality is directly related to QoS and its
underlying methods and measurements. Call quality and QoS measure and attempt to optimize,
in some cases dynamically, packet loss, delay, delay variation, and system availability. Voice
quality, however, is directly related to Quality of [User] Experience (QoE) and is very tightly
coupled with the opinion of the humans using the system. QoE is only loosely coupled with QoS
even though it is an automated approximation of human QoE opinions derived from the statistics
of QoS upon which Service Level Agreements (SLAs) and network-based alarms are based. The
reason for this is that QoE scores are based upon the opinion or averaged opinions of one or
more humans. Those humans are subject to a variety of factors that are impossible to model and
reproduce mathematically, but those humans are also not available to judge voice quality on a
given call, so it is important to use automated systems. The benefit of automated systems is that,
although they do not exactly reproduce the human scores, they are consistent and always
available. The drawback is that they only roughly approximate the humans’ opinion of the voice
quality.
The two types of testing are non-intrusive and intrusive. Non-intrusive testing does not require a
special test call and is based on actual conversations or is calculated from metrics gathered
during an actual call. Intrusive requires a special test call and is based on comparison of source
signal with the signal after transmission. Both are valuable operationally but the non-intrusive
testing will raise a realistic red flag sooner as it relates directly to the quality of the voice as
experienced by the user because it is a set of measurements made on real calls. Intrusive testing
not only is artificial in that it is not based on actual calls but also can place an additional burden
on a network for the test calls themselves that may distort the test results and even have a
negative impact on real-live calls in progress. This is especially true if intrusive testing is
performed at the most desirable time to do testing: when the live call load on the network is
particularly high and nearing upper limits.
Non-intrusive tests include the Mean Opinion Score (MOS), derived MOS, the E-Model (ITU-T
G.107) R value, and, more recently, ITU-T Calculated Planning Impairment Factor (ICPIF)
Scores (ITU-T G.113). MOS testing has been adopted from traditional telephony, and
historically has been, at least for North American telco operations, the actual opinion of voice
quality from a panel of human judges from central Illinois. Even though one might question
whether it is statistically significant, the traditional panel has been composed of 16 people: 8 men
and 8 women. The panel of judges is sequestered in a sound-controlled laboratory environment
and judges the quality of voice on a variety of systems on a scale from 1 (lowest) to 5 (highest).
MOS is useful for human user QoE analysis but is virtually useless for SLAs and network
monitoring due to the difficulty in reproducing the results reliably and under a wide variety of
circumstances.




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The E-Model (ITU-T G.107) is a much newer innovation and is intended as a design tool that
predicts average voice call quality using a mathematical model that takes into account the
estimated impact of delay, delay variation (also known as jitter), packet loss, and performance of
the codec (the voice coder that translates human speech into bits at the source and decodes the
bits back into analog waves at the destination). The result of the E-Model calculation is the R
factor or R value. The R value is an estimated voice quality rating with a range from 0 to 100. 0
indicates lowest call quality and 100 indicates a perfect quality score. From the E-Model’s
output, the R value and derived MOS can be calculated, as Figure 6.9 shows.




Figure 6.9: Correlation of E-Model R values to MOS scores for derived MOS.

In actual operations environments, the author recommends a target of 100, which matches to a
MOS of 5.0—a perfect and unachievable score. But, like archers, Voice over Packet operations’
personnel need to aim above the mark to actually hit the mark and, in this case, aiming for a
perfect score will probably put the voice quality in your network consistently in the R value
range of 94 or so. This matches a derived MOS of 4.4, which is in the same 4.3 to 4.5 range
generally achieved by most traditional voice networks based upon Pulse Code Modulation
(PCM) and TDM transport.
The real value of the E-Model lies beyond the design work for which it was originally created
and is realized more in the operations area, which is the subject of this chapter. Operationally, E-
Model scores are based upon measurable parameters found in the network and which can, if
needed, be reproduced in the lab. In actual operation, E-Model evaluates Real-Time Protocol
(RTP) Streams based upon source and destination addresses, port numbers, and sequence
numbers and creates a jitter profile and predicts the impact of the Internet Protocol (IP) bearer on
MOS with an 80 to 90 percent correlation to actual human scores.




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An additional non-intrusive evaluation technique is the ITU-T ICPIF score (ITU-T G.113). Some
manufacturers are beginning to adopt ICPIF for voice quality calculations due to the additional
levels of sophistication being required in assessment of voice quality. In fact, from the
perspective of VoIP service providers, this author has always maintained that when all prices are
equal—and as low as they can realistically go without bankrupting the service provider—other
differentiation factors will emerge that will be the basis for purchase decisions and that the
human interpretation of voice quality will be one of those factors. We are already beginning to
see this in services offered to the public. From an organizational point of view, this has never
ceased to be true for two important reasons. The first reason is that in the
organizational/enterprise/corporate/agency environment, cost savings are not seen and felt
directly by users in most cases: they only see an improvement, degradation, or no change in the
quality of voice calls and the related telephony and multimedia services available to them. The
second factor is that although most people’s opinion of what call quality is possible, and
reasonable, has been strongly, and downwardly, influenced by the use of cell phones, many in
the business environment still have a traditional, reliable, high-quality telephone with which to
compare new service. Therefore, a lot of work is being done in the area of voice quality, much of
it in anticipation of the shift of importance to voice quality and the ICPIF score, available since
February of 1996, is a good example of this.
ICPIF takes into account the factors that the E-Model does and additionally gives weight in the
evaluation to signal attenuation distortion and loss, circuit noise, codec distortion, group delay
distortion, one-way transmission time, talker echo, and some additional parameters of special
importance. Figure 6.10 shows the rough correlation of ICPIF scores to MOS. It is, therefore,
possible to derive MOS values from ICPIF scores just as it is possible to do so using R values.




Figure 6.10: Correlation of ICPIF to MOS.




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Intrusive evaluations, while not as valuable operationally as non-intrusive tests, still have
operational value. Intrusive tests are often used to perform automated testing during busy hours
or non-busy hours, and several companies have systems that are automated and operate free of
human intervention. The family of intrusive tests includes Perceptual Analysis Measurement
System (PAMS), Perceptual Speech Quality Measure (PSQM), Perceptual Evaluation of Speech
Quality (PESQ), and PESQ+. All of these evaluation approaches are derived from PAMS, which
was developed by British Telecom as a replacement for human MOS values. Although PAMS
and its close relatives that follow do not replace MOS, only approximating it within +/- 10 to 20
percent (a window of up to 40%), they are still excellent for benchmarks and comparisons. This
group of tests work by comparing the original analog voice wave with reproduced/transmitted
speech using a complex weighting method intended to take into account characteristics important
to the human ear. The scale is from 0 to 6.5 with 0 being “perfect” (that is, no difference between
waves). PSQM, ITU-T Recommendation P.861, and PESQ, ITU-T Recommendation P.862 and
PESQ+ operate very similarly to PAMs but come close to approximating MOS values.

Measurement and Monitoring
So what is the value of this in operation? The SLAs that the operations group inherited from the
design and procurement process should be based, to some degree or other, on MOS. Because it is
not practical to line up humans to listen to all calls, an automated method is needed and this, or
some other variants—some proprietary and some standard—can be established as the network-
wide standard. Non-intrusive tests can be used as the basis for evaluating human user complaints
and intrusive testing can be used to anticipate issues and take a proactive approach to fixing them
before they reach a level sufficiently bad to generate user complaints and the resulting trouble
tickets. Baselines need to be established early in the operational life cycle, consistent with the
SLA, and any variation above or below the baseline by more than your tolerance—+/- 5% gives
a 10% window that is reasonable in most circumstances—is reported, a root cause analysis is
done, and whatever steps are needed are performed to get the observed scores back into line with
the baseline values for the network.

NetFlow/IPFix
An important consideration for measurement and monitoring is the exact methodology that will
be user to implement it. In many cases, organizations opt for bulk statistics or exception
reporting through SNMP or Remote MONitoring (RMON), but a methodology that is being
standardized and receiving a lot of attention recently is NetFlow, also known, in the standards
area, as IPFix. NetFlow is an open but proprietary network protocol developed by Cisco Systems
to run on Cisco IOS-enabled equipment for collecting IP traffic information. Cisco routers that
have the NetFlow feature enabled generate NetFlow records; these are exported from the router
in User Datagram Protocol (UDP) or Stream Control Transmission Protocol (SCTP) packets and
collected using a NetFlow collector. Juniper Networks provides a similar feature for its routers
called cflowd, which is basically NetFlow 5. Huawei Technology routers also support the same
technology, but call it NetStream.




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Historically, network flows have been defined in many ways. In the case of NetFlow, Cisco uses
the common 5-tuple definition, where a flow is defined as a unidirectional sequence of packets
sharing all the following five values:
   •   Source IP address
   •   Destination IP address
   •   Source TCP port
   •   Destination TCP port
   •   IP
The router will output a flow record when it determines that the flow is finished. It does so by
flow aging: when the router sees new traffic for an existing flow, it resets the aging counter.
Also, TCP session termination in a TCP flow causes the router to expire the flow. Routers can
also be configured to output a flow record at a fixed interval even if the flow is still ongoing. In
Flexible NetFlow (FNF), an administrator could actually define flow properties on the router.
Beyond the Cisco realm, the Internet Engineering Task Force (IETF) is standardizing NetFlow
under the title IPFix—IP Flow Information Export—and it is being made available on a wide
variety of network products. The gathering and analysis of NetFlow/IPFix data is often best
performed on separate management platforms using third-party software of which a wide range
is available.

Growth Management
Growth of a Voice over Packet network is a part of the bigger project of managing growth of a
multimedia IP network. As such, it has much in common with the bigger project but also has
issues of its own. Growth must be managed in terms of fixed assets and equipment budgets,
access bandwidth, usage of the shared bandwidth, and shared resources such as switches, routers,
telephony servers and gateways, and numbers and licenses.

Fixed Assets and Equipment Budgets
The first job will be keeping track of physical assets, which have both hardware and software
elements. The Voice over Packet network will include hardware and possibly software telephony
clients for the end users. In many cases, this will include the reallocation of traditional telephony
devices such as desk phones to the Voice over Packet inventory. Fixed assets will also include
telephony-only systems, such as SIP servers and gateways, and might include an allocation for
the part of shared systems, such as routers and switches that are used by voice traffic. Although
this is often a secondary consideration on the list of many seemingly more critical items for
operations staff, this can be a time-consuming and possibly career-threatening item if ignored.
Proper tracking and forecasting of growth includes awareness of pending management decisions
regarding mergers and acquisitions, layoffs, new product lines, or anything that can substantially
impact the number, and profile, of the user community. The recordkeeping associated with this
task will be addressed further in subsequent sections.




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Network Bandwidth: Access and Backbone
The second job will be managing the bandwidth pool. Bandwidth utilization issues have been
addressed in a prior section and at this point in the life cycle of the Voice over Packet network,
calculated values should have been compared with actual values and you should have a clear
picture of the bandwidth that is used by each type, or profile, of user. You should also be aware
of the number of each type of users and the likelihood of that sub-population to substantially
grow, shrink, or be reallocated to a different profile.
Recall that an element of the user profile was the degree of mobility, which can impact user
access as well as the amount of shared bandwidth they use in the network backbone. All of these
factors need to be managed; human resources and management hiring and acquisition of other
organizations needs to be factored in and a budget developed. One of the biggest factors that can
influence a finely tuned and well-performing network is a shift in users or among and between
user profiles.
How can this be managed operationally? Most systems, especially VPN systems and others that
require user sign-ons, allow resources to be tracked to some level by individual user and, at the
very least, by user group. The key is to anticipate the need and to establish appropriate profiles in
advance and to use the profiles to establish user groupings. Another benefit of such profiles is
that, very often, user assignment to specific profiles can be done using a standard template that
substantially reduces the chances of errors and makes adding new users much faster and easier.

Shared Resources: Switches, Routers, Call Servers, IP PBXs, and Gateways
Telephony users are typically multimedia users, not telephony-only, though in either case, it is
possible to determine the amount of shared resources that a user consumes. This is a very
important growth planning and management element and should be tracked, reported, and
reviewed regularly. Specific alarm thresholds being set to send up some sort of electronic signal
flare should utilization approach or exceed thresholds between reviews. Shared resources to
consider include the switches and routers that are responsible for transmitting and receiving all
traffic and the call servers, IP-PBXs, and gateways responsible for handling voice traffic.

Numbers and Licenses
Resources that are often not considered, but must be, are numbers and licenses. The two main
numbers that are required are phone numbers and IP addresses. The licenses that need to be
managed are any software licenses that are charged per user, per seat, per port, or on any other
incremental basis.
Phone numbers, most often, must be purchased from an outside telephony organization so that
they may be registered in the proper places and so that appropriate routing may be established in
the Public Switched Telephone Network (PSTN). Organizations are strongly advised not to use
telephone numbers of their own creation, even inside what seems to be a private network, due to
the fact that almost invariably there will be a need to connect to the PSTN and number
duplication is inevitable. It is worth the extra cost. This is a bit less true with IP addresses
because NAT can be used to map internal, potentially conflicting, IP addresses to external,
registered IP addresses, but this point should be very carefully considered. It is possible to create
and manage an inventory of available phone numbers and IP addresses, to assign them as
needed, and return them to the inventory for reassignment when they are not in use.



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Licenses, likewise, should be inventoried and returned to the inventory for reassignment when
not in use. When a user leaves or is terminated, the company almost always remembers to get
that user’s keys and delete their passwords. A part of this procedure should also be to return any
licenses to the common inventory. Licenses cost money and are needed to operate most Voice
over Packet systems.

Recordkeeping and Documentation
This is the boring but inevitable part of any operations environment: Keeping track of the assets.
Although boring, and inevitable, this is still a very important function and one that must be kept
up to date as information may be requested at any time.

Fixed Asset Accounting
Fixed assets include all PCs, laptops, palm devices, routers, switches, Power over Ethernet racks,
and just about anything that one can touch. These are the assets of the organization that have
been entrusted to the care of the network or IT department for the benefit of the company. These
assets must be managed.
In many cases, the finance department will insist that you use their fixed asset system or, at the
very least, provide your data in a manner that can be input to theirs electronically. The problem
with using a generic system is that it often does not capture specific characteristics that are
desirable to have to manage your network properly. Information such as release levels, software
and firmware revisions, and other key characteristics are often lost. It is also true that many of
the automated tracking and inventory systems for networked components are not directly
compatible with generic organizational fixed asset systems and must be put through an
intermediate step, such as a Comma Separated Value (CSV) or Tab Separated Value (TSV) files,
Excel spreadsheet, and/or XML output.
If you have not inherited a fixed asset system nor have one already in place, you must procure
one as part of the project. The considerations for a fixed asset accounting system closely mirror
the considerations for any new software application. What are your basic system requirements?
Will the system run on servers with existing applications or on a different platform? What
hardware does your MIS or IT department recommend? What operating systems can be
supported? How many users will access the system and what are the security requirements? Will
shared access over the network be allowed? What additional user rights and permissions must be
configured and enabled to support this?
Will the new fixed asset accounting system be a separate, standalone system or an add-on
module to your current accounting system? If you require a standalone system, it is important to
understand the system’s interface capabilities. Will it interface with general ledger accounting
modules and tax applications to avoid duplicate entry of data? Does the system allow data import
and export? If so, what import and export formats does it support? If you require an add-on
module to your current accounting system, you must make sure that the add-on module will
interface with your current accounting system.




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What do you want in terms of reporting capabilities? Standard reports should include
depreciation, such as projected depreciation and depreciation method comparisons, to aid
planning. What kind of tracking does it do for financial reporting? Also be sure to investigate the
ad hoc reporting capabilities. Is there a limit to the number of user-definable reports you can
create? Is it difficult to create ad hoc reports?
What are your data entry and depreciation needs? Many packages offer features that significantly
reduce fixed asset entry and calculation complexities. Is the user interface easy to use? Does it
offer standard templates to ease data entry? Does it offer automatic data validation? Does it offer
automatic calculations on a periodic and daily basis? Does it allow you to customize the
information you want to enter? How many depreciation methods are supported?
What are your fixed asset tracking and management requirements? Whether you are a large or
small company, it is important to understand your asset tracking and management needs. Does it
keep physical track of where assets are located? Does it track multiple locations? Does it manage
multiple companies? Does it offer bar code capabilities? How does it track assets of mobile users
such as cell phones, PDAs, notebook computers, and similar devices that may be on the road as
much as their human users? All of these concerns, and more, must be addressed in close
collaboration with the accounting department.

Shipping, Warehousing, and Tracking
Shipping, warehousing, and tracking of assets are critical. In many cases, devices and systems
are ordered and shipped directly to the end user, very often being charged to the user’s personal
or company credit card and charged back to the company through an expense reporting and
reimbursement procedure. Are these assets tracked? What is the dollar value threshold for your
organization that must be tracked? How much money is effectively lost each year by not doing
proper tracking? Or maybe assets such as SIP phones or VoIP over WiFi devices are centrally
purchased and shipped to the user from a warehouse. How are these assets tagged and tracked?

Repair/Refurbishment/Return to Service/End-of-Life
Another area of consideration is repair, refurbishment, and return to service. How is this tracked?
Are users provided with a loan or replacement system? How often are systems refurbished, either
via software refresh and upgrade or cosmetically? What happens to assets at the end of their
accounting life? Are they sold, given to schools or charities, or recycled?




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Administration
In most organizations, operations consist of some level of operations management, often, in
larger organizations, headed by a Vice President of Operations or similar function. Below that
rectangle on the organizational chart, the line usually splits into at least two directions. One
direction is the operations group of engineers, technicians, and often Help desk people who
manage the technical operations and keep the hardware, software, and network running. The
other line goes to a group of administrative people who do the paperwork, manage the budgets
and inventories, and, basically, keep the group running that keeps the services running. There are
many different functions of that second group, but we will consider some of the primary
functions here that may be somewhat different or have some unique characteristics for Voice
over Packet networks.

End-User Cost Allocation
I well remember the first time I heard the term “funny money.” Besides being a two-word poem
it caused me to ponder what was so funny about it. Upon further inquiry I learned that computer
usage—at the time access to a mainframe computer over 9.6 kbps leased lines—was used to
allocate costs to departments but that paper checks were never written nor did real money change
hands. The transactions were purely bookkeeping transactions but, to the department heads, were
real nonetheless. If they had a surplus of money in their budget, they might not get the full
allocation next year; if they ran out of money, they might not be able to access the computer
system, and so on—normal budgeting issues. Many organizations still track usage of telephony
systems, both the variety of usage that results in monthly charges and usage that is part of a fixed
cost system such as a private IP network or flat-rate VPN.
The most logical, and common, metric to track is MOU or Minutes of Use. This is the easiest to
track in most cases but might not provide a clear picture of the assets, or parts of shared assets
being consumed, such as bandwidth. Two other facets of the MOU tracking can be the quality, or
Class of Service (CoS), of the call and the bandwidth being consumed as these other factors
should have a “cost” associated with them. An additional reason to use MOU as the basic unit of
measure, at least while the telephony network is still a circuit/packet hybrid using both the VoIP
and PSTN networks, is that MOU will provide a common tracking and cost allocation metric that
is used across both telephony platforms.

Detailed Usage and Billing
There is an inclination when making the move to IPT, and VoIP specifically, to abandon many
prior financial controls and recordkeeping. Detailed usage and billing, however, is still required
in many organizations for needs that go far beyond “funny money” accounting needs. In the case
of professional services, detailed usage information supports client billing: not for the phone call
but rather for the professional services delivered over that phone call. Detailed usage is also used
to track sales progress or the efficiency of phone support, not to mention use as a management
tool to review how an employee spends their time and to question non-business use—an item
that is ever smaller in terms of network services costs but ever increasing in the cost of the
human time wasted.




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Vendor/MSP/Carrier Billing
An important area of any operation is ensuring the accuracy of bills, and nowhere is this more
important than in the area of recurring charges—those charges that come up each and every
month and must be paid over and over again. A problem with the bill rarely gets fixed and just
becomes a part of a bigger problem over time. It is critical, for budgetary purposes, to review all
vendor, managed service provider, and carrier bills each and every month against the orders for
the services that appear on those invoices. The same holds true for periodic equipment invoices,
though one-time invoices for purchases are more likely to be audited at the point of receipt of the
product.
In many instances, organizations use the move to a Voice over Packet system as an opportunity
to clean up their billing that has slowly become inaccurate over time. They consider the new
packet voice system as a clean piece of paper and meticulously account for each and every new
charge and track it. When employed by a new operations manager who is looking for a way to
sort out the mess they have inherited from a predecessor, this approach has a reasonable chance
of succeeding, assuming the new operations manager has the discipline to keep the new system
updated. When used by an existing manager who created the mess in the first place, this is often
a formula for assured failure: if this manager could not manage the old system what changes are
being made to ensure that they can manage the new system?

Cost Reduction and Network Utilization Maximization
The job of looking for ways to reduce costs and maximize network utilization is the task of
network operations. The job of putting those ways into production and modifying the underlying
operations procedures is the job of network optimization and is covered in the next chapter.


Provisioning
Traditional telephony operates on a system of OAP&M—Operations, Administration,
Provisioning, and Maintenance. In effect, these procedural steps are outlined in this operations
chapter. The “P” part of the task—provisioning—includes setting up, configuring, installing,
testing, and “turning up” any new service or feature a user may need. This includes the initial
installation; any subsequent moves, adds, and changes; and the refresh of the technology and
audit of the installed system.
Initial installation includes proper configuration, verification, and testing and should also include
a security assessment to ensure that installation of the system is proper from a security standpoint
and does not introduce any security vulnerabilities into the system. Initial installation should also
include user training and execution of any asset tracking documents or other financial
recordkeeping.




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Maintenance
Once the system and its components have been placed into service, the maintenance cycle
begins. Maintenance of hardware, software, and QoS and user QoE levels is a key function of
ongoing operations. Basic maintenance functions include managing moves, adds, and changes
and handling technology refresh and periodic audits.

Moves, Adds, and Changes
Moves, adds, and changes (MAC), including de-installation and de-commissioning of
equipment, are the things that absorb a large part—in some cases, the majority—of the
operations budget of any organization. VoIP systems, using Dynamic Host Configuration
Protocol (DHCP) and other dynamic configuration options, were supposed to solve the MAC
problem and allow users to easily relocate to any location where Internet access was available—
inside or outside the company—and to simply connect. Although users of public VoIP services
have, in fact, achieved this level of dynamic connection, many within the enterprise and agency
networks have failed to do so. Inside the walls of the organization, many times, are internal
defensive mechanisms—firewalls and similar systems—deployed in a defense-in-depth
arrangement that often prevents easy, user-initiated MAC. An additional tool, but one that also
adds complexity, is a VPN, such as those based on wide area Ethernet services and MPLS, and
Virtual LANs (VLANs) to carry multimedia traffic. All of these areas must be considered in the
light of your specific implementation and security needs.

Technology Refresh
Periodically, a technology refresh will be required. This may be as simple as ensuring that
software and firmware are up to the most current release level or may require hardware upgrades
or replacement as well. The rate of technology refresh should not be based on the rate of change
of the technology or availability of new features but, rather, should be based on the need for the
new technologies or features. For example, if it has been determined that a specific profile of
VoIP user can get all the functionality that they need from plugging their old desk phones into
Analog Terminal Adapters (ATAs) but new, low-cost SIP phones become available, the
consideration to replace the ATAs should be relative to the basic operations needed by that
profile and not based upon the additional features, however desirable they appear to be. A refresh
is just that—refreshing capabilities that exist and have been previously approved not an upgrade
or adding additional functionality.

Audits
Periodic audits must occur to verify both functions and assets in place. Audits should include
functionality, security, hardware, software, services, and circuits. A comprehensive periodic
audit, aided by automated auditing tools, should be performed independently of the requirements
of the finance department. Audits of Voice over Packet systems should be sure to include all
components and end systems, including those possessed by mobile users and users in their small
office/home office environment.




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Operations Tools
Decisions must be made about the tools needed, the amount of graphical reporting required and
the testing methodology. Whether this is active and non-intrusive during the call, or intrusive and
separate from live traffic as well as tracking relative to the SLA and/or other baselines, and the
amount of integration that is desirable.

Integration
Generally speaking, the complete operations picture will require several tools—both on the
technical operations and financial/budgeting sides. Some degree of integration should be
expected between the different tools and may require some custom development work in order to
get a full, comprehensive solution. This should be anticipated and worked into any plan that is
developed.

Operation
Ongoing operation of network monitoring and management tools will require proper budgeting
for training and planning to ensure that qualified personnel are available to use the systems and
will most likely be integrated with ongoing network operations. The special training and skills
needed to get the most out of the system will be the biggest concern. Besides considering
employees, this is also a task that can be outsourced to a third party and managed by the
organization.


Summary
Chances are that if you are responsible for the ongoing operation of the Voice over Packet
network and its associated services, you have inherited something that is a bit chaotic and not
quite fully formed. That is the nature of new things. If the job has been done correctly, many of
the important operational considerations—baselines, SLAs, accounting, and asset tracking
systems—will have been made and some initial steps taken to equip you to do your job. If not,
the scope of your job is bigger.
The job of the operations group is less glamorous in many ways than the job of designing and
procuring a new system, wrestling it into place, and making it work—but is no less important. It
is the job of the operations group to ensure that the system performs to specifications
consistently—to keep humming along and doing its job day and night, weekdays, weekends, and
holidays without glitches or errors, through upgrades and surges in user demand.
The next chapter will consider the task of network optimization: improving the network
operation in some way and providing new baselines and operational guidelines for the operations
group to maintain in, hopefully, a never-ending loop of improvement—operations, improvement,
operations…on and on and on.




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Chapter 7: Optimization
This chapter discusses the process of reviewing the performance of your IPT system and setting
new standards and benchmarks. In other words, taking what is already a world-class system and
making it even better. This chapter will make the distinction between operations, the objective of
which is to keep things as they are, and optimization, the objective of which is to modify the
telephony solution in a manner as to improve its operational characteristics.
The chapter will also make a distinction between optimization and troubleshooting and repair,
the objective of which is to make something that is not operating or not operating within
guidelines to work once again according to the guidelines. In fact, the underlying assumption of
troubleshooting and repair is that some capability, system, or feature operated properly at one
time and that something has caused it to no longer work as desired. This differs from
optimization in that operation in an optimized state is desirable but the system, feature, or
capability never operated within that state before but will after the optimization. In fact, the
optimized condition will become the new baseline for proper operation and the next review cycle
may improve further on prior work in a never-ending loop of improvement.


SLAs and the Scope of Optimization
Because the SLA represents the classes of services that will be provided by the network and
because it was, or at least should have been, the focal point of formalization of the needs for the
network and IPT services, the discussion will return once again to the SLA. Many of the most
important optimization topics can be addressed via the SLA, and those that cannot, become
candidates to upgrade the SLA to meet current and anticipated needs.




Figure 7.1: Scope of optimization and the SLA.

When SLAs were originally introduced, they were a tool of traditional service providers that
offered services such as Frame Relay, ATM, and Internet access. The “service” for which the
level was agreed to was a network-based service. Therefore, the points between which the SLA
was defined were within the range of control of the service provider. It was only the rare case in
the early days of SLAs when an SLA included the access circuits: SLAs were almost universally
measured within “the cloud,” as Figure 7.1 shows.




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The view of the SLA covering just the Layer-2 aspects of network services has undergone quite a
bit of change since the early days. SLAs today are, increasingly, service-aware IP SLAs or even
Multiprotocol Label Switching (MPLS) SLAs that add value to the IP infrastructure. What is
required for a comprehensive and effective approach to optimization is to extend the scope of the
SLA. The new definition, the domain of optimization, should not include only the Layer-2
services—which, when strung together end to end, create a path over which applications bits
flow—but also the emerging concept of the application-aware IP SLA and a new, user-centered
view of the end-to-end service. In this manner, the concept of the SLA is being maintained,
while the scope is expanding to meet the real, service-oriented needs of optimization so that you
can maintain alignment of service-level objectives and the SLA.
There is an entire industry dedicated to application monitoring and optimization. This industry is
focused on the use of the actual application such as SAP. There is also an entire industry
dedicated to monitoring and optimizing the packets, frames and, ultimately, bits, that flow over
circuits. What is being proposed is something somewhere between the optimization of network
services, in our case IPT, including the operation of all signaling and other associated functions
that support the connection, transmission, and disconnection of packet voice calls.


The Optimization Cycle
By this point, you have set up a world-class packet telephony network, you have assured that, at
the very least, the telephony service you are providing meets users’ quality expectations and
delivers the same basic services as the traditional telephony network it is replacing. It is also
quite possible that you have added a number of important features and applications, such as
unified communications, to the new network. The new applications make the users more
effective in their jobs or make special new capabilities available, which is the real reason to go to
all the trouble of implementing a new network. After all, they already had dial tone. By this
point, you have also done all the broad, sweeping, system-wide things that can be done to ensure
proper operation of your packet telephony services. It is, therefore, time to take a closer look at a
number of very specific areas and to optimize what you have created.
Optimization is an ongoing process that is required to keep network costs in check and to avoid
the demands of the users running ahead of the capabilities and budgets of the network. IPT
networks have a tendency to consume huge amounts of resources and must be continually
managed. Once the network is in place and IP-tone has replaced dial tone, the next step might be
static image cameras on phones and then full motion, ad hoc, video conferencing, also referred to
as video-tone. Who drives the changes? How much control does the internal or external service
provider have? To what extent is the SLA considered? Is the SLA changed to reflect new needs
or are the new, emerging requirements somehow shoe-horned into current categories?
Independent of the answers to these questions, the network and its services must constantly be
optimized, and, beyond the administrative questions raised here, the optimization process will be
the subject of this chapter.




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Figure 7.2: The IPT optimization cycle.



The Use of Software Tools
The software tools that will be employed for optimization include both the measurement and
monitoring tools used in day-to-day operations as well as specialized modeling and simulation
tools last employed at the time of the initial design process. The use of software tools is
emphasized within the optimization phase due to the ubiquity of the tools within the network—
the fact that they are constantly testing and monitoring many aspects of system operation, the
repeatability of the tests and the ability to play “what if” games and simulate different scenarios
in order to understand optimization options prior to putting them into operation.

Tool Ubiquity
Software tools can monitor a complex set of metrics across a network simultaneously. The
implementation of monitoring tools must, of course, have been accomplished during the
implementation phase as retro-fitting an active network with monitoring tools is far more
complex, time consuming, and expensive than doing so up front. However, if the proper tools are
not in place, it is time to circle back and put them in place. Both intrusive, often called active,
and non-intrusive, often called passive, testing will be employed and will provide baseline
operational data on the aspects of IPT system operation that will be employed. It is also
important to have underlying network infrastructure statistics; however, the two families of tools
that you need should be clearly understood and their roles defined.
On the one hand, specialized tools are required that monitor specific characteristics of real-time
services; in this case, IPT. For IPT, the types of statistics that must be monitored include
calculated MOS, delay, delay variation, E-Model R Value, and other metrics of importance
specifically to the real-time voice class of service. It is also necessary for these statistics to be
made available in a high-level summary with graphics—in effect, a simple network-wide report
card—as well as at the individual call level so that the impact of optimization decisions on
individual calls can be assessed.
On the other hand, you still need to understand the operational characteristics of the underlying
delivery system: the network. The second set of statistics you are seeking comes from traditional
network monitoring and management tools and includes availability of network components,
trunk congestion, router buffer and queue management information, and similar types of metrics.




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Constant Testing and Monitoring
Constant testing and monitoring is as important to ongoing operations as it is to optimization. In
operations, it is desirable to see how the network, and overlaying services such as IPT, is
performing in an attempt to ensure that they are performing within established guidelines.
Optimization seeks, in contrast, to improve the operational characteristics and, therefore, to reset
the baselines to higher standards. One of the aspects that constant testing and monitoring will
show is any trends in network operating characteristics that might highlight important changes
that must be considered within the context of the optimization process.


Exception Reporting and Filtering
A multimedia network of any size will generate far more information than can reasonably be
assimilated by administrators working without automated tools. For this reason, exception
reporting is used within the operations process to automate the function of identifying when
certain aspects of operations are above or below certain pre-set thresholds. If, for instance,
packet loss for VoIP connections exceeds a preset threshold of 3.5%, the operations personnel
are alerted of an exception condition. This can be done through a variety of mechanisms, such as
SNMP traps that cause red dots to appear on the screens of network operators to emails to
beepers or IM messages to key operational personnel—and all this is done in real time.
Because optimization is not done in real time and, in fact, uses historical data, there is not
“exception reporting,” per se, in optimization. The optimization equivalent of exception
reporting is filtering. Software tools allow optimization engineers to sift through mountains of
data using specific filters to identify exceptions and trends and any specific areas in which
anomalies might exist that can be exploited by optimization.

“What If” Scenarios
The natural outgrowth of being able to work with mountains of real data from a real, working
network is the ability to perform “what if” scenarios that will have outcomes that much more
closely match the real operating environment. Having real operational data allows far more
accurate prediction than was possible during the design phase of the project and, in fact, almost
certainly means that the first optimization phase or two will yield much better results than
subsequent optimizations. During the design phase, inputs to the “what if” process were either
simulated traffic—because no similar real network previously existed from which to capture
traffic—or was based upon an IP network that existed before the current network but supported
data only and has been substantially changed in the process of implementing the new IPT
solution.
“What if” scenarios can actually be executed using one or both of two approaches. Ideally, both
approaches will be used. The first approach for the “what if” exercise is to have a specific list of
characteristics, mostly encompassing the SLA metrics and possibly a few others, and to be
looking specifically for ways to improve those metrics. This is the most common approach. The
second approach is to rummage around in the historical network performance data looking for
optimization opportunities. This second direction will often yield surprising and good results. In
this second approach, it is also important to keep an open mind to anything you might find and,
in fact, to discipline yourself and your team not to try to guess at or pre-determine the outcome.


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Trending and Capacity Planning
Optimization includes trending and capacity planning and extends to minimizing infrastructure
costs, such as decommissioning excess gateways, lines, and similar un-needed items;
maximizing call success rates, security, business continuity; and future proofing the network.
Some of the areas to which optimization may most productively be applied are:
   •   Bandwidth
   •   Ports and lines
   •   Phone numbers and numbering plan
   •   Grooming
   •   VLAN capacity
   •   VPN capacity
   •   Infrastructure
Let’s take a closer look at each category and their interdependencies.

Bandwidth
Bandwidth is as inexpensive as it has ever been and the price keeps going down; it is worthwhile
to invest in bandwidth to ensure a higher voice quality. But, even considering these two points,
there is no reason to waste bandwidth. The optimization phase is the time to reconsider several
options relative to bandwidth. The first item to reconsider is the choice of coding scheme. Pulse
Code Modulation (PCM) has been strongly recommended throughout this guide as the best
choice for a variety of reasons, but PCM also requires the most bandwidth of any of the voice
coding choices.
The optimization process should begin by reviewing the reasons for choosing the specific voice
coding scheme currently being used and determining whether it is still the proper choice. In this
example, let’s assume that this is the first optimization and that PCM was chosen. What were the
reasons for choosing PCM coding for voice? Are those reasons still valid? The first reason is that
PCM delivers the best voice quality under a variety of impairments and network conditions.
PCM translates sound, not just voice, into ones and zeros for transmission across a digital circuit
or packet network and will deliver a voice and “sound” quality that is as near as possible to the
circuit system being replaced. PCM-based systems will often deliver a voice quality in the 4.2 to
4.4 range on a 5.0 MOS scale, within the target range for voice quality over a packet network.




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Once the transition is made from circuit PCM to packet PCM, it should be easier to wean users
over to other, less traditional-sounding codecs. This process can be accomplished slowly over
time; it would be very difficult to do all at once. It is also true that newer digital codecs,
particularly broadband codecs such as Global IP Sounds (GIPS), designed specifically for use in
packet networks are improving constantly both in terms of better MOS and lower bandwidth
utilization.
After considering the effect of codecs on the bandwidth of individual connections, especially on
access or LAN connections, the cumulative impact on shared connections, such as shared
network access and backbones, must be considered. A small change on individual user
connections, such as their Virtual LAN (VLAN) connection, may seem insignificant in the
bigger picture but the cumulative impact can be profound when hundreds or thousands of
simultaneous connections are considered. Also remember that the major goal of broadband
systems, after a large enough pipe has been provided, is enhanced performance and that the
instantaneous performance of a broadband system is enhanced by more smaller packets than by
fewer larger packets. In addition, although this is at odds with conventional LAN thinking—a
frame of reference built on large quantities of cheap bandwidth going short distances—it is a
guiding principle for this IPT deployment, especially if the communicating endpoints are
somewhat distant.

Ports/Lines
Ports and lines are two hardware units of measure—not to be confused with logical Layer-4
ports—that must be optimized. They are also two terms that have different meanings based on
context; the two possible contexts being traditional telephony and IPT.
In the traditional telephony context, a port is a physical connection into which one would plug a
single telephone device or a device capable of connecting single telephone devices in groups of 1
to 24 per physical port. As the use of IPT grows, the use of traditional telephony ports should
decline, though they may not completely go away. It is possible, for instance, that a traditional
phone service may be maintained for faxing and/or for use with 9-1-1 calls. One such situation
exists at Pauling County School District in Georgia and is detailed in the following sidebar.




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                                   VoIP, 9-1-1, and Port Optimization
Before VoIP, telephone numbers have corresponded to a specific physical location within a customer
premise, which had a one-to-one correlation to a physical location within the telephone company Central
Office (CO) known as a wire center. Each wire center has specific geographic coordinates and a
corresponding Public Safety Answering Point (PSAP) from which emergency services such as fire, police,
and ambulance are dispatched when a citizen dials 9-1-1.
Since the “break up” of the monopoly phone companies, many competitors have entered the market. And
there may not be enough telephone numbers in their geographical areas of operation without purchasing
another large block of phone numbers that may go unused. The solution? Assign virtual phone numbers
from other areas that do not correspond to nearby wire centers. This is great for the new VoIP phone
company, but this arrangement breaks the traditional pairings of wire centers and PSAPs. Is this a
problem? For cell phones, no. Cell phones are mobile and can be anywhere. Cell phones also use a
different system to report their whereabouts to emergency services. However, this setup can be a big
problem for those who must rely on 9-1-1, such as schools and businesses that may need to summon
help. In many cases, new VoIP users, especially residential and small business users, may not realize
that emergency services will be dispatched by 9-1-1 from dozens or even hundreds of miles away…until it
is too late! Workarounds exist, but 9-1-1 must be tested while setting up new VoIP systems. A county
school district in Georgia solved this problem; let’s explore the workarounds they developed.
Before the implementation of their new VoIP system, Paulding County School District had purchased their
voice services exclusively from their Regional Bell Operating Company (RBOC) BellSouth, now AT&T. As
a part of the complete re-evaluation of their network, Paulding County’s Chief Technology Officer (CTO)
reviewed alternatives and chose USLEC, an alternative carrier offering lower pricing, better installation
lead times, and the full range of services available from AT&T but at better prices. The CTO also found
the USLEC account team to be more cooperative in handling deadlines and the exceptions that come up
during the cutover of a network his size. And, with seven special T1s – ISDN Primary Rate Interface (PRI)
lines, Paulding County was an important customer to USLEC but was very much the “small fry” with
AT&T. One problem that was encountered during the cutover was with 9-1-1. As a new telecom carrier,
USLEC was assigned a group of numbers to provide to their customers that had previously been
geographically associated with Fayetteville, Georgia, some distance away.
Ordinarily, the intelligent call routing inside the USLEC network gets all of Paulding County schools’ calls
to their proper destinations, but if Paulding County schools dialed 9-1-1, the emergency services would
be dispatched by the Fayetteville PSAP and, in many cases, may take 45 minutes or more to arrive at the
school. This was clearly not acceptable.
The solution that was adopted was both clever and effective. Each school was to have a FAX/9-1-1 line
and because AT&T held the numbers that would allow proper dispatch of emergency services from the
proper PSAP, Paulding County had to order the FAX/9-1-1 lines from AT&T. And they did. But, Paulding
County wanted USLEC to provide all their voice services. Thus, once the lines were installed and running,
Paulding County requested that AT&T transfer the service, with the corresponding correct number for 9-1-
1 dispatch, over to USLEC. This is a process known as “porting” and one with which BellSouth is legally
bound to comply in less than 30 days from the date of the request.
From the standpoint of optimization, special procedures should be put into place to ensure that the
numbers going to special phone numbers or lines, such as traditional analog fax or 9-1-1 lines, not be
changed, reconfigured, or optimized in any way, including changing the way the calls are routed, their
codecs, or other operational characteristics.




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Just as the term port has a special meaning in the traditional telephony context so, too, does the
term line. A line is also a physical thing and consists of a two-wire or four-wire twisted pair
connection between a subscriber premise—a residential or business location—and the telephone
company. It is critical that lines be removed, a common term is “retired,” when they are no
longer needed because there is a monthly charge per line that does not always magically
disappear from the traditional telephony bill as soon as they are retired. In fact, there are auditing
services companies and consultants who do no more than audit the lines that are being used
versus the lines are being charged and splitting the savings for the first month or so with the
client. This is a big and time-consuming task for an organization but one that must be performed
by someone. It is also noteworthy that in many cases, especially if a traditional phone company
is being replaced by the organization’s internal IPT service or by a competing VoIP service
provider, that they are often remiss in their duty of taking items off the bill and often do not
allow retroactive credits for items erroneously charged. Auditing and removing unused lines
from the phone bill is a critical element in any optimization exercise and one which, in many
cases, automated auditing tools can help.
In the VoIP context, the terms port and line have their own meaning. In the VoIP context, a port
is a physical thing, usually an Ethernet port, to which a VoIP phone or other system is attached.
The idea of a line in the vocabulary of IPT is a bit less formal. Generally speaking, from a VoIP
point of view, optimization of ports has two aspects. The first is that there be enough physical
ports on Ethernet devices such as VLAN switches to accommodate the growing population of
Ethernet-connected devices. The second thing to monitor is related to bandwidth: to ensure that
each of the ports has sufficient bandwidth to give the call and voice quality needed, especially
when aggregated with other traditional data, emerging video, and increasingly common Unified
Communications demands.

Phone Numbers/Numbering Plan
Although the world of IPT is designed to work without traditional phone numbers (instead with
alphanumeric Uniform Resource Identifiers), phone numbers will persist until all ties have been
cut with traditional telephony and are on IPT-only telephony infrastructures. Experts suggest that
that day is a long way off. In the mean time, optimization of phone numbers and auditing of the
numbering plan ensures three things:
   •   There is a sufficient inventory of new telephone numbers to be assigned to new users.
   •   Numbers that are un-assigned are returned to the telephone number pool for
       reassignment.
   •   Any internal numbering plan that is in place is being followed. Internal numbering plans
       are used as often for accounting and cost allocation purposes as they are for geographical
       number routing.
Each of these steps should occur regularly but should be audited and verified as a part of the
optimization exercise. This is another area in which automated tools and log analyzers can assist
in the hunt for missing or improperly assigned telephone numbers.




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Grooming
Grooming is an old term from the fading era of traditional channelized TDM telephony.
Grooming refers to ensuring that the least number of T1s and T3s are used in order to optimize
costs and network availability. In general, organizations have connected their traditional PBX or
other phone systems to the phone company’s Central Office switching equipment using T1s that
provide 1 to 24 individual voice channels or Integrated Services Digital Network (ISDN) lines,
which are specially formatted T1s that provide 1 to 23 channels usable for subscribers’ voice
connections. In today’s IPT world, channelized T1s still exist but, for the most part, connect to
softswitches, voice servers, Session Border Controllers (SBCs), and gateways rather than their
PBX counterparts.
Grooming is an optimization technique that is performed when you have two or more T1s or
their larger counterparts, T3s. Figure 7.3 shows the location of T1s or T3s used for telco network
access. Each T3, if they are all used, contains 28 T1s for a total of 24 × 28, or 672, voice
connections.




Figure 7.3: Traditional telephony interconnection.

The first consideration is redundancy and reliability. If this is a major consideration, you will
have pairs of T1s or T3s, each leaving the building through a different exit and, ideally, traveling
over a completely different path once they leave the building. It is even possible to have them
ultimately connect to two different central offices of the incumbent phone company or two
different Points of Presence (POPs) of competing long-distance carriers. This is all to say that
during the grooming process, provisions should be made for redundancy and reliability.




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Having addressed reliability and redundancy, let’s move to the task of grooming T1s. Each
specific channel of a T1 has one or more phone numbers associated with it. These may be
individual lines or may be grouped together in a “hunt group” that allows multiple phones to ring
in a round-robin cycle before the call is rolled to voicemail or some other method of handling. As
the phone numbers associated with the individual channels are moved to VoIP or in some other
way released, it is possible to regroup T1s and retire complete T1s. At that point, the billing
should cease and further cost savings will be realized.
How does this work? Well, let’s say that there are two T1s. One has 24 phone numbers assigned,
in the format XXX-YYY-ZZnn where XXX is the area code, YYY is the exchange, and ZZnn
represents the subscriber part of the number where nn goes from 01 to 24. Let’s say that the other
T1 uses the same setup, but the number is in the form XXX-AAA-ZZnn where AAA is a
different exchange. Due to IPT activity and number reassignment, 16 numbers are released on T1
number one and 10 numbers are released on T1 number two. That leaves 24-16, or 8 channels,
assigned on T1 number one and 24-10, or 14 channels, on T1 number two. 8 + 14 is only 22,
which is less than 24, so all the channels can now be combined on—groomed to—one of the
T1s. It is not important that the numbers are not numerically contiguous: the hunt group structure
takes care of that, if needed, otherwise it is not an issue. At this point, the T1s have been
groomed and optimized and one of them may be released from service.

Gateways
VoIP gateways, be they standalone or TDM ports on softswitches or VoIP servers, will be
connected to the traditional network and usage will, most likely, be increasing. Grooming,
therefore, as it relates to gateways and similar systems, will involve increasing capacity to meet
demands. There are many ways to calculate the proper sizing for TDM capacity, but the method
used by the phone company involves an Erlang calculation of the number of simultaneous
channels used for a population of potential callers based upon the length of their calls and their
behavior if they can’t make a call. Erlang calculations also involve a determination of the
likelihood of blocking of calls using a unit called ‘P’ where P.1 means that 10% (10 in 100) calls
don’t get through during the busy hour, P.01 means that 1% (1 in 100) calls don’t get through,
and so on. The P calculations are intended to predict a Grade of Service (GoS) during the busy
hour, which is the time during which the network will be requested to make the maximum
number of simultaneous calls.

   Optimization of GoS is specifically addressed later in this chapter.




   Although the specifics of these calculations are outside the scope of this chapter, it is important to
   point out that these calculations should have been done at the time of the network design and
   validation and should be performed periodically for optimization.


TDM Grooming
TDM grooming will be a matter of reducing capacity, as described earlier, or entirely removing
T1s that are no longer needed because the old equipment is being decommissioned. Optimization
includes auditing the bill to ensure that they are removed.


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VLAN Capacity
The bandwidth and number of simultaneous connections on the VLAN must also be optimized.
The demand on the VLAN will be increasing as more and more IPT traffic is moved off the
traditional telephone system and over to the IPT system. In addition to bandwidth and
simultaneous calls, intermediate devices, in this case VLAN switches, should be monitored to
ensure that they have enough memory and processor capability for the increased load.

VPN Capacity
In conjunction with the internal or external carrier, the capacity of the virtual private WAN
network must be optimized. This is yet another area in which automated tools can be very
valuable. The specifics of the optimization will be different depending upon the specific
underlying technology of the VPN. For instance, a different set of metrics must be optimized if
the VPN is based upon MPLS technology than what would be done for a VPN based upon
Carrier Ethernet transport.

Infrastructure
In terms of optimization, the “infrastructure” is considered to be the hardware and software that
comprises the network. The transport bandwidth interconnecting the hardware components is
considered separately for a variety of reasons, the primary one being that each optimization
requires a fundamentally different skill set and set of automation tools.


Hardware
Hardware in the IPT sense potentially covers a lot of different components. Consider end-to-end
the devices that may be involved and the optimization effort to determine their continued
suitability for use in providing the IPT services of your organization. Consider first the wide
range of possible hardware devices, the range and diversity of which is only hinted at in Figure
7.4.




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Figure 7.4: Infrastructure in transition.

Beginning in the upper left corner is a device that might be a simple traditional telephone. It may
also be a more complicated Private Branch eXchange (PBX) phone. That device, be it a
mechanical traditional telephone with a keypad and ringer or of the more sophisticated PBX
variety, is connected to the network via an Analog Terminal Adapter. The ATA is connected to
an Ethernet network, which, in an organization of any size, employs Ethernet VLAN switching
and, most likely, traverses the WAN using one or a combination of VPN technologies. Then,
there are, potentially, VoIP servers, softswitches, feature servers, gateways, firewalls, SBCs,
NAT gateways and other devices. Each component that plays a role in the end-to-end connection
must be evaluated and optimized.

Repair
The optimization is at once both a technical and financial “tune up.” As such, repair procedures
must be reviewed for relevance and currency as well as for proficiency of the technicians. It is
also important, as a part of the optimization cycle, to audit the repair process to ensure that units
that have been repaired are being returned to service.




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Obsolescence/Refresh
Optimization also involves the decision to decommission certain devices or to refresh the
hardware. Specific guidelines should be in place to determine which devices will be
decommissioned, or possibly moved to another part of the network, and which will actually be
taken out of service. One effective way to implement a decommissioning policy is to take
equipment out of service and replace it with its upgraded counterpart when it comes in for repair.
This could, dependent upon the component, be less disruptive and less costly, especially for
components used by end users, such as IP phones.
                                    Managing Surplus Equipment
In many cases, especially during a migration from an older technology to a newer technology, there is a
great deal of value in the older equipment, especially in a large organization that has an extended
timeframe for migration. One of the primary reasons for a migration to a next-generation telephony
system—any next-generation system—for many company sites has nothing to do with advanced new
capabilities. The real reason for migration is manufacturer discontinuation of older systems and models.
When a manufacturer discontinues a product line, availability and rising cost of components for
maintenance, repair, and expansion of existing, older, vintage telephony systems becomes a problem. In
addition to purchasing upgrades and replacements on the secondary equipment, a manufacturer may
market a solution that can extend the lifeline of the older systems and create bottom-line benefits. For
example, take equipment that had been decommissioned during the implementation of the new system
and send it for refurbishing and placement back in service. This process recycles products either for
repair or upgrades, in other locations that are still using the older equipment.



Software
Because almost all the components of the IPT infrastructure are, to some extent, computers,
software maintenance, enhancement, upgrade, and the management of patches, licenses and
security is of the utmost importance and all these areas must be audited routinely. There is no
better time to accomplish this task than during the optimization phase.

Maintenance
Software maintenance is an ongoing, often evolving, process. Optimization of the software
maintenance process ensures that all maintenance is performed in the simplest, most secure
manner and utilizes as few human and network resources as possible.




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Enhancements
Enhancements are differentiated from upgrades in that enhancements bring additional
functionality and are voluntary. Upgrades are needed to keep software and firmware within a
common operating range and they are mandatory.
When enhancements are made available, the process should ensure that all users to whom the
enhancement apply are notified and given the opportunity to download it. Part of this process
should be to ensure that users to whom an enhancement does not apply are not “teased” with
knowledge of an enhancement they cannot get unless they are also allowed to change their
equipment type, OS, and so on.
The optimization process involves looking at enhancements that have been made available,
judging user acceptance, and making a determination about whether an enhancement will be
supported going forward. It is important to note that the decision to offer, or allow, any
enhancement should be made very carefully on the front end because all enhancements have
costs—resources such as memory, processing capacity, bandwidth, user training, support—and
that once a feature is offered, it is almost impossible to withdraw the offer or to withdraw support
for enhancements and features that are already deployed.

Upgrades
An upgrade is involuntary, so it is often given a lower priority by users, is disruptive, or may
change the way a system operates, or impact its ease of use. Optimization of upgrades involves
first auditing to ensure that all pertinent upgrades have been made and second to assess the
impact of the upgrades. Another factor that should be considered as part of upgrade optimization
is licensing. As licenses can be either per end-point or time based an audit should ensure that an
organization is not paying for unused licenses or breaching licensing agreements.

Patches and Security Audits
Periodically, manufacturers release software “band-aids,” small pieces of computer code that are
intended as a temporary fix to a problem, most often a security vulnerability. These “band-aids”
are called patches. Very often patches fix one problem or close one hole or vulnerability but
create others. The assumption will be made here that proper due diligence was applied to the
decision to apply or not apply certain patches and that the optimization process will audit to
ensure that all appropriate patches are applied and that they are still needed or, if they have been
superseded by upgrades, that the patch is no longer being applied and taking resources.




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SLA Improvements
One of the most important aspects of the initial guidelines and baselines in the SLA was to make
the transition from traditional telephony as seamless and easy as possible. Admittedly, this was
not really necessary for many of the users as their perception of QoE was forged more as a result
of recent emphasis on less-than-reliable “can you hear me now?” cell phones rather than
historically superior hard-wired land-line phones. Nevertheless, emphasizing quality of the new
telephony services relative to traditional “toll quality” is a safe bet as it will optimize the quality
of [user] experience for all users.
At this point in the life cycle of the IPT system, the baselines of the new network relative to all
of the important SLA metrics—delay, delay variation, packet loss, and availability—are the ones
with which the users are familiar day-to-day. The optimization effort should be to improve the
metrics but not so quickly as to cause users undue concern. One of the subtleties of “quality” is
not that “quality” represents “the best” but rather that to be considered of a high quality, a
product or service should be consistent. There is a lot to be said for the confidence one gets from
consistency. These observations provide guidelines for the rate of improvement in SLA metrics.
Keep in mind that “quality” is relative and that improvement, regardless how well intentioned,
represents change and that change, if noticed, always brings concern and sometimes stress and
resistance.
The following metrics, often tracked in the SLA and compliance process, are among the
measurements of the greatest importance and the ones where third-party measurement and
reporting tools can have the biggest impact. Beyond their use in the optimization cycle, the
regular tracking of these metrics and exception reporting when the associated targets are not met,
comprise the “dash board” or “control panel” of any modern multi-media network and its
associated services.

Delay
Delay impacts the user perception of quality in many ways. There are a variety of things that can
be done to reduce delay, or the perception of delay, and, therefore, positively impact the user’s
QoE and set new bars for future performance. Any call, regardless of whether it is a data call,
phone call, video connection, connection oriented, or connectionless, has three distinct phases:
call setup, information transfer, and call tear down. There are several points in the process for
optimization.




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Call Setup
On the downside, if call setup takes too long, an impatient user, more familiar with the near
instantaneous call setups of the traditional telephone network, might abandon a call and retry or
even attempt to place a service order. For this reason, there are certain standards that must be
applied. The first element is the time (or delay) to dial tone. If an audible dial tone, or some other
indication that a call can be placed, is not presented to the caller within a very short time, the call
may be abandoned. Beyond that, once the digits are pressed or the destination number otherwise
signaled to the system, the next delay is the call setup.
Call setup times can be impacted by a variety of factors that can be optimized. For instance, if
call servers are centralized, optimization can improve call setup times by increasing bandwidth
between the telephony device and the call server or optimizing the placement of other systems
that might contend with the call setup for resources. If, for instance, data servers and telephony
servers are on the same VLAN, they can be separated. It is also possible to further divide
telephony users into multiple VLANs or to upgrade telephony servers to ensure that they are
responding as quickly as possible to call setup requests.

During Call
Analysis of traffic patterns, endpoint pairs, time of day, and time of month variations and shifts
due to time zones will yield a lot of information about the consistency or lack of consistency of
performance during a call. For instance, using automated tools to gather and compare delay,
delay variation, and loss statistics for all calls and then plotting them graphically will show times
when variations are most likely to occur and the causes of variations can be isolated and
optimized. This is an important part of any ongoing optimization effort and, if done properly, can
spot problems before they are reported by users.

Call Tear Down/Release
Once the call is completed, the call is released. This is an area of optimization that is most
important if calls will be serial, such as in out-bound call centers. If a call is terminated and then
the telephony user goes about non-telephony tasks, this delay is buried in the network, invisible
to the user and, therefore, far less important to optimize. If, however, one call ends and another
call must be made immediately, the call release time becomes important. The good news is that
minimizing call setup time and performance during the call will usually have a positive impact
on the call tear down delay because the entire end-to-end process has been optimized.

Delay Variation
Delay variation, also known as jitter, is greatly impacted by other traffic. As with delay
minimization discussed earlier, delay variation should be tracked regularly and any variations
outside of a set range should be reported and researched immediately, independent of the
optimization cycle. But, as a part of the optimization cycle, delay variation must also be
analyzed, trends noted, and improvements made. The same process applied for delay
minimization will yield improvements in delay variation.




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Sample and Packet Loss
Sample loss, defined as loss of the actual content representing human speech being transmitted,
is one of the network impairments that will have the greatest audible impact on voice quality. If
human reports or the results from automated measurement tools indicate that the observed or
calculated MOS due to sample loss is outside of established benchmarks, the problem should be
addressed through troubleshooting and repair. The optimization process, however, should
establish new standards for packet loss.
Sample loss and packet loss are very closely related and should be optimized together. Sample
loss refers specifically to the loss of samples of human speech that are created at the speaker’s
end of the call and must arrive at the listener’s end and be played out as a part of the call. Packet
loss refers to the loss of the protocol packets containing the human speech samples. If each
packet contains a single speech sample, sample loss and packet loss are one and the same.
However, for purposes of bandwidth efficiency, it is common that more than one speech sample
be placed in each packet. The first task of optimization, therefore, will be to determine the
settings for the network being optimized and to understand what, if anything, can be done to
improve the current settings.
Research has shown that humans can tolerate a sample loss of up to 10% and in fact this may
have been the original benchmark for the network, especially if the IPT system user’s frame of
reference for voice quality was cell phones. In any case, the goal of optimization is to analyze the
loss, optimize the transport network and its components, and to establish new benchmarks.

Availability
Availability should be tracked regularly by operations personnel and will usually be shown as a
general network availability figure, unless specialty tools are used to monitor availability of IPT
as a business service. Basic tools that use simple network functions such as PING will provide
sufficiently accurate availability information for IP networks. Although this is fine for most
needs, it is insufficient for purposes of optimization of IPT services. For IPT services,
availability should go beyond the ability to physically connect to the network and should include
the simultaneous availability of all constituent components needed for the end-to-end call, call
server, and so on. In other words, the availability of all individual components is important in the
context of the availability of the telephony service as was discussed in the section earlier titled
Scope of Optimization. Automated tools are very important in assessing service availability and
isolating the specific components responsible for any degradation of availability.




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Telephony Service Availability
Beyond the availability of the IPT network and its IP infrastructure, it is critical to monitor and
optimize the availability of the traditional telephone service. This usually means the connections
to the PSTN.
IP networks and traditional telephone networks are different in many regards, one of the most
profound being the way in which each handles offered load. In the case of IP networks, which
operate on a “best effort” basis, additional load is accepted and, in the absence of the more
sophisticated QoS mechanisms that are just now being implemented in IP networks, connections
compete for network resources on an equal basis. Basically, IP allows packets to fight it out
among themselves on a level playing field.
Telephony networks have specific, clearly identified TDM channels that may be occupied by one
and only one call at a time. And, with the exception of certain military and public safety systems,
the rule is “first come first served.” The metric within IP networks that must be optimized is
Quality of Service (QoS). The metric within the traditional telephony network that must be
optimized is Grade of Service (GoS).

GoS
GoS represents the probability that an attempted call will get through. A typical target used by
telephone companies is P.02, which means that only two calls out of every hundred attempted
during the busy hour would not go through. Depending upon the number of people who might
place calls, the length of the calls they might place, the number of telephone devices, and the
person’s behavior if their call does not go through, GoS creates some level of oversubscription.
That is to say, GoS measurements let telephone companies provide fewer actual dial tones than
there are telephone devices because, except in certain rare cases, all the phones will not be in use
simultaneously. This design approach is called a blocking system because there is a statistical
probability that under certain circumstances, which are specifically predicted and allowed for in
the design, calls may not be connected. Just what is meant by the “busy hour” and the calls
attempted during that time is detailed in the following sidebar.




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                                        The Busy Hour and BHCAs
The “busy hour” is a somewhat mythical and mystical thing referred to by traditional telephony so
frequently and so casually that an outsider might draw the conclusion that it actually exists. In fact, there
are many busy hours. Because of the natural ebb and flow of the human daily cycle, the first busy hour,
both for phone and data, is roughly 10 am to 11 am, plus or minus, local time. This is referred to more
specifically as the morning busy hour. The afternoon busy hour occurs at roughly 1:30 to 2:30 pm local
time. The evening busy hour occurs at approximately 7 to 8 pm local time. Many things can impact the
busy hour.
To begin with, busy hours tend to have more calls occurring in a smaller time window if the calls are
between points in the same time zone. Busy hour calls between points in adjacent time zones have the
effect of having fewer simultaneous calls. Busy hour calls between points in non-adjacent time zones
have the effect of yet fewer simultaneous calls. It is also true that the morning and afternoon busy hours
will be of greater interest for business telephony systems while residential telephony systems will focus
more on the evening busy hour. It is also true that calls between residential and business systems will
more likely occur during the morning and afternoon busy hours. Work at home and Small Office /Home
Office (SOHO) will also have an impact on call distribution. All of these variations should be taken into
account when considering your own busy hour for optimization and planning purposes.
Another important consideration is Busy Hour Call Attempts (BHCAs). The number of simultaneous calls
traversing a gateway, for instance, can be very large and is limited only by the number of TDM circuits
interconnecting the gateway to a PBX, PSTN, or a long-distance carrier and the amount of memory and
table space in the gateway. The amount of resources taken for converting IP call packets to TDM bits is
small compared with the amount of processing resources for setting up calls and maintaining call state
information. For this reason, many times gateways can handle a large number of calls of longer duration
than they can shorter calls requiring thousands of call setups per second. Most enterprise gateways, and,
in fact, many service provider and carrier-class gateways, are not capable of handing the BHCAs needed
for large networks. BHCAs are another very important measurement and optimization number.

If the IPT system being optimized is trending toward more on-net calls, meaning more on IP net
calls, then GoS on gateways connecting the IP network to the traditional telephone network will
be continually improving because the number of simultaneous calls will be decreasing. In this
case, the optimization process will be a financial one: decommissioning lines and optimizing
access T1s as previously discussed. However, if growth in IPT users means that more calls are
flowing across the gateway between the IP network and the PSTN, the number of traditional
circuits interconnecting the two will need to be increased.

Blocking/Non Blocking Access
Ideally, the number of circuits available between the IP and traditional parts of a network should
be such that call attempts are never blocked at the gateway. Non-blocking access as this is called,
may be somewhat expensive to provide under normal circumstances but is a default situation
under others. For example, if there are 12 total phones on the IP network and the IP network is
connected to the PSTN with a T1, the interface is, by default, non-blocking because the T1
allows 24 simultaneous connections and only 12, at most, will ever be needed. However, if the
number of phones exceeded 24 and a T1 were used, there would be a potential for calls to be
blocked. If for instance, there were 25 phones and each caller wanted to go through the gateway
simultaneously and make a call to a destination on the PSTN, one of them would not be allowed
through. However, it is likely that someone might not be in the office, that they don’t all need to
be on the phone simultaneously, or that one or more people within the office might be calling
each other and, therefore, not going across the gateway to the PSTN.


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Success Rates of Call Setup
IPT gateway systems and their PSTN counterparts on the other side of the connection have the
ability to report the success rates of call setups and to distinguish between call attempts and call
completions. These are metrics that should be monitored and optimized as they are a key
component of the overall IPT QoS delivered to the caller. Erlang B calculations are used to
estimate the actual number of interconnect circuits needed, but observed statistics should form
the basis of the actual number installed.

Related Gateway Issues
Not only are circuit connections required between gateways and the PSTN or other TDM devices
such as PBXs but also the gateway must be sized to handle the call volumes. BHCAs should be
closely monitored and the gateway’s resources should be sufficient to ensure that BHCAs will be
maximized and the number of blocked calls will be minimized. It is also important to note the
distinct differentiation between calls that are intentionally blocked, and reported as such, due to
lack of available circuit connections and those that are lost into the black hole of telephony due
to lack of memory, buffers, or processor capacity.


Prices and Costs
Cost reduction must certainly be a part of each and every organization’s optimization exercise.
To clearly understand the component elements for which budget dollars are extended, this
optimization should be divided into hardware costs and service and support costs.

Hardware Costs
To some extent as a result of Moore’s Law—the observation made in the mid 1960s by Intel
cofounder Gordon Moore that products based on computer chips have a predictable increase in
capability while their cost goes down—end-user organizations have an expectation that hardware
prices will continue to decline. To some extent, this is a realistic expectation, largely because
they have and, in the hardware cost category, at least, they should continue to. Keep in mind,
however, that declines are percentages of the new cost, not the original cost, so while there will
be continuing decreases they will, in actual dollars, be less and less.
The specific prediction upon which Moore’s Law is based says that capacity will double and cost
will be cut in half approximately every 14 to 18 months. What this means in real terms is that
something that cost $100 in the year 2000 will cost $1 in 2010. So what does all this mean in
terms of actual hardware costs? In terms of the cost of the raw hardware—purchased as a
commodity item in an open market and priced independently from any support services—costs
will continue to come down. Larger organizations, who have a lot of clout as a result of their
total dollar expenditures, often write a 20% year-over-year decline in prices into multi-year
purchase contracts. In fact, the sellers are usually happy to have such contracts in place and are
willing to include such clauses to reduce the chances of losing the business to a competitor and
to lower their costs of constantly selling to the large account.




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Hardware should originally have been chosen to yield the greatest possible Return on Investment
(ROI) and lowest Total Cost of Ownership (TCO). In other words, it should originally have been
selected not only for the lowest price but also for the longest service life, greatest upgrade
potential, optimum ease of use and management, and a variety of related factors. Hardware
optimization should first consider the extent to which technology refresh or upgrades could be
done using existing equipment before purchases of new hardware are considered.
One example of these considerations for an IPT environment is in the choice of user connectivity
options. When the IPT service was originally installed, for instance, software upgradeable SIP
phones may have been considered and compared with the less expensive option of using a low-
cost ATA and keeping the existing traditional analog phone. At the point when that consideration
was first made, it is possible that the cost of the software upgradeable SIP phone exceeded $600
each while the cost of using an ATA with the existing phone was $120. Now that it is time to
optimize the hardware, it might be prudent to review the earlier decision and consider the impact
of time on costs. What you might find is that, due in part to Moore’s Law and in part to
competitive and market pressures, SIP phones costing around $100 are now available and, even
though ATAs now cost less than $40, it might be wise to begin using SIP phones—even if not a
complete replacement, at least for new service and to replace the older ATAs when they come
back in for repair.

Service and Support Costs
Anecdotal evidence tells of organizations that are so aware of the constant decrease in prices of
technology that they never make a purchase decision for fear of not getting the best deal. This
statement may be a bit extreme, but is really not far from the truth in many cases. As true as this
constant reduction of price, with increased functionality, is in hardware, the opposite is true in
service and support costs.
Service and support costs are based upon the cost of human resources. Savings have been
achieved in the past, but service levels have often suffered, by tapping workers in India,
Pakistan, Ireland, the Philippines, China, and elsewhere, but even now there is an increase in the
price of the best of these resources accompanied by the desire to improve service levels. A
qualified VoIP support technician in, India, for instance, might know the technical aspects of the
system and be a world-class troubleshooter but if they are unable to clearly communicate
verbally with the customer and determine the nature of the problem, service suffers.
Service and support costs must be carefully weighed against the actual cost of an outage or poor
support. This is one of the areas in which all costs need to be factored in: What are the
acquisition costs for the service or support resource? Will it be outsourced or performed in
house? Certain aspects of service cannot, for instance, be outsourced to India. Installation, for
example, may require a costly truck roll and in-person visit by a qualified, and expensive,
technician, but carefully designed self-installation tools might lower the instance of truck rolls,
lower costs, use the qualified technicians more wisely, and get services up and running faster.
Assuming, of course, that this process does not frustrate the end user and make their job
substantially more difficult.




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In-Sourcing vs. Outsourcing
One last consideration that should always be made, which is a hybrid of hardware and software
optimization, is consideration of in-sourcing versus outsourcing. If you are presently outsourcing
some or all your IPT infrastructure and/or management, it is possible that the assumptions upon
which that decision was made have changed and it is prudent to review the outsourcing decision.
If, however, you have chosen in-sourcing, or providing some or all of your own components and
support in-house, it is also prudent to review that decision. In many cases, there is a compelling
business case for outsourcing that is counter-balanced with a desire to more tightly control the
mission-critical IPT services, and tighter control often wins because it is driven by fear of the
unknown. After the organization is more familiar with the operation of the IPT services, and the
fear of the unknown has become familiarity with the known, the organization is better able, and
more comfortable, managing those services through an outside organization. At this point, it may
be time to take the path of the more compelling business case and the organization has developed
the operational maturity to do so.
There is a variety of aspects of the ongoing operation, and in fact the optimization of, an IPT
system that could be considered for outsourcing. This is an area in which some creativity and
new approaches to running the IPT aspect of the business could pay big dividends.


IP Contact Center Optimization
Although all the foregoing optimization techniques and approaches apply to IP Contact Centers
(IPCCs), the IPCC is also very different and has its own special needs and requirements.
Optimization of the IPCC requires a combination of technical optimization, as discussed in depth
earlier in the chapter, and operator and supervisor training to optimize the key metrics upon
which the IPCC is measured.
For example, the maximum number of calls in the queue, average and maximum call waiting
times, and call lengths can all be optimized by properly incenting agents and providing improved
agent training and more agents, but there are important dependencies to consider. For instance,
providing bonuses for shortest call times or most calls handled will likely provide an incentive
for quickly closing calls regardless of whether the problem is solved. To ensure that the callers’
problems are being addressed satisfactorily, a follow-up method is required. A combination of
scheduling optimization to ensure that the proper skills are available when they are most needed
will be required.
Agent performance can be reported on a variety of metrics that vary by call center and
applications, but typically involve sorting out and reporting an agent or group of agent’s
performance statistics by skill group, geographic coverage, closed sales, caller satisfaction, or
any other performance-based characteristic. Statistics that can be optimized include number of
calls handled, average time per call, number of aborted calls, agent downtime, revenue per agent,
and even Interactive Voice Response (IVR) statistics including the number of calls handled by
the IVR that did not require a human agent, number of IVR calls abandoned, number of IVR
calls handed off to a human operator, and similar statistics.




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Summary
This chapter has taken a broad brush at many of the most important areas to consider in
optimizing the operations and costs of your IPT services. It has gone deeper into topics that are
less implementation-specific in nature, such as hardware and service and support costs. The
chapter emphasized that optimization is a cycle: optimize your current network, set new
operational baselines, operate your “new and improved” network for a period of time, collect
data, and begin the optimization cycle all over again. The next, and final, chapter will look at all
the details and loose ends that did not fit comfortably into other chapters but that are nonetheless
important considerations in order to have a truly successful implementation of VoIP and IPT in
your environment.




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Chapter 8: Sweating the Details and Assuring Success
In 1926 self-help author Robert Collier wrote “The great successful people of the world have
used their imaginations, they think ahead and create their mental picture, and then go to work
materializing that picture in all its details, filling in here, adding a little there, altering this a bit
and that bit, but steadily building, steadily building.” We will likely find no better theme as we
enter the final chapter of this guide. We started by imagining what could be:
    •   A new world of telephony where dial tone was only the beginning
    •   A new world of telephony that provides a cornerstone for the carrier triple play, Unified
        Communications, and Multimedia, Internet, Communications, and Entertainment (MICE)
        systems.
    •   A new world of productivity and communications enabled by the Internet Protocol, SIP,
        and related protocols and technologies.
We then filled in the details, planned our approach, vetted our designs, and developed a strategy
for continually refreshing our system and improving, enhancing, and optimizing it throughout its,
hopefully long, life cycle. All along the way we strove to keep the best parts of the system we are
replacing and to add new capabilities as yet undreamed of when that early system was
architected and built.
In this chapter, we will discuss many of the details required to ensure success in this new system.
Some of the points have been covered elsewhere; others have not. We will also provide a
comprehensive checklist for the implementation of a modern, packet-based telephony system
that can be used to assure that all the critical details, large and small, are covered.


Legal and Regulatory Issues
Technology has always been ahead of the law. It is the nature of technology, and of the law. The
gun was invented before there were gun laws. The car was invented before there were
automobile regulations. Nuclear power plants existed before there were regulatory controls. In
each of these cases, the technology existed before the needed legal and regulatory framework
was in place. In each case, however, there was also a body of law that could readily be applied to
the new situation. Though not an ideal fit and not completely regulating all aspects of the new
technology, it existed, nonetheless. For instance, gun laws contain details regulating specific
aspects of guns but prior to law regarding murder, self defense and other areas of gun-related
behavior had been around for millennia. The same is true of VoIP and IPT.
There is a body of telecom regulation still on the books from earlier times, much of which is
being used as a stop-gap measure until more specific regulation of packet-based, as opposed to
circuit-based, services comes along. Some of the earlier regulation actually appears as though it
may be applied to packet-based systems on a more permanent basis. These issues and their
impact on the VoIP and IPT initiatives of all enterprises and organizations warrant the greatest
consideration to assure the overall success of any packet voice initiative. Or stated in a more
negative way, if these aspects of implementation are not considered, they could cause
implementation of the new system to be stopped dead in its tracks.




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General Considerations
The first question to ask might be “Do specific regulations apply to my organization?” A better
question, however, might be “Does specific regulation apply to my organization directly or
might it in some way impact me indirectly?” The answers to both questions will vary depending
upon the role of your organization. If your organization is the end-user organization
implementing VoIP or IPT, then one set of regulations may apply to you. If your organization is
a service provider offering packet telephony services commercially in a country, to businesses
and/or consumers, then your organization may be subject to a different set of regulations. There
is also a third situation that may or may not be worthy of consideration: that is, if a person is an
individual user of a consumer, or residential VoIP service that is used for business and
reimbursed by the company.
In most cases, if you are creating and sending your own packets, there is no specific regulation
on your activities, though there may be an indirect impact on you if you are using the services of
a carrier or service provider and they run afoul of the law and their service is suspended or shut-
down, thereby impacting the voice traffic they are carrying for you.
The major categories of issues that must be considered are:
   •   Countrywide bans of VoIP services or service providers
   •   Penalties for non-compliance
   •   Emergency services
   •   Security and wiretapping/call recording
   •   Records retention
   •   Taxes and fees
Each will be considered in broad terms here. It is your responsibility to research the specific
countries in which you will be operating to determine the present state of regulation in that
country and to plan accordingly, including continuing to use traditional circuit telephony until
regulations change or become clear.




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Countrywide Bans of VoIP Services or Service Providers
Numerous countries in the Middle East and some in Asia and South America have either fully or
partially banned VoIP services. Often they have banned VoIP services from any provider except
the registered telephone companies, many of which are fully or partially state-owned Postal,
Telephone & Telegraph (PTT) organizations. In some cases, a limited set of outside providers is
allowed to operate. The most commonly blocked VoIP service seems to be the one whose
technology is most adept at evading being blocked: Skype.
The impact on a multi-country enterprise is that in order to provide packet-based services across
the entire user base, possibly incorporating suppliers and customers, a patchwork quilt of
providers and services may need to be stitched together. Alternatively, a service-neutral Virtual
Private Network (VPN) service will need to be employed that operates in all countries that
require it and that will transparently and securely move IP-based voice packets. In many cases,
this approach will introduce a level of complexity and cost that many organizations are trying to
avoid through the adoption of VoIP. And, in some cases, if the VoIP traffic is not completely
new traffic but rather traffic that previously moved across traditional telephone lines, its absence
will be missed and it might raise a red flag for authorities to determine where their previous
telephony network revenue is going. It is worth noting that many governments around the world
view the settlements on their traditional telephony services as an important source of hard
currency.

Penalties for Non-Compliance
There are instances where penalties ranging from fines to incarceration have been levied for both
private use of VoIP and the offering of VoIP as a service to the public. It is very important to
understand the penalties that are possible and to err on the side of compliance. One does not need
to go as far as Namibia or Vietnam to find examples of penalties: there are plenty in the U.S. that
have had an indirect impact on companies using the services of certain providers, including
services intended as residential services that are used for small office/home office connectivity
and have caused service interruptions, delays in bringing services to new areas, and increases in
the cost, and therefore price, for the service.




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Emergency Services
Most industrialized nations have a system for summoning police, fire, and emergency medical
services with a single nationwide number, such as 911 in the U.S. and Canada, and 999 in the
United Kingdom. On the private enterprise side, it is important to assure that key information
such as the location of the party calling for help—for instance, their office building, floor and
area on the floor—be transmitted to emergency call centers as opposed to the billing location or
the location where the voice switch is located. It is also important for all employees and any
other persons in a facility to know exactly how to dial to emergency services. Should they dial 9-
1-1? Or 9 for an outside line and then 9-1-1? The legal liabilities are beyond the scope of this
chapter, but this is a very serious area in its own right that deserves full consideration,
implementation, and testing prior to actual release of VoIP or IPT systems for general use. The
law has generally found companies liable for non-working or inaccurate emergency calling
systems. It is also worth noting that most companies’ present systems also are not set up
properly. Companies often do not think about this until the system is pressed into service and it
does not work. Many companies also rely on cell phones for such services, which have their own
set of problems and issues, especially inside buildings.
From the standpoint of service providers, there are country-by-country regulations on their
obligations to provide emergency services. Companies should ensure that any service providers
used fully comply with all regulations and the systems should be tested.

Security and Wiretapping/Call Recording
Security and wiretapping/call recording are issues that have different dimensions if they are
considered in an enterprise or service provider environment. In the enterprise environment, the
enterprise has a need to monitor and assess what their employees and/or contractors are doing,
the nature of their communications and the applicability of organizational policy and applicable
law to those activities. Certain industries, such as the financial industry, also have particular
requirements to record calls to verify certain transactions and not just “for training purposes.”
For the service provider, call recording reaches into the realm of “legal intercept” referred to
commonly as wiretapping. Different countries have different regulations but generally there is a
trend to apply traditional wiretapping laws to packet-based services. In general, service providers
must comply and may not inform the subscribers of the ongoing investigation. This is especially
true in organized crime and terrorism cases. In the United States, the appropriate law is the
Communications Assistance to Law Enforcement Act (CALEA), which has recently been
expanded to include packet-based services.




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Records Retention
As with Instant Messages and emails there is an increasingly strong preference both for and
against retention of voice packet traffic. On the “for” side are certain regulations, such as certain
provisions of the Sarbanes-Oxley law and SEC regulations in the US, that currently require
retention of IM and email and may be broadened to include voice packets. On the “against” side
is an increasing number of cases in which corporate or government wrong-doing has been able to
be proven by the use of records which, if they did not exist, could not have been used for this
purpose. Retention of voice packets is an area of current legal interest and one that will continue
to be refined in the future. It is also an area that must be considered in any corporate policy.

Taxes and Fees
One of the biggest benefits of a move from traditional telephony to VoIP was that VoIP was free
of much of the regulation and nearly all of the fees traditionally levied against phone companies
and the services they provide. This was translated into lower prices and was responsible for
much of the initial interest in VoIP. As governments around the world become aware of VoIP as
a source of revenue, from the local government to the national level, taxes and fees are being
dreamed up for VoIP. Absent strong precedent many taxing authorities may, at least for some
initial period of time, be able to levy unbelievably high fees, such as the city of Baltimore has
done, with a $3.50 per number tax on VoIP subscribers. This will impact service providers
directly and enterprises indirectly due to the service providers passing on the costs of fees and
compliance.

Country-Specific Regulatory Issues
It is imperative that you check current regulations in every country in which you operate to
assure that you are meeting all legal and regulatory requirements. In addition to national
requirements there may be state or provincial requirements and even requirements imposed by
local governments that can harm your packet-based voice initiative.

Contractual Issues
Many organizations, especially large ones, although the same may be true for small to medium
ones with especially high call volumes to specific countries, have specific contracts, often called
tariffs with the previously popular “Tariff 12” being a good example - in place with traditional
telephony carriers. The tariffs are filed with the FCC or other applicable authority in other
countries than the US and spell out the pricing and contract conditions, and often have minimum
numbers of minutes that must be paid for whether delivered by the carrier or not. Before
calculating the “savings” which can be realized by a move to non-traditional telephony
organizations should carefully review and understand the conditions of the current contracts. In
many cases there are guaranteed minimum numbers of minutes, or payments, or both. Moving
minutes to a VPN or another telephony service, even of the same carrier or service provider, may
not satisfy the minimum service requirements and payment must be made for the minimum
traffic levels whether the traffic is transported by the carrier or not. Many organizations have
been on the path to implementing voice packet services and have found themselves in the
position of having to delay implementation until the end of a multiyear contract for traditional
telephony services.


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Patent Issues
It is said that history repeats itself and so it does more often than we might realize. When
statements are made that VoIP and IPT technologies are “mature” and “ready for prime time”
one must look no further than the early days of traditional telephony to see evidence that, in fact,
VoIP and IPT are in their earliest days. If one were to read the newspapers from the ‘70s, that is
to say the 1870s, not the 1970s, one would find that patent disputes are common in the early days
of the commercialization of any new technology and packet telephony is no different. The 1870s
saw bitter legal battles between competitors for the claims to telephony patents. There are many
examples just in the lawsuits against Alexander Graham Bell from a variety of individuals
including Antonio Meucci and Elisha Gray and the problems Bell had due to his patents being
filed in America before the were filed in the UK. History is repeating itself and this is why the
final topic in this section is the business impact of the increasing reliance on patent filings and
the enforcement of patents as a business strategy on VoIP, IPT and related Unified
Communications projects.
Once the dust settled in the telephone industry, solid patents and standards became the bedrock
of reliable, interoperable global phone systems. All things considered there were really very few
variations in phone systems globally, compared with their data counterparts and almost none
whatsoever considering the number of variations that could exist. That having been said, there
are still hundreds of important variations from country to country and beyond the national
variations there are even manufacturer-specific variations. But, even with all of the variations on
the themes underlying issues of patent protection, once settled, became a non-issue throughout
the remainder of the life-cycle of the traditional telephony system.
Because VoIP, IPT, Unified Communications, Internet Multimedia Subsystem (IMS) and related
systems are in their commercial infancy, patent issues remain an important consideration for any
organization implementing VoIP and IPT systems and/or services. In fact, these are the two
categories that must be considered: systems and services. If a company is purchasing systems
then they must be cognizant of the fact that the systems they are purchasing, and the underlying
hardware and/or software, even if purchased from a historically reliable manufacturer may
contain technologies that might later be proven to be infringing on patents that the manufacturer
does not own or have license to use. Likewise, when purchasing services the purchaser must be
aware that the services may be based upon patents that the service provider does not own or have
license to use. What is the “bottom line”? The bottom line is that while most legal outcomes do
not require that technology purchased be given back they do make acquisition of subsequent
products more expensive or impossible and may cause the suspension of services.
While there is really no analysis or report on what the risks are this is still an area that must be
considered and, more than anything else, will impact the decision of companies to adopt VoIP
and IPT until many of the early patent issues have been decided. The alternative? Rely on
extending current technologies and only using VoIP and IPT technologies where they are
strategically vital in the mean time.




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Technical Issues
Even though many technical issues have been addressed in other areas of this eBook they are
being re-emphasized here because they often get lost in the background noise but have
confounded a proper implementation of VoIP and IPT in many organizations. These topics are
presented in no particular order of importance and if properly addressed during the
implementation phase none will be particularly thorny but if left unaddressed can become very
serious impediments to productivity.

Voice Band Data
There is a general expectation that telephony systems can handle a variety of Voice Band Data
(VBD) options such as:
   •   FAX
   •   Modems
   •   Dual Tone Multi-Frequency (DTMF, also known as touch tone and used to access
       voicemail, bank balances, flight information and other services)
   •   TTY/TDD devices for the deaf
   •   Alarm systems
   •   Other telemetry systems
There is, however, no set of standards for supporting these capabilities within VoIP and IPT
systems and each manufacturer and service provider will offer their own set of options.
If an organization implements VoIP using Ethernet-based devices, and a standard Ethernet
physical interface such as the ubiquitous eight wire RJ45 modular connector then the expectation
of VBD support usually goes away right along with the traditional phone jack. This is because
the Ethernet physical connector is different from the four or two wire phone jack and there is no
place to plug in the fax, modem, TTY terminal or alarm system. If, on the other hand, Internet
Protocol Telephony is implemented using some sort of analog terminal adapter or gateway that
connects via a standard phone plug then the expectation that the system can handle VBD usually
still exists.
If VBD is to be handled properly in the new system then proper arrangements need to be made.
If the new system is not intended to provide support or VBD then proper training and support
needs to be provided to deal with the lack of the aforementioned capabilities or to provide
workarounds. If, for instance, faxes are to be supported then the new system must either use
64Kbps Pulse Code Modulation (PCM) or, possibly, 32Kbps ADPCM voice coding which
requires additional bandwidth for all calls. If a lower bit rate codec is to be used for normal voice
calls then the system must be capable of detecting the presence of the fax signal and switching
over to PCM or ADPCM or employing FAX T.38 forwarding, fax servers or similar capabilities.




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Numbering Plans
VoIP and IPT do not use numbering plans or phone numbers, per se, but they can, and do, have
the capability of mimicking phone numbers, at least for an interim period of time. Many in the IP
standards bodies have been chiding the current purveyors of VoIP services claiming that they
don’t provider “real VoIP” or “real SIP implementation” because they still use phone numbers.
The use of phone numbers, at least in an alphanumeric form as part of a SIP User Resource
Identifier (URI), will be critical until the migration is made from the older telephony networks
and, perhaps, that older telephony network may still be needed for some time to come for its
ability to locate subscribers and route calls to them as this is a scalability and interoperability
capability as yet to be proven by SIP networks.

Voice VPNs
Many organizations, from multi-branch single-city banks to multinational manufacturers and
consulting firms have long ago established voice Virtual Private Networks (VPNs) with their
own specialized dialing plans and associated volume pricing agreements. The particulars of the
use of the specialized dialing plans has become so engrained in these organizations that any
productivity gains associated with a shift to packet telephony would be more than offset for
many years to come by making changes to the dialing plans. Employees know each other’s
extension numbers: clients and suppliers know Direct Inward Dialing (DID) numbers based upon
those extensions, employees know specific internal numbers such as Human Resources and
Security, directories are published, phone numbers are part of documentation and often posted on
telephones, bulletin boards and web sites. In short, the entire ability of a company to
communicate via voice, both internally and with the outside world, is based upon previously
established phone numbers and associated dialing plans. For these reasons it is imperative that
the prior dialing plan and associated phone numbers are maintained. It is also imperative, to the
extent possible, that the special characteristics associated with the old system, such as the exact
sound of inside and outside dial tone, the delay that occurs after dialing the digit (0? 8? 9? 7?)
used to get an outside line and all other characteristics be kept unchanged.

E.164 Compliance
E.164 is the global ITU (http://www.itu.org) standard for telephone numbers. E.164 provides an
international numbering plan for public telephone systems in which each assigned number
contains a country code (CC), a national destination code (NDC), and a subscriber number (SN).
There can be up to 15 digits in an E.164 number and each E.164 number is unique worldwide.
With up to 15 digits possible in a number, there are 100 trillion possible E.164 phone numbers,
more than 10,000 for every human being on earth. This makes it possible, in theory, to direct-dial
from any conventional phone to any other conventional phone in the world by inputting no more
than 15 single digits. The hierarchical nature of E.164 as well as the number and use of digits is
based largely on work done in the 1880s and 1890s by Alexander Graham Bell and outlined in
his famous letter to the board of directors of the Bell Telephone Company in 1894. This same
letter described the use of telephone exchanges and foretold of automated switching, something
for which Dr. Bell could see the need but could neither describe nor produce.




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Maintaining strict compliance with E.164 standards globally at every gateway between the
packet world and the circuit world is critical in maintaining the bridge between the two, and
assuring interoperability as long as the bridge between the two worlds of telephony is still
needed, which is to say, well into the foreseeable future.

URIs/URLs
The Internet uses Uniform Resource Locators (URLs) such as www.prognosis.com to locate
hosts that have some sort of geographic alignment. SIP uses a very similar addressing construct,
the Uniform Resource Identifier (URI). A SIP URI identifies a communications resource. Like
all URIs, SIP URIs may be placed in web pages, email messages, or printed literature. They
contain sufficient information to initiate and maintain a communication session with the
resource.
                                    SIP Uniform Resource Identifiers
The "sip:" scheme follows the guidelines in Internet Engineering Task Force (IETC) Request for Comment
(RFC) 2396 - Uniform Resource Identifiers (URI): Generic Syntax. They use a form similar to the mailto
URL, allowing the specification of SIP request-header fields and the SIP message-body. This makes it
possible to specify the subject, media type, or urgency of sessions initiated by using a URI on a Web
page or in an email message. In general, the form of a SIP URI, is
                        sip:user:password@host:port;uri-parameters?headers
    ●   In a SIP URI user is the identifier of a particular resource at the host being addressed. The term
        "host" in this context frequently refers to a domain.
    ●   The "userinfo" of a URI consists of this user field, the password field
    ●   The @ sign following the userinfo fields. The userinfo part of a URI is optional and may be absent
        when the destination host does not have a notion of users or when the host itself is the resource
        being identified.
    ●   If the @ sign is present in a SIP URI, the user field must not be empty.
    ●   If the host being addressed can process telephone numbers, for instance it is an Internet
        telephony gateway, a telephone subscriber field may be used to populate the user field.”
    ●   Password is optional and is a password associated with the user. While the SIP URI syntax
        allows this field to be present, its use is optional and not recommended, because the passing of
        authentication information in clear text (such as URIs) has proven to be a security risk in almost
        every case where it has been used.
    ●   The host field provides the identifier for the SIP resource. The host part contains either a fully-
        qualified domain name or numeric IPv4 or IPv6 address. Using the fully-qualified domain name
        form, such as phn.company.com, is recommended whenever possible.
    ●   In addition to the host name a port field may be provided and, if provided, designates the port
        number where the request is to be sent.
If a telephone number is to be transmitted it is converted into a SIP URI of the form:
               sip:nnnnn@domain;user=phone or sip:nnnnn@host:5060;user=phone
    ●   The "user=phone" parameter is a hint that the part to the left of the '@' sign is actually a phone
        number, in case there are SIP users whose names happen to consist of all digits. Typically a
        proxy server for the domain will use a dial plan to resolve this into a real destination.




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ENUM and Other Numbering Issues
The ITU and the Internet Engineering Task Force (IETF) are currently working on a plan called
ENUM that will expand E.164 to encompass both traditional analog phones and digital devices,
including computers and other devices on the Internet. All types of communications devices—
including analog phones and fax machines, digital phones and fax machines, wireless (cellular)
phones, pagers, digital modems, digital video terminals, and VoIP devices -- will have unique
E.164 addresses with direct dialing possible from any device to any other. Considerations for
ENUM and extensions to corporate directories using Light weight Directory Access Protocol
(LDAP) and Domain Name Service (DNS) systems must be considered as a part of any
comprehensive implementation.

Interworking Issues with IPT Service Providers
Invariably a packet telephony project of any size will need to connect to IP Telephony service
providers for either connectivity between geographically distant internal locations, often
including small office/home office sites, or customer or supplier sites but usually all of the
above. There are several options and issues that must be considered, all of which have analogs in
the earlier phone system.

The New Demark
The issue of where the responsibility of the “phone company” ends and the responsibility of the
customer begins has been the subject of very specific regulation over the years and the issue is
very clear. The point at which the “hand off” occurs and responsibility shifts is called the point
of demarcation, or “demark”. With individual voice circuits the point of demarcation is marked
at a wire connecting a punch-down block owned by the phone company (normally attached to a
piece of plywood on the user premise) to another punch-down block (on the same or a different
piece of plywood on the user premise). In the case of T1 systems the demark is a phone company
or customer owned T1 DSU/CSU device. There are other examples, but the point is that the stage
(or place) where the hand-off occurs is very specific in traditional systems.
In emerging VoIP and IPT systems the point of demarcation is usually as specific because it is
the same carrier, or type of carrier, providing the wires but often a different service provider
using those wires to provide the service. This is why a fresh interest must be taken in clearly
defining customer responsibilities and service provider responsibilities.




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End-to-End Service Management
An increasingly important issue that impacts all aspects of multimedia, whether the service is
provided by an internal or external organization, is “end-to-end” service management. In the past
“end-to-end” has been defined more as provider edge to provider edge, not including the local
access links on the end, unless equipment was co-located in a service provider Point-of-Presence
(POP) or carrier Central Office (CO). That definition was expanded by managed service
providers to include the local access links and the managed customer premise equipment, such as
the router, but did not include the actual service that used the underlying connections nor did it
include things such as servers and client end-systems.
VoIP and IPT services should be managed - and key metrics such as Quality of Service (QoS)
and Quality of [user] Experience (QoE) measured—from the user phone device on one end to the
user phone device on the other end. Monitoring, measurement and management of all
intermediate pieces from which the end-to-end call is created remains very important but it is the
actual service itself that should be measured and managed. It is also critical to view VoIP and
IPT traffic in its context as part of the bigger picture of multimedia traffic.

Tasks

Even back when IP networks were “data only” there were many “flavors” of data - browser
traffic, file transfer protocol, terminal emulation, email—all of which have their own
performance requirements Assessing IP performance has become even more complex with the
addition of two more distinct types of traffic, voice and video, to the multimedia mix. All of the
tasks required of today’s IP multimedia networks—functions referred to as applications in an
environment often called the single network voice, data, video “triple play”—are easier to deal
with if we place them into one of three “loss” categories and cross reference those with two
delay-related categories, understanding that these categories are relative to each other and not to
some standard baseline. It is in the dynamic prioritization of each type of traffic in the network
where we achieve true quality of service that matches, or doesn’t match, the needs of each type
of traffic.
As shown in Table 8.1 applications on a multimedia network can be broadly divided into those
that are less sensitive to delay and delay variation and those that are more sensitive. We can
further divide those two categories into low, medium and high in terms of the loss that is
relatively acceptable for each application. If we were considering the underlying protocols we
would actually place the applications into only two categories, those that use the reliable, error-
checking, retransmit-on-loss connection-oriented TCP protocol or the connection-less User
Datagram Protocol (UDP), often referred to by this author as the “Unreliable Datagram
Protocol”. In this case there would be no loss in the TCP examples, but possible loss in the UDP
examples, but we are looking at this from the user’s perspective and will view the “Acceptable
Loss” based on its impact on the total user experience, often referred to as QoE, or Quality of
Experience.




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Table 8.1: Acceptable Multimedia Loss vs. Delay.

These relationships also imply a certain prioritization relative to Quality of Experience (QoE)
which comes into play if packets are competing for network resources. For instance, from the
table we can rightly imply that if an email packet were competing with a real-time voice (VoIP)
packet for network resources that the VoIP packet would get the resources. Likewise if a device
such as a router had to discard a packet and a VoIP packet were competing with a browser packet
that the browser packet would win because the discarding of a browser packet would cause the
retransmission of the packet, placing a further demand on the network to do the retransmission.
Quite the opposite is true of voice. Voice will sustain a 3-5% loss before the quality drop is
noticeable and can randomly lose up to around 10% of voice samples before intelligibility begins
to decline substantially. Considering these dynamic trade-offs, based upon the demand on the
network at a specific moment in time, is an important a part of assessing the performance of your
multimedia IP network.
If this explanation seems to run counter to traditional thinking, based upon the well-known
characteristics of IP, TCP and UDP you are quite right. The underlying protocols were designed
for a uni-media, not multi-media, world and the shift in network operations is a function of
changes in the software code in routers and switches that lie in the path between one
communicating system and another, not in a fundamental re-write of TCP or UDP.

Tools

A specific, finite list of characteristics may be ascribed to certain types of packets either as they
are formed in their respective end systems, as they enter the network, or at any other point where
they are handled by systems capable of such discrimination and traffic marking. Each of the
combination of characteristics can be matched to a specific designation, or class of service, with
the intention that traffic so-marked will achieve the corresponding promised quality of service.
Any given network may or may not implement all of the classes of service or may have some
special hybrid.
Constant Bit Rate (CBR) promises service most like a dedicated circuit. It promises a constant
bit rate and near zero loss. Real-Time (RT) and near-Real-Time (nRT) are two flavors of
Variable Bit Rate (VBR) classes of service that promise performance below that of a guaranteed
circuit but better than the best effort objective of historical IP networks. Available Bit Rate
(ABR), which promises a modest “best effort” service with a minimal assured throughput, even
in times of congestion, and Unspecified Bit Rate (UBR), which makes no guarantees whatsoever,
in most cases, round out the classes of service.
By aligning the needs of each of the applications to specific classes of service it is possible to
provide the full range of tools needed to provide the mixed quality-of-service needed for
multimedia traffic and to request the network to provide for the specific needs of each.


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Differentiation and Outcomes

The key to success is to properly differentiate and then to aim for the best but plan for the worst.
In today’s multi-terabit networks it is easy to picture an optical highway system so empty that
every packet can receive first class treatment. While this may be an accurate high level view it
neglects two important facts: the first is that one must actually be able to get on the highway and
the second is that regardless of the low and declining cost per bit per second, businesses are
always driven to save money and often sacrifice performance to do so.
In this frame of reference it is easy to see that trade-offs are made in how the carrier or service
provider’s bandwidth is carved up and how it is paid for: regardless of low price there is still a
cost. The key is to properly differentiate traffic and not to over-provision or under-provision the
system relative to the need.
Table 8.2 shows which class of service is ordinarily applied to each sample application. In a
perfect world—and it does happen in the networking world, especially at off-peak times—all
traffic is marked according to what class of service it is promised, but all packets, regardless of
marking, traverse the network in near-wire-speed times and none of them are delayed or
discarded. In that case the differentiation and class-of-service marking of packets is
inconsequential and it could even be argued adds to delay and overhead. So, why do it? Because
the network is not a perfect place, especially at peak times, and the marking tells the network
how to resolve conflicts for resources when they ultimately arise in the shared environment of
the IP network.




Table 8.2: Class of Service (CoS) Examples.




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Monitoring and Testing
Each of the classes of service will have some associated metrics that can be measured and
interpreted relative to its impact on the application. The metrics that are normally gathered and
used are packet loss, delay, delay variation (jitter) and service availability. Constant Bit Rate, for
instance, has a packet loss near zero, delay near wire speed (meaning that distance is a more
important factor than the performance of intermediate pieces of equipment), delay variation near
zero and a 99.99+% availability. On the other end of the spectrum is Unspecified Bit Rate, with
no guarantees at all, in most cases. To judge both is easy, and fair. A packet loss of a whopping
2.6% for CBR is considered to be abysmal performance and most likely subject to a financial
penalty against the carrier. A loss of only 2.6%, especially during peak traffic times, is probably
pretty good for Unspecified Bit Rate. The key is to have the target metrics, and penalties, spelled
out in advance and to monitor performance relative to the agreed upon targets. This is best done
in the Service Level Agreement and monitored via a combined program of intrusive testing and
passive monitoring.

SIP-T
Traditional telephone networks, as shown in Figure 1, employ two distinct parallel networks to
create, facilitate and delete end-to-end telephone connections. The network interconnecting the
ISDN phone on the left via discrete circuits running through the grey cloud in the middle to the
analog phone on the right is only the network that carries the bits between the two phones.




Figure 8.1: Traditional telephony architecture.

The blue cloud in the back, in this case marked “SS7 Network” because it facilitates the
signaling system used in North America, is a parallel control network comprised of highly
reliable dual-home packet switches that does the set-up and tear-down of the circuits. Without
the secondary signaling network there would be no calls. This is quite different in IP-based
networks. IP-based networks are characterized by the fact that the same network, the IP-based
Internet, intranets or VPNs, carries both the content and the signaling traffic. One aspect of IP
networks that has concerned telephony engineers from the beginning, is that while their SS7
networks provide highly reliable out-of-band signaling to insure the integrity of the network and
minimize risks to network reliability and, therefore revenue, IP networks started out as “best
effort” with control and user traffic competing for shared resources on an even footing. This,
fortunately, has changed for a variety of reasons and control traffic can receive priority
treatment.




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It is important to consider the signaling needs of the network during design and implementation.
The Session Initiation Protocol (SIP) is the most likely choice for voice sessions in the network
and, therefore, SIP-T, Session Initiation Protocol for Telephones, will be the most likely choice
for signaling, call set-up and tear-down. SIP-T provides a framework for the integration of
legacy telephony signaling into SIP messages. SIP-T accomplishes this using two techniques
known as ‘encapsulation' and 'translation'. At a SIP gateway, SS7 messages are encapsulated
within SIP in order that information necessary for services is not discarded in the SIP request.
However, intermediaries like proxy servers that make routing decisions for SIP requests cannot
be expected to understand SS7’s particular signaling protocol called the ISDN-User Part (ISUP),
so simultaneously, some critical information is translated from an ISUP message into the
corresponding SIP headers in order to determine how the SIP request will be routed.
While pure SIP has all the requisite instruments for the establishment and termination of calls, it
does not have any baseline mechanism to carry any mid-call information (such as the ISUP
INFormation/Information Requst (INF/INR) query) along the SIP signaling path during the
session. This mid-call information does not result in any change in the state of SIP calls or the
parameters of the sessions that SIP initiates. A provision to transmit such optional application-
layer information is also needed.

Electrical Power
AC wall power or Power over Ethernet (PoE) must be provided to power most IP phones. In
many cases consideration of power is not a part of network design and results in substantial
additional costs and implementation delays.


Business Issues
Are packet-based telephony systems less expensive than their circuit-based counterparts? Is it
possible to tell? Are the systems so different that it becomes an apples vs. watermelon
comparison? It is most likely to be true that IPT systems, which use a new engine to deliver
traditional voice services will be less expensive, possibly 30 to 60% less expensive, than their
traditional telephony counterparts. Most organizations, however, eschew a strict replacement
approach for one of replacing dial tone and then adding new capabilities, placing them squarely
in the VoIP realm, regardless of their choice of underlying technology. With VoIP the
organization adds additional capability but also brings additional training requirements and
budgetary and staffing needs.




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Budgets
Packet-based telephony systems are different from their circuit counter-parts in many important
ways that have an impact on budgets. For instance, hardware becomes obsolete much more
quickly and refresh or replacement must be budgeted. In the old telephony networks it was
possible to have a standard touch-tone phone in service for a decade or two. Not so with today’s
IP phones, palm devices and wireless phones: the technologies and capabilities are changing too
rapidly. A reasonable refresh cycle is twelve to eighteen months depending on your industry.
Software and services are also areas that are quite different than in the old telephony days. IP
phones often have software licenses or costs for specific features that were either non-existent or
associated with the phone switch or PBX in traditional systems. These items must be managed
and budgeted.

Staff
One of the biggest misconceptions in the early days of IP telephony was that voice was “just
another IP application” and that it could be managed successfully by the existing staff with
existing tools. That is a misconception that is now understood to be wildly wrong. What many
organizations are now coming to grips with is that they must keep at least some of the former
telephony staff members. Ideally from this process a “multi-tech” with the requisite skills to
handle voice, data and video issues will emerge but wide-spread availability of such super-techs
is still some way off. Organizations should guard against skills attrition in their staffing efforts
and assure that they are providing both training of new staff and cross-training of existing staff in
all aspects of voice, data and video.

Repositioning/Redeployment of Old Equipment
While some parts of the organization may be moving rapidly to deploy packet technology it may
be difficult or impossible for other areas to do so and it might make very good sense to reposition
or redeploy equipment that is being removed from one area of the organization to another area.
What if one division has chosen not to make a move to packet telephony at this time? They may
still need to upgrade their old key system to handle more phone lines. Or, what if one location is
in an area where regulations forbid VoIP or there is a minimum number of minutes required on a
tariff that can’t be met if traffic is shifted to VoIP before the end of the contract period? These
are great places to reposition older equipment.

Secondary Market for Old Equipment
For a variety of reasons, the most important of which is manufacturer discontinuation of specific
lines of traditional telephony equipment there is a thriving secondary market for older telephony
equipment. The secondary market should always be considered vs discarding older equipment.




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Donation of Old Equipment
In the early days of packet telephony, when the direction was less certain, donation of old
equipment to educational or non-profit organizations for their own internal use or training
purposes made sense and the donations were welcome. The farther we go toward world packet
domination the less true this is and the more that donations of this type are seen as liabilities. It is
still possible to donate old equipment but the context needs to change: donations to
manufacturers, carriers or other museums are becoming more important, especially for pieces of
equipment that are particularly unique or have an interesting role they played in a historical
event. Organizations should not underestimate the value of their old, outdated equipment and
should always look at the value of donation of that equipment as opposed to discarding the
equipment.

Revisiting TCO and ROI Models for your organization
In Chapter 1 we described several different approaches to determining Total Cost of Ownership
(TCO) and Return on Investment (RoI) models. Your approach might exactly parallel one of the
suggested methodologies, might closely resemble one or two or might be completely different. In
any case once the new system has been installed and stabilized, and once a sufficiently large
number of users have been using the system to get some meaningful numbers it is time to revisit
your TCO and RoI models and see how you’ve done.
Is the Total Cost of Ownership lower or higher than originally estimated? Was the investment
returned in the time frame estimated or does it appear as though it will be? Will it take longer?
Less time? Why or why not? Many organizations never go back and recheck the initial
assumptions but it is important to do so. The first reason might be for “bragging rights” to a
properly performed prognostication or, in any case, maybe some original assumptions were
inaccurate or never made it into the final design. Regardless, it is a valuable and important
exercise to revisit the original planning exercises and clear up any discrepancies.




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Security, Privacy, and Safety
Generally speaking because VoIP and IPT operate over IP networks they are subject to the same
security vulnerabilities and issues as any IP service. However, to the extent that the calls may
traverse traditional phone systems and networks they are subject to the security vulnerabilities
and issues of traditional telephone networks as well. For this reason a plan must be put into place
to assure the integrity, security and privacy of all phone calls regardless of the system over which
they are connected. This would also be a good place to discuss wireless vs. wireline issues as
well. A complete and comprehensive security analysis at each design step followed by security
assessments of the actual system once it is installed and periodically thereafter are critical to the
operation of the system.
The four areas of security that will need to be addressed are:
   •   Privacy
   •   Operational continuity
   •   System integrity
   •   Theft of service
Privacy assures that information being transmitted remains between the communicating parties
and cannot be “overheard” by any third party. Privacy can be accomplished through a variety of
means, the most effective of which is end-to-end encryption. End-to-end encryption scrambles
the signal at one end-user device and unscrambles it at the other end. End-to-end encryption
assures that the message is as secure as possible from one end to the other.
Operational continuity is the process of protecting a system from issues such as Denial of
Service (DoS) attacks and assuring that calls can be processed under any foreseeable
circumstance. Denial of Service attacks that are beginning to emerge in VoIP and IPT systems
include restarting call servers so that the traffic generated by all of the connections attempting to
re-establish themselves brings the system down.
System Integrity problems include situations where hackers can masquerade as internal callers or
spoof IDs and phone numbers as well as assuring the integrity of call logs and other tools that
can be used after-the-fact for their forensic value.
Theft of service, often called ‘toll fraud” in the traditional phone network has almost disappeared
as an issue from the lists of most managers of packet voice systems but shouldn’t. In the past the
number one impact of toll fraud was the high cost. The cost of stolen minutes had to be paid to
the telephone company and this had a direct financial impact. Packet-based systems are so
inexpensive, by comparison, and are often flat-rate so the financial impact is not so noticeable.
Theft of service, or unauthorized use by employees, steals time from the job and often has other
implications such as setting up an organization for prosecution if their systems are being used by
hackers, criminals or terrorists.




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Emergency Calls—911, 999, 112, and So On
In addition to the regulatory issues discussed earlier it is imperative that an organization fully
understand the liabilities associated with not providing proper emergency calling. Not only are
the lives of employees, contractors and visitors placed at risk with a non-functional emergency
calling system property can be placed at risk if it is not possible to summon fire fighters in case
of a fire in the building. As stated earlier it is imperative that emergency calling procedures are
well documented and tested to assure they will operate properly when needed and summon
assistance to the proper location.


Telephony Migration Checklist
The telephony migration checklist provided at the end of this chapter offers a single list that can
be used to double check all of the key aspects of your migration from traditional telephony to IP-
based communications.


Summary
The successful implementation of VoIP and IP Telephony is a topic that is broad and deep all at
once. In this eight chapter e-book we have made every attempt to touch on all of the issues that
you might encounter and to delve deeper into those that are likely to be of greatest importance.
We have provided templates and checklists, guidance and “war stories”. With all of this input
and your own experiences to guide you it is still impossible to get everything exactly right on the
first go but with careful planning and implementation it will be possible to get closer to
perfection than would ever have been possible by just “taking a stab at it”.
If we have provided some valuable insights, if we have shown some possible booby-traps and
some practical work-arounds and saved you time, aggravation and money we have accomplished
what we set out to accomplish at the beginning. Best of luck to you as you pioneer a new way of
communicating and find the best path forward for your organization and its users.




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Telephony Migration Checklist

Justification
        _ Return on Investment (RoI)
        _ Return on Effort (RoE)
        _ Total Cost of Ownership
        _ Strategic capabilities
        _ Manufacturer discontinuation of older systems
Expectation Management
      _ Users
          _ Different is not bad
          _ Provides all important functions of the current system
          _ Will provide important new capabilities.
      _ Management
          _ Total CAPEX cost may be the same or higher
          _ Total OPEX cost may be the same or higher
          _ Must assure the continuity of voice communications
          _ Acquire advanced capabilities of strategic benefit
          _ VoIP is not “free voice”
          _ VoIP is not “just another application.”
Budgeting and Planning
      _ Reuse/Redeployment of existing assets
      _ Acquisition
         _ Products
         _ Services
         _ Managed services and outsourcing
      _ Support
         _ Internal Help Desk
         _External Help Desk
Service Level Agreements (SLA)
       _ Parameters
          _ Availability
          _ QoS: Packet loss
          _ QoS: Delay
          _ QoS: Delay Variation
          _ QoE: MOS
          _ QoE: E-Model/R value
          _ QoE: ITU metrics




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      _ Classes of Service (CoS)
         _ Differentiation
         _ Prioritization
         _ Queue management
         _ Per CoS shaping and policing
      _ Penalties
SLAs, Monitoring and Management
      _ Pre-deployment simulation and modeling
      _ Manufacturer-specific monitoring and management
      _ Real-time business views
          _ Call detail records
          _ Calls in progress
      _ Delay-to-dial-tone rates
      _ Gateway channel utilization and loading
      _ Real-time call monitoring
      _ Phone and multi-media device availability and monitoring
      _ Successful vs. failed call completion rates
      _ Poorly performing components
      _ Service level breaches and SLA compliance
      _ Real-time interface to Manager of Managers (such as HP OpenView)
      _ Summary and exception reporting
      _ Utilization trends over time
      _ Managed devices by company, department, and location
      _ Asset tracking
      _ Capacity planning
          _ Incoming and outgoing calls
          _ Loading by dial plan, routing rules and gateway.
          _ Bandwidth utilization
          _ Delay and delay variation (jitter)
          _ Packet loss
          _ Route patterns, utilization, and availability
      _ Customer Network Management (CNM)
      _ MSP provided tools
      _ Enterprise Tools
          _ Manufacturer provided tools
          _ Third-party tools




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Assessing Urgency and Prioritizing Migration
       _ No flash cut
       _ Assess urgency
       _ Prioritize migration order
              •         Which office or region has outgrown, or is closest to outgrowing
                  their existing system?
              •        Which region has bandwidth or IP network infrastructure good
                  enough to carry voice, and other resources to accommodate the new
                  telephony upgrades?
              •        Which region has the most technically savvy support staff?
              •         Which region was involved with early testing and/or staging and
                  is, therefore, most familiar with the new system?
              •         Which office or region is closest, because the earliest migrations
                  are always learning experiences and should be expected to go slower
                  and be more problematic?
Migration
      _ Systems Integrator vs. Do It Yourself
      _ Managed Service vs. Do It Yourself
Planning & Assessment
      _ Inventory of existing capabilities
      _ Network infrastructure
            •        Identify circuits used for wide area networking and Internet
               connectivity
              •      Inventory the network hardware—routers, switches, cabling. Don’t
                overlook elements like router operating system version. This is a good
                time to get your routed network all on a common foundation.
       _ Current voice services and equipment
             •       Document any inbound and outbound call center environments
              •        List the details of hunt groups, call pickup groups and automatic all
                 distribution groups.
       _ Current business data applications
              •        What servers are in place today? What’s changing or evolving in
                 parallel? Identify any linkages between your existing business
                 applications and voice services. If you have a CRM system in place, this
                 piece will be absolutely crucial.
       _ Security posture
              •        Inventory your security environment early to ensure there are no
                 surprises later on.
       _ Existing voice needs—The calling patterns today
          _Call Detail Records (CDRs)


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_ Network Readiness
      •      Estimate the bandwidth needed to support a known number of
        phone lines
      •       Calculate the bandwidth required to support busy-hour call
          volumes
      •       Determine how many paths might be needed across a wide area
          network (WAN)
      •       Delay budget is important and includes all sources of fixed delay.
          Delay less than 150ms one-way is ideal; >300ms will impact voice
          QoE. Delay should be specified in the SLA.
      •        Delay variation (aka jitter) is a bigger problem than delay. Delay is
          fixed and easier for the listener to adjust to. Delay variation is difficult
          to adjust for and causes anxiety and stress on the part of listener.
      •         The human ear can adjust to packet loss up to about 10 percent.
          Packet loss >1.5 to 3 percent can have impact on voice quality. Packet
          loss impact can be reduced by shortened voice samples and fewer
          samples per RTP packet.
      •        Network availability directly impacts QoE. No dial tone is a BIG
         problem and causes loss of confidence in the system.
_ CODEC Selection
   _ Voice Coding & Quality
   _ Bandwidth Use
       _ Constant Bit Rate
       _ Variable Bit Rate
       _ Silence Suppression / Voice Activity Detection
_ Gateways & Signaling
   _ Trunking Gateways
   _ Signaling
_ Echo Cancellation
_ Assessment Tools & Modeling Process
   _ Traffic collection from existing IP network
   _ Call traffic collection from existing telephony network
   _ Import of Call Detail Records (CDRs) from telephony network/bill
   _ Off-line simulation of VoP/VoIP network impact
   _Simulated VoP/VoIP traffic insertion into live network
   _ Real-time assessment of VoP/VoIP traffic impact
   _ Real-time tuning of VoP/VoIP parameters
   _ Manufacturer-specific parameters, metrics and analysis
   _ Call Server, IP-PBX or Switch Manufacturer-provided tool
   _ Simulate complex traffic patterns
   _ Modeling/sizing of softswitches, IP PBX and IP Centrex
   _ Modeling/sizing of gateways
   _Transfer pre-deployment assessment info to NMS


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_ Model different codecs
    _ Pulse Code Modulation (G.711/ PCM)
    _ Adaptive Differential Pulse Code Modulation (G.721/G.726/G.727 ADPCM)
    _ Linear Predictive (LP) codecs: G.729, G.729a or G.723.1, G.726
    _ Other Proprietary Linear Predictive (LP) codecs
    _ Wideband codecs
_ Different voice sample sizes
_ Different packet fills
_ Voice activity detection/silence suppression
_ VoIP w/SIP
_ VoIP w/H.323
_ Voice over Frame Relay (VoFR)
_ Voice over ATM (VoATM)
_ Voice over DSL (VoDSL)
_ Modeling/sizing VoP gateway performance
_ Performance through multiple gateways
_ Delay
_ Delay Variation / Jitter
_ Service Level Agreement Classes of Service
_ Multimedia VoIP/VoP calls with video
_ Impact of VoIP/VoP on data and video performance
_ Impact of data and video performance on VoP/VoIP
_ Performance of multimedia sessions/applications
_ Simulate combined multimedia nets (i.e. different divs or depts)
_ Simulate multimedia MPLS VPN
_ Simulate multimedia Metro or Wide Area Ethernet VPN
_ Simulate multimedia VLANs
_ Forecast Mean Opinion Score (MOS) using G.107 eModel
_ Forecast Mean Opinion Score (MOS) using proprietary method
_ Forecast Delay
_ Forecast Delay Variation / Jitter
_ Forecast VoP Service availability
_ Model voice-only QoS/QoE
_ Model non-voice multimedia QoS/QoE (i.e. video)
_ Model combined multimedia QoS/QoE
_ Model packet loss and impact
_ Draft "Buy vs. Build" reports to support VoP/VoIP approach
_ Cost/price analysis for design option
_ Reports/tables/charts showing voice quality metrics vs costs
_ Draft Return on Investment (RoI) Calculations
_ Draft Total Cost of Ownership (TCO) Calculations
_ Draft Cost/Minute or Cost/User breakdown
_ Draft Cost/Minute or Cost/User analysis by SLA category
_ SLA compliance and out-of-compliance reports


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Design and Pre-Deployment Testing
      _ The Team
          _ Current IP Network Technical
          _ Current IP Network Management
          _ Current Voice Technical
          _ Current Voice Management
          _ Temp/Contract Multi-Technical
          _ Current Multi-Technical (If they exist)
          _ New Multi-Technical
          _ IT/Network Management
          _ Consultant
          _ Manufacturer Professional Services
          _ Internet Service Provider
          _ Data Systems Integrator
          _ Voice Systems Integrator
          _ Telephony Service Provider
      _ Service/Feature Support
          _ Needed telephony features
          _ Voice Band Data (FAX, Modem)
          _ DTMF Support
      _ SLA Compliance Modeling and Prediction
          _ Predict service availability for voice/telephony services
          _ Predict MOS for voice/telephony services
          _ Predict service availability for data services
          _ Predict service availability for video/IPTV services
          _ Predict Time-to-Dial Tone (TDTT) for voice/telephony services
          _ Predict Call Set-Up Time for voice/telephony services
          _ Predict Call Tear-Down Time for voice/telephony services
          _ Predict delay for voice/telephony services
          _ Predict delay for data services
          _ Predict delay for video/IPTV services
          _ Predict delay variation for voice/telephony services
          _ Predict delay variation for data services
          _ Predict delay variation for video/IPTV services
          _ Predict packet loss for voice/telephony services
          _ Predict packet loss for data services
          _ Predict packet loss for video/IPTV services
          _ Distance sensitivity for delay and delay variation
          _ Predict high/low performance for mix traffic loads
          _ Predict Mean-Time-To-Repair
          _ Predict Mean-Time-Between-Failure
          _ Compare different SLA Scenarios
          _ Model Network-Only SLA Compliance
          _ Model Network+Access SLA Compliance

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   _ Model End System-to-End System SLA Compliance
_ Component Simulation
   _Simulate IPT Phone Performance
      •       All codecs
     •       Compression
     •       Encryption
     •       Silence Suppression/Voice Activity Detection
     •       Multi-Line
     •       Multi-Media
     •       Ethernet-10M,100M,1G,WiFi
   _ Simulate VoIP SoftPhone Performance
      •       All codecs
     •       Compression
     •       Encryption
     •       Silence Suppression/Voice Activity Detection
     •       Multi-Line
     •       Multi-Media
      •       Ethernet-10M,100M,1G,WiFi
   _ Simulate Analog Terminal Adapter Performance
      •       All codecs
     •       Compression
     •       Encryption
     •       Silence Suppression/Voice Activity Detection
     •       Multi-Line
     •       Multi-Media
     •       Ethernet-10M,100M,1G,WiFi
   _ Simulate SoftSwitch/IP PBX Performance
      •       Simultaneous Calls
     •       Busy Hour Call Attempts
     •       Max number of stations
   _ Simulate Gateway/BGC Performance
      •       Total Simultaneous Calls
     •       Busy Hour Call Attempts
     •       Security Filtering/Monitoring
_ Power & Power Back-Up

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      _ QoS / QoE Forecasting
         _ Forecast IPT Mean Opinion Score (MOS) using G.107 E-Model
         _ Forecast Data QoS parameters
         _ Forecast Video QoS
         _ Simulate Differentiated Services QoS (DiffServ)
         _ Simulate 802.1p/q VLAN Prioritization
         _ Simulate Congestion Avoidance (RED, WRED, Flow-based WRED)
         _ Simulate QoS Packet Marking
         _ Simulate QoS Congestion Management (WFQ,CBWFQ, FIFO,LLQ, PQ)
         _ Simulate RTP Priority Mechanisms (IP & FR)
         _ Simulate RTCP Feedback Loops
         _ Simulate NetFlow application performance
         _ Simulate impact of access bandwidth differences
         _ Simulate impact of backbone bandwidth differences
      _ Validation of Design
      _ Design Tuning
      _ Test Bed Architecture and Proof of Concept
Implementation and Migration
      _ Planning
          _ Plan Validation
          Visualization
          Role Play
          Naysayer Validation
          Non-Expert Validation
          Red Teams
      _ Implementation Plan
      _ Training Plan(s)
      _ Test/Acceptance Plan(s)
      _ Migration Plan(s)
      _ On-Going Operations Plan(s)
      _ Contingency Plan
      _ Business/System Continuity Plan
      _ Escalation Procedure(s)
      _ Security Issues
          _ Liaison with Security Department
          _ Firewalls & Proxy Server Issues
      _ IP Addresses & Port Numbers
      _ NAT, Internal and External IP Addresses
      _ Internal & External Phone number mapping
      _ Purchasing & Project Management
          _ Establish Site Profiles
          _ Rough-Order-of-Magnitude Pricing
          _ Management Review and Site Prioritization

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    _ Initial Migration/Green Field Plan & Renegotiation of Favorable Contracts
    _ Joint Migration/Green Field Planning Effort
    _ Contracts & Delivery
    _ Surplus Equipment Management Effort
    _ Inventory & Fixed Asset Accounting & Disposition of Old Equipment
_ Implementation of Changes to Business Operations
    _ Training
    _ Dealing with User Reluctance
    _ Standardization
    _ Impact Assessment
    _ Changes to Processes
    _ Staff Implications
_ Review of Standards Status and Maturity
_ Develop Private Net, IP VPN and Carrier/PTT Strategies
_ Private Backbone Design/Upsize for Voice Traffic and Backbone Re-engineering
_ Establish Common Standards and Test Plan
_ Establish Private Net, IP VPN and Carrier/PTT Demo/Test Capabilities
_ Training/Staging for Migration Teams
_ Training for Support Teams
_ Review of Planning and New Telephony Project Roll-Outs
_ Legal
    _ Closing contracts for old telephony system
    _ New contracts
    _ Compliance with National Law & Telco/PTT Regulation
_ Security & Privacy Issues
_ Carrier/PTT IP Bypass & Local Tail Drop-Off
_ PBX and Gateway Issues
_ Contact Centers / Call Centers
    _ Special Considerations & Contingency Planning
    _ Phased Implementation & Call-taker Training




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Ongoing Operations
      _ Review the original plans, goals and objectives
          Are the original goals and objectives still valid?
          Have technologies changed significantly?
          Are standards more mature, and does it matter?
          Have laws or regulations in any of the areas in which we operate changed in
          any important ways?
          Have our operating procedures or business environment evolved in a way that
          impacts our new telephony system?
      _ Growth Management
          _ Fixed Assets and Equipment Budgets
          _ Network Bandwidth: Access and Backbone
          _ Shared Resources: Switches, Routers, Call Servers, IP PBXs, Gateways
          _ Numbers and Licenses
          _ Record Keeping & Documentation
      _ Fixed Asset Accounting
      _ Shipping / Warehousing / Tracking
      _ Repair / Refurbishment / Return to Service / End-of-Life
Optimization
      _ Constant Testing and Monitoring
      _ Exception Reporting and Filtering
       _ “What If” Scenarios and Contingency Plans
      _ Trending and Capacity Planning
          _ Bandwidth
          _ Ports and Lines
          _ Phone Numbers and Numbering Plan
          _ Grooming
          _ VLAN Capacity
          _ VPN Capacity
          _ Infrastructure
      _ Hardware
          _ Repair
          _ Obsolescence / Refresh
      _ Software
          _ Maintenance
          _ Enhancements
          _ Upgrades
          _ Patches & Security Audit




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       _ SLA Improvements
          _ Delay
              _ Call Set-up
              _ During Call
              _ Call Tear Down / Release
          _ Delay Variation
          _ Sample and Packet Loss
          _ Availability
              _ Telephony Service Availability
              _ Grade of Service (GoS) Measurement
              _ Busy Hour Call Attempts (BHCAs)
              _ Busy Hour Call Completions (BHCCs)
          _ Price and Cost Optimization
              _ Hardware Costs
              _ Service and Support Costs
              _ In-Sourcing vs. Out-Sourcing
          _ IP Contact Center Optimization
Operations & Security
      _ Tracking of SLA parameters: packet loss
      _ Tracking of SLA parameters: delay
      _ Tracking of SLA parameters: delay variation/jitter
      _ Tracking of SLA parameters: availability
      _ Tracking of SLA parameters: user profiles
      _ Tracking of SLA parameters: voice quality metrics
      _ Tracking of SLA parameters: MTBF
      _ Tracking of SLA parameters: MTTR
      _ Tracking of SLA parameters: Bandwidth per call
      _ SLA Compliance Exception Reporting
      _ Tracking of protocols used
      _ Tracking of services and applications used
      _ Tracking of out-of-service time for circuits
      _ Tracking of out-of-service time for components
      _Availability tracking for TDM circuits
      _ Security events reporting: encryption and key management
      _ Security events reporting: tunnels and secure tunnels
      _ Security events reporting: user access info & violations
      _ Security events reporting: DoS attempts
      _ Security events reporting: firewall performance/blocked intrusions
      _ Security events reporting: SW release levels & patch history
      _ Security events reporting: Access Retries
      _ Security events reporting: Session Hijacking Attempts
      _ Security events reporting: Eavesdropping Attempts




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Administration
     _ Track/Audit service provider call detail records (CDR)
     _ Track/Audit bandwidth and resource usage by user and user profile
     _ Track/Audit minutes of use by user and user profile
     _ Track/Audit applications by user and user profile
     _ Track/Audit features and services used by user and user profile
     _ Reconcile Telco Carrier Billing
     _ Reconcile Managed Service Provider Billing
     _ Reconcile Service Provider Billing
     _ Automate SLA Compliance Penalties
     _ Report Calculated MOS by user and user profile
     _ Correlate MOS with SLA compliance
     _ Report costs per user and user profile
Provisioning
       _ Remote configuration of hard phones
       _ Remote configuration of soft phones
       _ Remote configuration of call servers
       _ Remote configuration of IP PBXs
       _ Remote configuration of gateways and Session Border Controllers
       _ Implement Secure Shell (SSH)
       _ Supports multiple levels of service technician log-in
       _ Tracking of moves/adds/changes by service tech
       _ Automated fall-back to last known good configuration
       _ Automated bulk configuration using profiles and templates
       _ Pre-configuration of IP Phones and ATAs prior to shipping
       _ Tracking of user changes
       _ Support for Tripwire or similar anti-tampering functionality
       _ Other configuration stabilization and anti-tampering
       _ Support for secure FTP
       _ Support of Trivial File Transfer Protocol for config download
       _ Support of anonymous File Transfer Protocol for config download
       _ Support of Telnet for remote configuration




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Maintenance
      _ Track moves/adds/changes
      _ Track software levels, patches and upgrades
      _ Track firmware levels, patches and upgrades
      _ Track/Audit IP hard phones and ATAs
      _ Track/Audit IP soft phones
      _ Track/Audit IP PBXs
      _ Track/Audit IP gateways and Session Border Controllers (SBC)
      _ Track/Audit services and features used
      _ Testing of Services and Features
      _ Remote Testing of Services and Features
      _ Automated Testing of Services and Features
      _ Hardware preventative maintenance
      _ Spare inventory management
      _ Maintenance contract tracking
      _ Warranty tracking
      _ Service and Trouble Ticket Tracking
      _ Trouble Ticket Trends and Statistics
      _ Automated/Scripted Root Cause Analysis (RCA)
      _ Escalation Procedures




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