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The Basics

of
Telecommunications

Presented by the
International Engineering Consortium

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ISSN: 0886-229X
ISBN: 0-933217-84-6

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The Basics of Telecommunications


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Table of Contents
Asymmetric Digital Subscriber Line (ADSL) .........................................1
Asynchronous Transfer Mode (ATM) Fundamentals ............................15
Cable Modems .......................................................................................41
Cellular Communications.......................................................................57
Fiber-Optic Technology .........................................................................81
Fundamentals of Telecommunications.................................................103
Intelligent Networks (INs) ...................................................................129
Internet Access .....................................................................................161
Internet Telephony................................................................................183
Intranets and Virtual Private Networks (VPNs) ..................................199
Operations Support Systems (OSSs)....................................................211

Optical Networks .................................................................................235


Personal Communications Services (PCS) ..........................................265
Signaling System 7 (SS7) ....................................................................291
Synchronous Optical Networks (SONETs)..........................................321
Voice and Fax over Internet Protocol (V/FoIP) ...................................387
Self-Test Correct Answers....................................................................417

vi

Asymmetric Digital
Subscriber Line (ADSL)
DEFINITION
Asymmetric digital subscriber line (ADSL) is a new modem technology that
converts existing twisted-pair telephone lines into access paths for high-speed
communications of various sorts.
ADSL can transmit more than 6 Mbps to a subscriberenough to provide
Internet access, video-on-demand, and LAN access. In interactive mode it can
transmit more than 640 kbps in both directions. This increases the existing
access capacity by more than fifty-fold, enabling the transformation of the
existing public network. No longer is it limited to voice, text, and low-resolution graphics. It promises to be nothing less than an ubiquitous system that
can provide multimedia (including full-motion video) to the entire country.
ADSL can perform as indicated in Table 1.

Table 1: ADSL Data Rates as a Function of Wire and Distance


Data Rate

Wire Gauge

Distance

Wire Size

Distance

1.5-2 Mbps

24 AWG

18,000 ft.

.5 mm

5.5 km

1.5-2 Mbps

26 AWG

15,000 ft.

.4 mm

4.6 km

6.1 Mbps

24 AWG

12,000 ft.

.5 mm

3.7 km

6.1 Mbps

26 AWG

9,000 ft.

.4 mm

2.7 km

The Basics of Telecommunications

TOPICS
1.

A SHORT HISTORY OF ANALOG MODEMS . . . . . . . . . . . . . . . . . . . .3

2.

THE ANALOG MODEM MARKET . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

3.

DIGITAL SUBSCRIBER LINE (DSL) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .6

4.

XDSL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .7

5.

THE MODEM MARKET . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .8

6.

ATM VERSUS IP TO THE DESKTOP . . . . . . . . . . . . . . . . . . . . . . . . . . .8

7.

CAP VERSUS DMT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .10

8.

THE FUTURE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .10

9.

SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .11

10. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

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1. A SHORT HISTORY OF ANALOG MODEMS


The term modem is actually an acronym that stands for modulation/demodulation. A modem enables two computers to communicate by using the public
switched telephone network (PSTN). This network can only carry sounds, so
modems must translate the computers digital information into a series of
high-pitched sounds that can be transported over the phone lines. When the
sounds arrive at their destination, they are demodulated or turned back into
digital information for the receiving computer (see Figure 1).

Figure 1. Analog Modem


All modems use some form of compression and error correction.
Compression algorithms enable throughput to be enhanced two to four times
over normal transmission. Error correction examines incoming data for
integrity and requests retransmission of a packet when it detects a problem.

2. THE ANALOG MODEM MARKET


The dynamics of the analog-modem market can be traced back to July 1968
when, in its landmark Carterfone decision, the FCC ruled that the provisions prohibiting the use of customer-provided interconnecting devices were
unreasonable.
On January 1, 1969, AT&T revised its tariffs to permit the attachment of customer-provided devices (such as modems) to the public switched network
subject to three important conditions:
The customer-provided equipment was restricted to certain output power and
energy levels so as not to interfere with or harm the telephone network in
any way.

The Basics of Telecommunications


The interconnection to the public switched network had to be made through
a telephone companyprovided protective device, sometimes referred to as a
data access arrangement (DAA).
All network-control signaling such as dialing, busy signals, and so on had to be
performed with telephone-company equipment at the interconnection point.
By 1976, the FCC had recommended a plan whereby current protective
devices would be phased out in favor of a so-called registration plan.
Registration would permit direct switched-network electrical connection of
equipment that had been inspected and registered by an independent agency
such as the FCC as technically safe for use on the switched network.
In the post-war era, heavy emphasis on information theory led to the profound and now famous 1948 paper by Claude Shannon, providing us with a
concise understanding of channel capacity for power and bandlimited gaussian noise channels (i.e., our analog telephone channel).
C = Bw * Log2(1+S/N)
This simply states that the channel capacity, C, is equal to the available channel bandwidth, Bw, times the log base 2 of 1 plus the signal-to-noise ratio in
that bandwidth. It does not explain how to accomplish this; it simply states
that this channel capacity can be approached with suitable techniques.
As customers started buying and using modems, speed and reliability became
important issues. Each vendor tried to get as close to the limit expressed by
Shannons Law as possible. Until Recommendation V.32, all modem standards
seemed to fall short of this capacity by 9 to 10 db S/N. Estimates of the channel capacity used assumed bandwidths of 2400 Hz to 2800 Hz, and S/N ratios
from 24 db to 30 db and generally arrived at a capacity of about 24,000 bits
per second (bps). It was clear that error-correction techniques would have to
become practical before this gap would be diminished.
Modems of the 1950s were all proprietaryprimarily FSK (300 bps to 600
bps) and vestigial sideband (1200 bps to 2400 bps). These devices used or
were built upon technology from RF radio techniques developed during the
wartime era and applied to wireline communications.

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International standardization of modems started in the 1960s. In the 1964
Plenary, the first CCITT Modem Recommendation, V.21 (1964), a 200bps
FSK modem (and now 300 bps), was ratified and is (still) used in the V.34/V.8
handshake. The preferred modulation progressed to 4-Phase (or 2X2 QAM) in
1968, and to 4X4 QAM with V.22bis in 1984. Additionally, in 1984, the next
major technological advancement in modem recommendations came with
V.32 and the addition of echo cancellation and trellis coding.
Trellis codes, first identified by Dr. Gottfred Ungerboeck, were a major breakthrough in that they made it practical to provide a level of forward error correction to modems, realizing a coding gain of 3.5 db, and closing over a third
of the gap in realizing the Shannon channel capacity. Recommendation
V.32bis built on this and realized improvement in typical-connection S/N
ratios and increased the data rates to 14,400 bps.
As work on V.34 started in earnest (1989/90), a recognition of further
improvement in the telephone networks in many areas of the world was evident. With this recognition, the initial goal of 19,200 bps moved to 24,000 bps
and then to 28,800 bps. The newer V.34 (1996) modem supports 33,600 bps.
Such modems achieve 10 bits per Hertz of bandwidth, a figure that approaches the theoretical limits. Recently, a number of companies have introduced a
56.6kbps analog modem designed to operate over standard phone lines.
However, the modem is asymmetrical (it operates at normal modem speeds
on the upstream end), and it requires a dedicated T1/E1 connection to the ISP
site to consistently reach its theoretical limits. For users without such a line,
the modem offers, inconsistently at best according to reports, a modest gain
in performance.
However, the bandwidth limitations of voice bandlines are not a function of
the subscriber line but the core network. Filters at the edge of the core network limit voice-grade bandwidth to approximately 3.3 kHz. Without such
filters, the copper access wires can pass frequencies into the MHz regions.
Attenuation determines the data rate over twisted-pair wire, and it, in turn, is
a function of line length and frequency. Table 1 indicates the practical limits
on data rates in one direction compared to line length.

The Basics of Telecommunications

3. DIGITAL SUBSCRIBER LINE (DSL)


Despite its name, DSL does not refer to a physical line but to a modem or
rather a pair of modems. A DSL modem pair creates a digital subscriber line,
but the network does not purchase the lines when it buys ADSL. It already
owns the lines; it purchases modems.
A DSL modem transmits duplex (i.e., data in both directions simultaneously)
at 160 kbps over copper lines of up to 18,000 feet. DSL modems use twistedpair bandwidth from 0 to approximately 80 kHz, which precludes the simultaneous use of analog telephone service in most cases (see Figure 2).

Figure 2. DSL Modem


T1 and E1
In the early 1960s, Bell Labs engineers created a voice multiplexing system
that digitized a voice sample into a 64kbps data stream (8,000 voltages samples per second) and organized these into a 24-element framed data stream
with conventions for determining precisely where the 8-bit slots went at the
receiving end. The frame was 193 bits long and created an equivalent data
rate of 1.544 Mbps. The engineers called their data stream DS1, but it has
since come to be known as T1. Technically, though, T1 refers to the raw data
rate, with DS1 referring to the framed rate.
In Europe, the worlds public telephone networks other than AT&T modified
the Bell Lab approach and created E1a multiplexing system for 30 voice
channels running at 2.048 Mbps.
Unfortunately, T1/E1 is not really suitable for connection to individual residences. The transmission protocol they used, alternate mark inversion (AMI),
required tranceivers 3,000 feet from the central office and every 6,000 feet
thereafter. AMI demands so much bandwidth and corrupts the cable spectrum so much that telephone companies could use only one circuit in any 50pair cable and none in any adjacent cables. Under these circumstances, pro-

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viding high-bandwidth service to homes would be equivalent to installing
new wire.

4. XDSL
High Data-Rate Digital Subscriber Line (HDSL)
HDSL is simply a better way of transmitting T1/E1 over copper wires, using
less bandwidth without repeaters. It uses more advanced modulation techniques to transmit 1.544 Mbps over lines up to 12,000 feet long.
Single-Line Digital Subscriber Line (SDSL)
SDSL is a single-line version of HDSL, transmitting T1/E1 signals over a single
twisted pair and able to operate over the plain old telephone service (POTS)
so that a single line can support POTS and T1/E1 at the same time. It fits the
market for residence connection, which must often work over a single telephone line. However, SDSL will not reach much beyond 10,000 feet. At the
same distance, ADSL reaches rates above 6 Mbps.
Asymmetric Digital Subscriber Line (ADSL)
ADSL is intended to complete the connection with the customers premises.
It transmits two separate data streams with much more bandwidth devoted
to the downstream leg to the customer than returning. It is effective because
symmetric signals in many pairs within a cable (as occurs in cables coming
out of the central office) significantly limit the data rate and possible line
length.
ADSL succeeds because it takes advantage of the fact that most of its target
applications (video-on-demand, home shopping, Internet access, remote LAN
access, multimedia, and PC services) function perfectly well with a relatively low
upstream data rate. MPEG movies require 1.5 or 3.0 Mbps downstream but need
only between 16 kbps and 64 kbps upstream. The protocols controlling Internet
or LAN access require somewhat higher upstream rates but in most cases can
get by with a 10 to 1 ratio of downstream to upstream bandwidth.

The Basics of Telecommunications

5. THE MODEM MARKET


Sales in the modem business started out slowly until customers started buying PCs. Likewise, costs were high until the volumes picked up. When the
14.4kbps modem was first introduced, it cost $14,400 or one dollar per bit.
Today, a much faster consumer-level modem with many more features costs
only $100 to $300, making it unusual for a home PC today to be without a
modem.
Over the years, customers watched modem vendors evolve their products on
a standards basis. This technique, although somewhat time consuming, was
important and led to significant feature enhancement. Initially, several modulation schemes were in use, but by the time the V.34 modem came out, all of
the major modem-modulation schemes were combined in that standard
giving the customer one modem that could be used in many applications. As
the modem market matured, customers became less concerned with the internals of standards and more concerned with features, size, and flexibility.
As a result of the progress in analog-modem technology and with the advent
of mass-market consumer-level PCs, there are over 500 million modems in
the world today.
The xDSL modem market will follow similar market patterns. Today, things
like modulation schemes, the type of protocol supported to the home or
small business, and costs of the units are the main topics. As the xDSL market
matures, most likely in a fashion similar to that of the analog modem, customers will become less concerned with modulation and protocols. On the
other hand, they will look for vendors that provide plug-and-play interoperability with their data equipment, ease of installation, the best operating characteristics on marginal lines, and minimalist size and power requirements.

6. ATM VERSUS IP TO THE DESKTOP


There is a great debate raging among potential service providers as to
whether there should be standard IP10t connections or ATM connections to
their customers PCs. The two are very similarthe difference is in the
specifics of the equipment and not in the amount of equipment required.
There are various advantages to each method of network access.

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IP Advantages
10t Ethernet is basically self-learning.
Inexpensive LAN PC cards already exist.
10t is an industry standard.
LAN networks are proven and work today.
There is much expertise in this technology.
PC software and OS drivers already interface to IPbased LANs.
ATM Advantages
Streaming video transport has already been proven.
Mixing of services (e.g., video, telephony, and data) is much easier.
Traffic speeds conform to standard telephony transport rates (e.g., DS3,
STS1).
New PC software and drivers will work with ATM.
The issue actually gets more interesting because both architectures usually
interface to an ATM backbone network for high-speed connections over a
wide area. Therefore, the real issues are the costs of building the network, the
services that are to be carried over it, and the time frame for the implementation. If the need is for data services (Internet connections, work-at-home, and
etc.), the obvious choice is an IP network. The hardware and software
required to implement this network is available and relatively inexpensive.
ATM would be the solution for multiple mixed QoS service requirements in
the near future. It is true that the IP technology is being extended to offer
tiered QoS with RSVP, and IP telephony is being refined to operate more efficiently. The paradox, however, is that these standards do not exist today.
ATM standards are quite complete. However, not all may be easily implementable. In spite of this, there are many ATM networks in existence or currently under construction.
This leaves the issue of costs. The true costs of creating and operating a largescale data-access network are not known. True, there are portions that are
understood, but many others are only projected. This creates great debate
over which technology is actually less costly. The only way for the costs to
be really known is to build reasonably large networks and compare costs. If
one technology is a clear winnera somewhat doubtful hypothesisthen
use that technology. If there is no clear cost advantage, then build the
network with the service set that matches the service needs of the potential

The Basics of Telecommunications


customers. The issue is to start the implementation phase where the real
answers will be determined and subsequently end the interminable discussion
phase.

7. CAP VERSUS DMT


These are the two primary xDSL standards over which much debate has
ensued. Although the debate continues, the real action is taking place in the
marketplace. CAP demonstrated a clear lead in getting the product to market.
Chips were available in quantity, and they worked. Numerous products that
incorporated these chips are installed in a number of locations by service
providers. Standards and interoperability issues between vendors and implementations are now being addressed.
DMT, on the other hand, has been in the standards arena for some time and
continues to evolve. It is now considered a standard by a number of service
providers. This technology featured some innovations that were not originally
in the CAP feature set such as rate adaptation. On the other hand, the chips
are just now finding their way into products. Trial activities are only now
beginning, and advanced chip sets that match the features of CAP chips are
now being promised for 3Q97.
The issue is which will win the market. The service providers who are building the xDSL network will select the technology that meets their needs.
Many vendors are offering products that use either technology. Some new
chips are being announced that allow adaptation between either technology.
The point here is that the technology of xDSL chips is not a roadblock to
deployment. Either appears to work well, and true interoperability remains in
the future, much like midspan meets for SONET equipment.

8. THE FUTURE
Look at the past of analog modems to crystal-ball the future of xDSL.
Standards were an issue with modems and will be an issue with xDSL products. However, it is not obvious to a technologist what technology will win
out. Remember that in the VCR arena, Betamax had the better-quality picture, but VHS eventually won out. In any event, only the marketplace and
time will answer these questions.
10

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Asymmetric Digital Subscriber Line (ADSL)

9. SELF-TEST
Multiple Choice
1.

ADSL increases existing twisted-pair access capacity by

a. two-fold
b. three-fold
c. thirty-fold
d. fifty-fold
2.

A modem translates

a. analog signals into digital signals


b. digital signals into analog signals
c. both of the above
3.

The 1948 theorem, which is the basis for understanding the relationship of channel capacity, bandwidth, signal-to-noise ratio, is known as
.
a. The Peter Principle
b. The Heisenberg Uncertainty Principle
c. Shannons Law
d. Boyles Law

4.

What appears to be the practical limit for analog modems over the standard
telephone network?
a. 33 kbps b. 28.8 kbps c. 24 kbps d. 19.2 kbps

5.

Digital subscriber line (DSL) refers to

a. a specific gauge of wire used in modem communications


b. a modem enabling high-speed communications
c. a connection created by a modem pair enabling high-speed
communications
d. a specific length of wire

11

The Basics of Telecommunications

Multiple Choice
6

What is the source of limitation on the bandwidth of the public switched network?
a. subscriber line
b. the core network

7.

The practical upper limit of line length of ADSL lines is

a. 6,000 ft.
b. 12,000 ft.
c. 18,000 ft.
d. 36,000 ft.
8.

T1 and DS1 refer to the same multiplexing system. Which one is generally used
to refer to the raw data rate?
a. T1
b. DS1

True or False
9.

T1/E1 and HDSL are essentially equivalent technologies.


a. true
b. talse

10. ADSL cannot handle Internet or LAN access.


a. true
b. false

12

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10. ACRONYM GUIDE


ADSL

asymmetric digital subscriber line

AMI

alternate mark inversion

ATM

asynchronous transfer mode

CAP

cellular array processor

DAA

data access arrangement

DMT

discrete multitone

DSL

digital subscriber line

FCC

Federal Communications Commission

HDSL

high data-rate subscriber line

IP

Internet Protocol

LAN

local-area network

Modem

modulation/demodulation

MPEG

Motion Pictures Expert Group

QoS

quality of service

SDSL

single-line digital subscriber line

SONET

synchronous optical network

13

Asynchronous Transfer Mode (ATM)


Fundamentals
DEFINITION
Asynchronous transfer mode (ATM) is a high-performance, cell-oriented
switching and multiplexing technology that utilizes fixed-length packets to
carry different types of traffic. ATM is a technology that will enable carriers
to capitalize on a number of revenue opportunities through multiple ATM
classes of services; high-speed local area network (LAN) interconnection;
voice, video, and future multimedia applications in business markets in the
short term; and in community and residential markets in the longer term.

TUTORIAL OVERVIEW
Changes in the structure of the telecommunications industry and market conditions have brought new opportunities and challenges for network operators
and public service providers. Networks that have been primarily focused on
providing better voice services are evolving to meet new multimedia communications challenges and competitive pressures.
Services based on ATM and synchronous digital hierarchy/synchronous optical network (SDH/SONET) architectures provide the flexible infrastructure
essential for success in this evolving market (see Figure 1).

15

The Basics of Telecommunications

Figure 1. ATM Technology, Services, and Standards Pyramid


ATM, which was once envisioned as the technology of future public networks, is now a reality, with service providers around the world introducing
and rolling out ATM and ATMbased services. The ability to successfully
exploit the benefits of ATM technology within the public network will provide strategic competitive advantage to both carriers and enterprises.
In addition to revenue opportunities, ATM reduces infrastructure costs
through efficient bandwidth management, operational simplicity, and the
consolidation of overlay networks. Carriers can no longer afford to go
through the financial burden and time required to deploy a separate network
for each new service requirement (e.g., dedicating a network for a single service such as transparent LAN or frame relay). ATM technology will allow
core network stability while allowing service interfaces and other equipment
to evolve rapidly.

16

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TOPICS
1.

DEFINITION OF ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .18

2.

BENEFITS OF ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .19

3.

ATM TECHNOLOGY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .20

4.

ATM CLASSES OF SERVICES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .22

5.

ATM STANDARDS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .25

6.

ATM LAN EMULATION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .27

7.

VOICE OVER ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .29

8.

VIDEO OVER ATM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .31

9.

ATM TRAFFIC MANAGEMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .32

10. ATM APPLICATIONS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .34


11. NORTEL NETWORKS ATM VISION . . . . . . . . . . . . . . . . . . . . . . . . . .36
12. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .37
13. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .39

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The Basics of Telecommunications

1. DEFINITION OF ATM
ATM is a technology that has its history in the development of broadband
ISDN in the 1970s and 1980s. Technically, it can be viewed as an evolution of
packet switching. Like packet switching for data (e.g., X. 25, frame relay,
transmission control protocol/Internet protocol [TCP/IP]), ATM integrates the
multiplexing and switching functions, is well suited for bursty traffic (in contrast to circuit switching), and allows communications between devices that
operate at different speeds. Unlike packet switching, ATM is designed for
high-performance multimedia networking. ATM technology has been implemented in a broad range of networking devices:
PC workstation and server network interface cards
switched-Ethernet and token-ring workgroup hubs
workgroup and campus ATM switches
ATM enterprise network switches
ATM multiplexors
ATMedge switches
ATMbackbone switches
ATM is also a capability that can be offered as an end-user service by service
providers (as a basis for tariffed services) or as a networking infrastructure for
these and other services. The most basic service building block is the ATM
virtual circuit, which is an end-to-end connection that has defined end points
and routes but does not have bandwidth dedicated to it. Bandwidth is allocated on demand by the network as users have traffic to transmit. ATM also
defines various classes of service to meet a broad range of application needs.
ATM is also a set of international interface and signaling standards defined by
the International Telecommunications Union (ITU) Telecommunications
Standards Sector. The ATM Forum has played a pivotal role in the ATM market since its formulation in 1991. The ATM Forum is an international voluntary organization composed of vendors, service providers, research organizations, and users. Its purpose is to accelerate the use of ATM products and
services through the rapid convergence of interoperability specifications, promotion of industry cooperation, and other activities. It does this by developing multivendor implementation agreements.
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2. BENEFITS OF ATM
The benefits of ATM are listed in Table 1 below.
Table 1. Benefits of ATM

The Advantages of ATM


high performance via hardware switching
dynamic bandwidth for bursty traffic
class-of-service support for multimedia
scalability in speed and network size
common LAN/WAN architecture
opportunities for simplification via VC architecture
international standards compliance

The high-level benefits delivered through ATM services deployed on ATM


technology using international ATM standards can be summarized as follows:
high performance via hardware switching with terabit switches on the horizon
dynamic bandwidth for bursty traffic meeting application needs and delivering high utilization of networking resources; most applications are or can be
viewed as inherently bursty; data applications are LANbased and are very
bursty; voice is bursty, as both parties are neither speaking at once nor all
the time; video is bursty, as the amount of motion and required resolution
varies over time
class-of-service support for multimedia traffic allowing applications with
varying throughput and latency requirements to be met on a single network

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The Basics of Telecommunications


scalability in speed and network size supporting link speeds of T1/E1 to
OC12 (622 Mbps) today and into the multiGbps range before the end of
the decade; networks that scale to the size of the telephone network (i.e.,
as required for residential applications) are envisaged
common LAN/WAN architecture allowing ATM to be used consistently
from one desktop to another; traditionally, LAN and WAN technologies
have been very different, with implications for performance and interoperability
opportunities for simplification via switched VC architecture; this is particularly true for LANbased traffic, which today is connectionless in nature;
the simplification possible through ATM VCs could be in areas such as
billing, traffic management, security, and configuration management
international standards compliance in central-office and customer-premises
environments allowing for multivendor operation

3. ATM TECHNOLOGY
In ATM networks, all information is formatted into fixed-length cells consisting of 48 bytes (8 bits per byte) of payload and 5 bytes of cell header (see
Figure 2). The fixed cell size ensures that time-critical information such as
voice or video is not adversely affected by long data frames or packets. The
header is organized for efficient switching in high-speed hardware implementations and carries payload-type information, virtual-circuit identifiers, and
header error check.

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Figure 2. Cell Structure


ATM is a connection-oriented technology. Organizing different streams of
traffic in separate cells allows the user to specify the resources required and
allows the network to allocate resources based on these needs. Multiplexing
multiple streams of traffic on each physical facility (between the end user and
the network or between network switches)combined with the ability to
send the streams to many different destinationsenables cost savings
through a reduction in the number of interfaces and facilities required to construct a network.
ATM standards defined two types of ATM connections: virtual path connections (VPCs), which contain virtual channel connections (VCCs).

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The Basics of Telecommunications


A virtual channel connection (or virtual circuit) is the basic unit, which carries
a single stream of cells, in order, from user to user.
A collection of virtual circuits can be bundled together into a virtual path connection. A virtual path connection can be created from end-to-end across an
ATM network. In this case, the ATM network does not route cells belonging
to a particular virtual circuit. All cells belonging to a particular virtual path are
routed the same way through the ATM network, thus resulting in faster
recovery in case of major failures.
An ATM network also uses virtual paths internally for the purpose of
bundling virtual circuits between switches. Two ATM switches may have
many different virtual channel connections between them, belonging to different users. These can be bundled by the two ATM switches into a virtual
path connection. This can serve the purpose of a virtual trunk between the
two switches. This virtual trunk can then be handled as a single entity by,
perhaps, multiple intermediate virtual path cross connects between the two
virtual circuit switches.
Virtual circuits can be statically configured as permanent virtual circuits
(PVCs) or dynamically controlled via signaling as switched virtual circuits
(SVCs). They can also be point-to-point or point-to-multipoint, thus providing
a rich set of service capabilities. SVCs are the preferred mode of operation
because they can be dynamically established, thus minimizing reconfiguration
complexity.

4. ATM CLASSES OF SERVICES


ATM is connection oriented and allows the user to dynamically specify the
resources required on a per-connection basis (per SVC). Five classes of service
are defined for ATM (as per ATM Forum UNI 4.0 specification). The QoS
parameters for these service classes are summarized in Table 2:

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Table 2: ATM Service Classes

Service

Class

Constant bit rate (CBR)

This class is used for emulating circuit


switching. The cell rate is constant with
time. CBR applications are quite sensitive
to cell-delay variation. Examples of applications that can use CBR are telephone
traffic (i.e., nx64 kbps), videoconferencing,
and television.

Variable bit ratenon


real time (VBRNRT)

This class allows users to send traffic at a


rate that varies with time depending on the
availability of user information. Statistical
multiplexing is provided to make optimum
use of network resources. Multimedia
e-mail is an example of VBRNRT.

Variable bit rate


real time (VBRRT)

This class is similar to VBR-NRT but is


designed for applications that are sensitive
to cell-delay variation. Examples for realtime VBR are voice with speech activity
detection (SAD) and interactive compressed video.

Available bit rate (ABR)

This class of ATM services provide ratebased flow control and is aimed at data
traffic such as file transfer and e-mail.
Although the standard does not require the
cell transfer delay and cell-loss ratio to be
guaranteed or minimized, it is desirable for
switches to minimize delay and loss as
much as possible. Depending on the
state of congestion in the network, the
source is required to control its rate. The
users are allowed to declare a minimum
cell rate, which is guaranteed to the
connection by the network.

Unspecified bit rate


(UBR)

This class is the catch-all other class, and


is widely used today for TCP/IP.

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The Basics of Telecommunications


The ATM Forum has identified the following technical parameters to be associated with a connection. These terms are outlined in Table 3:
Table 3: ATM Technical Parameters

Technical Parameter Definition


Cell-loss ratio (CLR)

Cell-loss ratio is the percentage of cells not


delivered at their destination because they
were lost in the network as a result of
congestion and buffer overflow.

Cell-transfer delay (CTD) The delay experienced by a cell between


network entry and exit points is called the
cell-transfer delay. It includes propagation
delays, queuing delays at various
intermediate switches, and service times at
queuing points.
Cell-delay variation (CDV) Cell-delay variation is a measure of the
variance of the cell-transfer delay. High
variation implies larger buffering for delaysensitive traffic such as voice and video.
Peak cell rate (PCR)

Peak cell rate is the maximum cell rate at


which the user will transmit. PCR is the
inverse of the minimum cell interarrival time.

Sustained cell rate (SCR) This is the average rate, as measured over
a long interval, in the order of the
connection lifetime.
Burst tolerance (BT)

This parameter determines the maximum


burst that can be sent at the peak rate.
This is the bucket-size parameter for the
enforcement algorithm that is used to
control the traffic entering the network.

Finally, there are a number of ATM classes of service. These classes are all
outlined in Table 4 below:
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Table 4: ATM Classes of Services

Services

CBR

CLR cell-loss ratio

Yes

Yes

CTD cell-transfer
delay

Yes

CDV cell-delay
variation
PCR peak cell rate

ABR

UBR

Yes

Yes

No

No

Yes

No

No

Yes

Yes

Yes

No

No

Yes

Yes

Yes

No

Yes

SCR sustained cell rate No

Yes

Yes

No

No

Burst tolerance @ PCR No

Yes

Yes

No

No

Flow control

No

No

Yes

No

No

VBRNRT VBRRT

Its extensive class-of-service capabilities makes ATM the technology of choice


for multimedia communications.

5. ATM STANDARDS
The ATM Forum has identified a cohesive set of specifications that provides a
stable ATM framework. The first and most basic ATM standards are those
which provide the end-to-end service definitions as described in Section 4.
An important ATM standard and service concept is that of service interworking between ATM and frame relay (a fast-growing pervasive service), whereby ATM services can be seamlessly extended to lower-speed frame-relay
users. Frame relay is a network technology that is also based on virtual circuits using variable-length frame transmission between users.
ATM user network interface (ATM UNI) standards specify how a user connects to the ATM network to access these services. A number of standards
have been defined for T1/E1, 25 Mbps, T3/E3, OC3 (1.55 Mbps), and OC12
with OC48 (2.4 Gbps) in the works. OC3 interfaces have been specified for
use over single-mode fiber (for wide-area applications) and over unshielded
twisted pair or multimode fiber for lower-cost-in-building applications.

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The Basics of Telecommunications


Two ATM networking standards have been defined that provide connectivity
between network switches and between networks:
Broadband intercarrier interface (BICI) and PNNI (the P being for public
or private, while NNI is for network-to-network interface or node-tonode interface).
PNNI is the more feature-rich of the two and supports class of servicesensitive routing and bandwidth reservation. It provides topology-distribution
mechanisms based on advertisement of link metrics and attributes, including
bandwidth metrics. It uses a multilevel hierarchical routing model providing
scalability to large networks. Parameters used as part of the path-computation
process include the destination ATM address, traffic class, traffic contract,
QoS requirements, and link constraints. Metrics that are part of the ATM
routing system are specific to the traffic class and include quality of
servicerelated metrics (e.g., CTD and CLR) and bandwidth-related metrics
(e.g., PCR). The path computation process includes overall network-impact
assessment, avoidance of loops, minimization of rerouting attempts, and use
of policy (inclusion/exclusion in rerouting, diverse routing, and carrier selection). Connection admission controls (CACs) define procedures used at the
edge of the network, whereby the call is accepted or rejected based on the
ability of the network to support the requested QoS. Once a VC has been
established across the network, network resources must be held and quality
service guaranteed for the duration of the connection.
All ATM traffic is carried in cells, yet no applications use cells. So, specific
ways of putting the data into cells are defined to enable the receiver to reconstruct the original traffic. Three important schemes are highlighted in Figure 3
and discussed in detail later in the tutorial.
1. RFC1483 specifies how interrouter traffic is encapsulated into ATM using
ATM adaptation layer 5 (AAL5). AAL5 is optimized for handling framed traffic and has similar functionality to that provided by HDLC framing in frame
relay, SDLC, and X.25.
2. ATM LAN emulation (LANE) and multiprotocol over ATM (MPOA) are
designed to support dynamic use of ATM SVCs primarily for TCP/IP. LANE is
a current standard that is widely deployed and will be a subset of the MPOA
standard. It will be discussed later in the tutorial.

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3. Voice and video adaptation schemes can use AAL1, which is defined for
high efficiency (i.e., for traffic that has no natural breaks, such as a circuit carrying bits at a fixed rate).

Figure 3. Three ATM Schemes

6. ATM LAN EMULATION


ATMbased Ethernet switches and ATM workgroup switches are being
deployed by end users at various corporate sites. The most widely used set of
standards in local ATM environments is ATM LAN emulation (LANE) (see
Figure 4). ATM LAN emulation is used to make the ATM SVC network
appear to be a collection of virtual-Ethernet/IEEE 802.3 and token-ring/IEEE
802.5 LANs. The replication of most of the characteristics of existing LANs
means that LAN emulation enables existing LAN applications to run over
ATM transparently, this latter characteristic leading to its wide deployment.
In ATM LAN emulation, most unicast LAN traffic moves directly between
clients over direct ATM SVCs, while multicast traffic is handled via a server
functionality. Bridging is used to interconnect real LANs and emulated LANs
running on ATM, while routing is used to interconnect ATMemulated LANs
and other WAN or LAN media for purposes of routing scalability, protocol
spoofing, or security firewalls.

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The Basics of Telecommunications

Figure 4. ATM LAN Emulation (LANE)


The ATM Forum LANE implementation agreement specifies two types of
LANE network components connected to an ATM network:

LANE Network Components in an ATM Network


1. LANE clients that function as end systems such as the following:
computers with ATM interfaces that operate as file servers
end-user workstations or personal computers
Ethernet or token-ring switches that support ATM networking
routers, bridges, and ATM ENS with membership in an emulated ATM
LAN
2. LANE servers that support ATM LANE service for configuration
management, multicast support, and address resolution
The LANemulation service may be implemented in the same devices as
clients or involve other ATM network devices. The communications interface,
LAN emulation user-network interface (LUNI), is the sequence and contents
of the messages that the clients ultimately use to transfer traffic of the type
expected on IEEE 802.3/5 LANs. The component of the LANemulation
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service that deals with initialization (i.e., emulates plugging the terminal into
a LAN hub), is the LAN emulation configuration server (LECS). It directs a
client to connect to a particular LAN emulation server (LES). The LES is the
component of the LANemulation service that performs the address registration and resolution. The LES is responsible for mapping IEEE 48-bit MAC
addresses and token-ring route descriptors to ATM addresses. One very
important MAC address for clients is the MAClayer broadcast address that
is used to send traffic to all locations on a LAN. In LAN emulation, this function is performed by the broadcast and unknown server (BUS). ATM LANE
is a comprehensive set of capabilities that has been widely deployed in ATM
networks.
ATM LANE is an element of the multiple protocol over ATM (MPOA) architecture that is being defined by the ATM Forum. This work is addressing
encapsulation of multiple protocols over ATM, automatic address resolution,
and the routing issues associated with minimizing multiple router hops in
ATM networks.

7. VOICE OVER ATM


As real-time voice services have traditionally been supported in the WAN via
circuit-based techniques (e.g., via T1 multiplexors or circuit switching), it is natural to map these circuits to ATM CBR PVCs using circuit emulation and ATM
adaptation Layer 1 (AAL1). However, there are significant disadvantages in
using circuit emulation in that the bandwidth must be dedicated for this type of
traffic (whether there is useful information being transmitted or not), providing
a disincentive for corporate users to implement circuit emulation as a long-term
strategy. For example a T1 1.544 Mbps circuit requires 1.74 Mbps of ATM
bandwidth when transmitted in circuit-emulation mode. This does not downplay its importance as a transitional strategy to address the installed base.
As technology has evolved, the inherent burstiness of voice and many realtime applications can be exploited (along with sophisticated compression
schemes) to significantly decrease the cost of transmission through the use of
VBRRT connections over ATM.

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The Basics of Telecommunications


VBR techniques for voice exploit the inherently bursty nature of voice communication, as there are silence periods that can result in increased efficiency.
These silence periods (in decreasing levels of importance) arise as follows:
when no call is up on a particular trunk; that is, the trunk is idle during offpeak hours; trunks are typically engineered for a certain call-blocking probability; at night, all the trunks could be idle
when the call is up, but only one person is talking at a given time
when the call is up, and no one is talking
Work is just starting in the ATM Forum on ATM adaptation for VBR voice.
The addition of more bandwidth-effective voice coding (e.g., standard voice
is coded using 64kbps PCM) is economically attractive, particularly over
long-haul circuits and T1 ATM interfaces. Various compression schemes have
been standardized in the industry (e.g., G720 series of standards). Making
these coding schemes dynamic provides the network operator with the
opportunity to free up bandwidth under network-congestion conditions. For
example, with the onset of congestion, increased levels of voice compression
could be dynamically invoked, thus freeing up bandwidth and potentially
alleviating the congestion while diminishing the quality of the voice during
these periods.
A further enhancement to the support of voice over ATM is to support voice
switching over SVCs. This entails interpreting PBX signaling and routing
voice calls to the appropriate destination PBX (see Figure 5). The advantage
from a traffic management perspective is that connection admission controls
can be applied to new voice calls; under network congestion conditions, these
calls could be rerouted over the public network and therefore not cause additional levels of congestion.
The ATM Forum is currently focusing its efforts on voice handled on CBR
SVCs. VBRRT voice is a future standards activity.

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Figure 5. Interpreting PBX Singaling and Routing Voice Calls

8. VIDEO OVER ATM


While circuit-based videoconferencing streams (including motion JPEG running at rates around 10 Mbps) can be handled by standard circuit emulation
using AAL1, the ATM Forum has specified the use of VBRRT VCs using
AAL5 for MPEG2 on ATM for video-on-demand applications, as this
approach makes better use of networking resources.
MPEG is a set of standards addressing coding of video and surround-sound
audio signals and synchronization of video and audio signals during the playback of MPEG data. It runs in the 2 Mbps to 15 Mbps range (with bursts
above these rates) corresponding to VCR and broadcast quality respectively.
The initial MPEG standard (MPEG1) was targeted at VHSquality video and
audio. MPEG2 targets applications requiring broadcast-quality video and
audio and HDTV. MPEG2 coding can result in one of two modes:
(i) Program streams: variable-length packets that carry a single program or
multiple programs with a common time base
(ii) Transport streams: 188-byte packets that contain multiple programs (for
examples, see Figure 6)

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The Basics of Telecommunications

Figure 6. Transport Streams with Multiple Programs


In both cases, time stamps are inserted into MPEG2 packets during the encoding and multiplexing process. MPEG2 assumes a constant-delay model across
the network, thus allowing the decoder to follow the original encoder source
clock exactly. Due to the cost of coding, MPEG2 is primarily used in a noninteractive broadcast mode, as would be the case for a point-to-multipoint
broadcast in residential video-on-demand applications and in a business TV
application for training or employee communications.

9. ATM TRAFFIC MANAGEMENT


Broadly speaking, the objectives of ATM traffic management are to deliver
quality-of-service guarantees for the multimedia applications and provide
overall optimization of network resources. Meeting these objectives enables
enhanced classes of service and offers the potential for service differentiation
and increased revenues while simplifying network operations and reducing
network cost.
ATM traffic management and its various functions can be categorized into
three distinct elements based on timing requirements:
First are nodal-level controls that operate in real time. These are implemented
in hardware and include queues supporting different loss and delay priorities,
fairly weighted queue-servicing algorithms, and rate controls that provide
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policing and traffic shaping. Well-designed switch-buffer architectures and
capacity are critical to effective network operation. Actual network experience and simulation have indicated that large, dynamically allocated output
buffers provide the flexibility to offer the best price performance for supporting various traffic types with guaranteed QoS. Dynamically managing buffer
space means that all shared buffer space is flexibly allocated to VCs on an asneeded basis. Additionally, per virtual connection (VC) queuing enables traffic
shaping and early and partial packet-level discard have been shown to significantly improve network performance.
Second, network-level controls operate in near real time. These are typically,
but not exclusively, implemented in software, including connection admission
control (CAC) for new connections, network routing and rerouting systems,
and flow-control-rate adaptation schemes. Network-level controls are the
heart of any traffic-management system. Connection admission controls support sophisticated equivalent-bandwidth algorithms with a high degree of
configuration flexibility, based on the cell rate for CBR VCs, on average cell
rate plus a configurable increment for VBR VCs, and on minimum cell rate for
ABR VCs. Dynamic class-of-service routing standards define support for fully
distributed link-state routing protocols, autoreconfiguration on failure and on
congestion, and dynamic load spreading on trunk groups.
Flow control involves adjusting the cell rate of the source in response to congestion conditions and requires the implementation of closed loop congestion
mechanisms. This does not apply to CBR traffic. For VBR and UBR traffic,
flow control is left as a CPE function. With ABR, resource management (RM)
cells are defined that allow signaling of the explicit rate to be used by traffic
sources. This is termed rate-based flow control. ABR is targeted at those
applications that do not have fixed or predictable bandwidth requirements
and require access to any spare bandwidth as quickly as possible while experiencing low cell loss. This allows network operators to maximize the bandwidth utilization of their network and sell spare capacity to users at a substantial discount while still providing QoS guarantees. To enhance the effectiveness of network-resource utilization, the ABR standard provides for endto-end, segment-by-segment, and hop-by-hop service adaptation.
Third, network engineering capabilities that operate in nonreal time support
data collection, configuration management, and planning tools (see Figure 7).

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The Basics of Telecommunications

Figure 7. Services Requested and Provided

10. ATM APPLICATIONS


ATM technologies, standards, and services are being applied in a wide range
of networking environments, as described briefly below (see Figure 8).

Figure 8. ATM Networking Environments


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ATM services: Service providers globally are introducing or already offering ATM services to their business users.
ATM workgroup and campus networks: Enterprise users are deploying
ATM campus networks based on the ATM LANE standards. Workgroup
ATM is more of a niche market with the wide acceptance of switchedEthernet desktop technologies.
ATM enterprise network consolidation: A new class of product has
evolved as an ATM multimedia network-consolidation vehicle. It is called
an ATM enterprise network switch. A full-featured ATM ENS offers a
broad range of in-building (e.g., voice, video, LAN, and ATM) and widearea interfaces (e.g., leased line, circuit switched, frame relay, and ATM at
narrowband and broadband speeds) and supports ATM switching, voice
networking, frame-relay SVCs, and integrated multiprotocol routing.
Multimedia virtual private networks and managed services: Service
providers are building on their ATM networks to offer a broad range of services. Examples include managed ATM, LAN, voice and video services
(these being provided on a per-application basis, typically including customer-located equipment and offered on an end-to-end basis) and full-service virtual private-networking capabilities (these include integrated multimedia access and network management).
Frame relay backbones: Frame-relay service providers are deploying ATM
backbones to meet the rapid growth of their frame-relay services to use as a
networking infrastructure for a range of data services and to enable frame
relay to ATM service interworking services.
Internet backbones: Internet service providers are likewise deploying
ATM backbones to meet the rapid growth of their frame-relay services, to
use as a networking infrastructure for a range of data services, and to enable
Internet class-of-service offerings and virtual private intranet services.
Residential broadband networks: ATM is the networking infrastructure
of choice for carriers establishing residential broadband services, driven by
the need for highly scalable solutions.
Carrier infrastructures for the telephone and private-line networks:
Some carriers have identified opportunities to make more effective use of
their SONET/SDH fiber infrastructures by building an ATM infrastructure
to carry their telephony and private-line traffic.
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The Basics of Telecommunications

11. NORTEL NETWORKS ATM VISION


Nortel Networks believes that ATM is the only viable backbone networking
technology that can meet the objective of making multimedia calls as easy,
reliable, and secure as voice calls are today.
ATM, coupled with SONET/SDH for fiber transport, sits at the core of
Nortels long-term architectural vision. That vision embraces various residential, business, and mobile access arrangements with a set of voice/data/video
and, ultimately, multimedia servers. There will be many ways of accessing
ATM networks including desktop ATM, switched Ethernet, wireless, and
xDSL, to name a few. The vision includes extensive support of multiple classes of service for native ATM, IPbased, frame-relay-based, and circuit-based
applications. ATM accommodates the inherently bursty nature of data, voice,
and video applications and the compressibility of these traffic types for
increased storage and bandwidth effectiveness. Nortel also believes that frame
relay and ATM, both being virtual circuit based, provide a service continuum
supporting the broadest sets of speeds from sub64 kbps all the way to Gbps.
Finally, Nortel envisages a family of application servers around the periphery
of this network to provide a range of data, image, video, and voice services
that take advantage of increasing insensitivity of the network to distance
(see Figure 9).

Figure 9. Nortels ATM Architectural Vision


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12. SELF-TEST
1.

Which is better suited to handle bursty traffic?


a. circuit switching
b. ATM

2.

Which organization is currently responsible for international signaling and interface standards?
a. ITU
b. CCITT

3.

In ATM networks all information is formatted into fixed-length cells consisting of


how many bytes?
a. 48 bytes
b. 64 bytes

4.

The basic connection unit in an ATM network is known as the


a. virtual channel connection
b. virtual path connection

5.

Which virtual circuit connection is the preferred mode of operation in an ATM network?
a. permanent virtual circuits
b. switched virtual circuits

6.

CBR is used for

a. multimedia e-mail
b. videoconferencing
7.

Which ATM networking standard supports the greater range of features?


a. PNNI
b. BICI

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The Basics of Telecommunications


8.

In ATM LAN emulation, most unicast LAN traffic moves directly between clients
over
.
a. servers
b. direct ATM SVCs

9.

The initial MPEG standard (MPEG1) was targeted at

a. VHSquality video and audio


b. broadcast-quality video and audio
10. ATM timing requirements feature how many elements?
a. 3
b. 4
c. 64k

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13. ACRONYM GUIDE


AAL1

ATM adaption Layer 1

AAL2

ATM adaption Layer 2

AAL3

ATM adaption Layer 3

AAL4

ATM adaption Layer 4

AAL5

ATM adaption Layer 5

ABR

available bit rate

ATM

asynchronous transfer mode

ATM UNI

ATM user network interface

BT

burst tolerance

CAC

connection admission control

CBR

constant bit rate

CCITT

Comit Consultif Internationale de Telegraphique and


Telephonique

CDV

cell-delay variation

CLR

cell-loss ratio

CTD

cell-transfer delay

IEEE

Institute of Electrical and Electronic Engineers

ITUTSS

International Telecommunications UnionTelecommunications


Standards Sector

LAN

local-area network

LANE

LAN emulation

LES

LAN emulation server

LUNI

LAN emulation user-network interface

MPOA

multiple protocol over ATM

PCR

peak cell rate

P-NNI

public network to network interface

PVC

permanent virtual circuit

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The Basics of Telecommunications


RM

resource management

SAD

speech activity detection

SCR

sustained cell rate

SDH/
SONET

synchronous digital
hierarchy/synchronous optical network

SVC

switched virtual circuit

TCP/IP

transmission control protocol/Internet protocol

UBR

unspecified bit rate

VBRNRT variable bit ratenonreal time


VCC

virtual-channel connections

VPC

virtual-path connections

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DEFINITION
Cable modems are devices that allow high-speed access to the Internet via a
cable television network. While similar in some respects to a traditional analog modem, a cable modem is significantly more powerful, capable of delivering data approximately 500 times faster.

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The Basics of Telecommunications

TOPICS
1.

HOW CABLE MODEMS WORK . . . . . . . . . . . . . . . . . . . . . . . . . . . . .43

2.

CABLE DATA SYSTEM FEATURES . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

3.

CABLE DATA NETWORK ARCHITECTURE . . . . . . . . . . . . . . . . . . . .46

4.

CABLE DATA NETWORK STANDARDS . . . . . . . . . . . . . . . . . . . . . . .49

5.

SUMMARY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52

6.

SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52

7.

ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .55

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1. HOW CABLE MODEMS WORK


Current Internet access via a 28.8. 33.6, or 56 kbps modem is referred to as
voiceband modem technology. Like voiceband modems, cable modems modulate and demodulate data signals. However, cable modems incorporate more
functionality suitable for todays high-speed Internet services. In a cable network, data from the network to the user is referred to as downstream, whereas data from the user to the network is referred to as upstream. From a user
perspective, a cable modem is a 64/256 QAM RF receiver capable of delivering up to 30 to 40 Mbps of data in one 6 MHz cable channel. This is approximately 500 times faster than a 56 kbps modem. Data from a user to the network is sent in a flexible and programmable system under control of the
headend. The data is modulated using a QPSK/16 QAM transmitter with data
rates from 320 kbps up to 10 Mbps. The upstream and downstream data rates
may be flexibly configured using cable modems to match subscriber needs.
For instance, a business service can be programmed to receive as well as
transmit higher bandwidth. A residential user, however, may be configured to
receive higher bandwidth access to the Internet while limited to low bandwidth transmission to the network.
A subscriber can continue to receive cable television service while simultaneously receiving data on cable modems to be delivered to a personal computer
(PC) with the help of a simple one-to-two splitter (see Figure 1). The data service offered by a cable modem may be shared by up to sixteen users in a
local area network (LAN) configuration.

Figure 1: Cable Modem at the Subscriber Location


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The Basics of Telecommunications


Because some cable networks are suited for broadcast television services,
cable modems may use either a standard telephone line or a QPSK/16 QAM
modem over a two-way cable system to transmit data upstream from a user
location to the network. When a telephone line is used in conjunction with a
one-way broadcast network, the cable data system is referred to as a telephony return interface (TRI) system. In this mode, a satellite or wireless cable
television network can also function as a data network.
At the cable headend, data from individual users is filtered by upstream
demodulators (or telephone-return systems, as appropriate) for further processing by a cable modem termination system (CMTS). A CMTS is a data
switching system specifically designed to route data from many cable modem
users over a multiplexed network interface. Likewise, a CMTS receives data
from the Internet and provides data switching necessary to route data to the
cable modem users. Data from the network to a user group is sent to a
64/256 QAM modulator. The result is user data modulated into one 6 MHz
channel, which is the spectrum allocated for a cable television channel such as
ABC, NBC, or TBS for broadcast to all users (see Figure 2).

Figure 2: Cable Modem Termination System and Cable Headend


Transmission
A cable headend combines the downstream data channels with the video,
pay-per-view, audio, and local advertiser programs that are received by television subscribers. The combined signal is then transmitted throughout the
cable distribution network. At the user location, the television signal is
received by a set top box, while user data is separately received by a cable
modem box and sent to a PC.

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A CMTS is an important new element for support of data services that integrates upstream and downstream communication over a cable data network. The number of upstream and downstream channels in a given CMTS
can be engineered based on serving area, number of users, data rates offered
to each user, and available spectrum.
Another important element in the operations and day-to-day management
of a cable data system is an element management system (EMS). An EMS is
an operations system designed specifically to configure and manage a
CMTS and associated cable modem subscribers. The operations tasks
include provisioning, day-to-day administration, monitoring, alarms, and
testing of various components of a CMTS. From a central network operations center (NOC), a single EMS can support many CMTS systems in the
geographic region.

Figure 3: Operations and Management of Cable Data Systems

2. CABLE DATA SYSTEM FEATURES


Beyond modulation and demodulation, a cable modem incorporates many
features necessary to extend broadband communications to wide area networks (WANs). The network layer is chosen as Internet protocol (IP) to support the Internet and World Wide Web services. The data link layer is comprised of three sublayers: logical link control sublayer, link security sublayer
conforming to the security requirements, and media access control (MAC)
sublayer suitable for cable system operations. Current cable modem systems
use Ethernet frame format for data transmission over upstream and downstream data channels. Each of the downstream data channels and the associ-

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The Basics of Telecommunications


ated upstream data channels on a cable network form an extended Ethernet
WAN. As the number of subscribers increases, a cable operator can add more
upstream and downstream data channels to support demand for additional
bandwidth in the cable data network. From this perspective, growth of new
cable data networks can be managed in much the same fashion as the growth
of Ethernet LANs within a corporate environment.
The link security sublayer requirements are further defined in three sets of
requirements: baseline privacy interface (BPI), security system interface (SSI),
and removable security module interface (RSMI). BPI provides cable modem
users with data privacy across the cable network by encrypting data traffic
between users cable modem and CMTS. The operational support provided
by the EMS allows a CMTS to map a cable modem identity to paying subscribers and thereby authorize subscriber access to data network services.
Thus, the privacy and security requirements protect user data as well as prevent theft of cable data services.
Early discussions in the Institute of Electrical and Electronic Engineers (IEEE)
802.14 Committee referred to the use of asynchronous transfer mode (ATM)
over cable data networks to facilitate multiple services including telephone,
data, and video, all of which are supported over cable modems. Although current cable modem standards incorporate Ethernet over cable modem, extensions are provided in the standards for future support of ATM or other protocol data units. IPtelephony support over cable data networks is expected to
be a new value-added service in the near term.

3. CABLE DATA NETWORK ARCHITECTURE


Cable data network architecture is similar to that of an office LAN. A CMTS
provides extended an Ethernet network over a WAN with a geographic reach
up to 100 miles. The cable data network may be fully managed by the local
cable operations unit. Alternatively, all operations may be aggregated at a
regional data center to realize economies of scale. A given geographic or metropolitan region may have a few cable television headend locations that are
connected together by fiber links. The day-to-day operations and management of a cable data network may be consolidated at a single location, such
as a super hub, while other headend locations may be economically managed
as basic hubs (see Figure 4).

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Figure 4: Basic Distribution Hub


A basic distribution hub is a minimal data network configuration that exists
within a cable television headend. A typical headend is equipped with satellite receivers, fiber connections to other regional headend locations, and
upstream RF receivers for the pay-per-view and data services. The minimal
data network configuration includes a CMTS system capable of upstream and
downstream data transport and an IP router to connect to the super hub location (see Figure 5).

Figure 5: Super Hub

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The Basics of Telecommunications


A super hub is a cable headend location with additional temperature-controlled facilities to house a variety of computer servers, which are necessary
to run cable data networks. The servers include file transfer, user authorization and accounting, log control (syslog), IP address assignment and administration (DHCP servers), DNS servers, and data over cable service interface
specifications (DOCSIS) control servers. In addition, a super hub may deploy
operations support and network management systems necessary for the television as well as data network operations.
User data from basic and super hub locations is received at a regional data
center for further aggregation and distribution through out the network (see
Figure 6). A super hub supports dynamic host configuration protocol (DHCP),
DNS (domain name server), and log control servers necessary for the cable
data network administration. A regional data center provides connectivity to
the Internet and the World Wide Web and contains the server farms necessary
to support Internet services. These servers include e-mail, web hosting, news,
chat, proxy, caching, and streaming media servers.

Figure 6: Regional Data Center


In addition to cable data networks, a regional data center may also support
dial-up modem services (e.g., 56kbps service) and business-to-business
Internet services. A network of switching, routers, and servers is employed at
the regional data center to aggregate dial-up, high-speed and business Internet
services.

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A super hub and a regional data center may be collocated and managed as a
single business entity. A super hub is managed by a cable television service
provider (TCI), while the regional data center is managed as a separate and
independent business (@home). In some regions, an existing Internet service
provider (ISP) may provide regional data center support for many basic and
super hub locations managed by independent cable data network providers.
A regional data center is connected to other regional data centers by a national backbone network (see Figure 7). In addition each regional data center is
also connected to the Internet and World Wide Web services. Traffic between
the regional networks, the Internet and all other regional networks is aggregated through the regional data center.

Figure 7: National Network

4. CABLE DATA NETWORK STANDARDS


A cable data system is comprised of many different technologies and standards. To develop a mass market for cable modems, products from different
vendors must be interoperable.
To accomplish the task of interoperable systems, the North American cable
television operators formed a limited partnership, Multimedia Cable Network
System (MCNS), and developed an initial set of cable modem requirements
(DOCSIS). MCNS was initially formed by Comcast, Cox, TCI, Time Warner,
Continental (now MediaOne), Rogers Cable, and CableLabs. The DOCSIS

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The Basics of Telecommunications


requirements are now managed by CableLabs. Vendor equipment compliance
to the DOCSIS requirements and interoperability tests are administered by a
CableLabs certification program.
For further details see http://www.cablemodem.com
Some of the details of cable modem requirements are listed below.

PHYSICAL LAYER
Downstream Data Channel: At the cable modem physical layer, downstream
data channel is based on North American digital video specifications (i.e.,
International Telecommunications Union [ITU]T Recommendation J.83
Annex B) and includes the following features:
64 and 256 QAM
6 MHzoccupied spectrum that coexists with other signals in cable plant
concatenation of Reed-Solomon block code and Trellis code, supports
operation in a higher percentage of the North American cable plants
variable length interleaving supports, both latency-sensitive and latencyinsensitive data services
contiguous serial bit-stream with no implied framing, provides complete
physical (PHY) and MAC layer decoupling
Upstream Data Channel: The upstream data channel is a shared channel
featuring the following:
QPSK and 16 QAM formats
multiple symbol rates
data rates from 320 kbps to 10 Mbps
flexible and programmable cable modem under control of CMTS
frequency agility
time-division multiple access
support of both fixed-frame and variable-length protocol data units

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programmable Reed-Solomon block coding
programmable preambles

MAC LAYER
The MAC layer provides the general requirements for many cable modem
subscribers to share a single upstream data channel for transmission to the
network. These requirements include collision detection and retransmission.
The large geographic reach of a cable data network poses special problems as
a result of the transmission delay between users close to headend versus users
at a distance from cable headend. To compensate for cable losses and delay as
a result of distance, the MAC layer performs ranging, by which each cable
modem can assess time delay in transmitting to the headend. The MAC layer
supports timing and synchronization, bandwidth allocation to cable modems
at the control of CMTS, error detection, handling and error recovery, and procedures for registering new cable modems.
Privacy: Privacy of user data is achieved by encrypting link-layer data between
cable modems and CMTS. Cable modems and CMTS headend controller
encrypt the payload data of link-layer frames transmitted on the cable network. A set of security parameters including keying data is assigned to a cable
modem by the Security Association (SA). All of the upstream transmissions
from a cable modem travel across a single upstream data channel and are
received by the CMTS. In the downstream data channel a CMTS must select
appropriate SA based on the destination address of the target cable modem.
Baseline privacy employs the data encryption standard (DES) block cipher for
encryption of user data. The encryption can be integrated directly within the
MAC hardware and software interface.

NETWORK LAYER
Cable data networks use IP for communication from cable modem to the network. The Internet Engineering Task Force (IETF) DHCP forms the basis for
all IP address assignment and administration in the cable network. A network
address translation (NAT) system may be used to map multiple computers
that use a single high-speed access via cable modem.

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The Basics of Telecommunications

TRANSPORT LAYER
Cable data networks support both transmission control protocol (TCP) and
user datagram protocol (UDP) at the transport layer.

APPLICATION LAYER
All of the Internet-related applications are supported here. These applications
include e-mail, ftp, tftp, http, news, chat, and signaling network management
protocol (SNMP). The use of SNMP provides for management of the CMTS
and cable data networks.

OPERATIONS SYSTEM
The operations support system interface (OSSI) requirements of DOCSIS
specify how a cable data network is managed. To date, the requirements
specify a RF MIB. This enables system vendors to develop an EMS to support
spectrum management, subscriber management, billing, and other operations.

5. SUMMARY
Cable modem technology offers high-speed access to the Internet and World
Wide Web services. Cable data networks integrate the elements necessary to
advance beyond modem technology and provide such measures as privacy,
security, data networking, Internet access, and quality of service features. The
end-to-end network architecture enables a user cable modem to connect to a
CMTS which, in turn, connects to a regional data center for access to Internet
services. Thus, through a system of network connections, a cable data network
is capable of connecting users to other users anywhere in the global network.

6. SELF-TEST QUESTIONS
1. In addition to their other functions, cable modems function as 64/256 QAM receivers
and QPSK/16 QAM transmitters.
a. true
b. false
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2. Cable modem users can modify the baseline privacy interface (BPI) to defeat data
encryption.
a. true
b. false

3. A CMTS provides an extended Ethernet network with a geographic reach of up to


______.
a. 1 mile
b. 10 miles
c. 100 miles
d. none of the above

4. The upstream data channel specification requirements feature which of the following?
a. 64 and 256 QAM formats
b.data rates from 320 kbps to 10 Mbps
c. 6 Mhzoccupied spectrum that coexists with other signals in the cable plant
d.all of the above

5. If a subscriber is receiving or sending data on a cable modem, regular cable television


service is momentarily interrupted.
a. true
b. false

6. Current cable modem standards have completely rejected the ATM protocols as a
data transmission method, as proposed by the IEEE 802-14 Committee.
a. true
b. false
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The Basics of Telecommunications


7. Cable modem operations can be managed by a local cable company or from a
remote location.
a. true
b. false

8. Among other functions, the media access control (MAC) layer provides control of
subscriber upstream transmissions such that no more than 25 percent of all such
transmissions can collide (thereby requiring retransmission).
a. true
b. false

9. As the number of subscribers increases in a cable data network, all of the users will
experience poor performance in their Internet access.
a. true
b. false

10. In a cable data network, Web television users can access all of their neighbors'
data on television.
a. true
b. false

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6. ACRONYM GUIDE
ATM

asynchronous transfer mode

BPI

baseline privacy interface

CMTS

cable modem termination system

DES

data encryption standard

DHCP

dynamic host configuration protocol

DNS

domain name server

DOCSIS

data over cable service interface specifications

EMS

element management system

IP

Internet protocol

ISP

Internet service provider

LAN

local-area network

MAC

media access control

MCNS

multimedia cable network system

NET

network address translation

NOC

network operations center

OSSI

operations support system interface

QAM

quadrature amplitude modulation

QPSK

quaternary phase shift keying

RF

radio frequency

SNMP

signaling network management protocol

SSI

security system interface

TCP

transmission control protocol

TRI

telephony return interface

UDP

user datagram protocol

WAN

wide-area network
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Cellular Communications
DEFINITION
A cellular mobile communications system uses a large number of low-power
wireless transmitters to create cellsthe basic geographic service area of a
wireless communications system. Variable power levels allow cells to be
sized according to the subscriber density and demand within a particular
region. As mobile users travel from cell to cell, their conversations are handed
off between cells to maintain seamless service. Channels (frequencies) used in
one cell can be reused in another cell some distance away. Cells can be added
to accommodate growth, creating new cells in unserved areas or overlaying
cells in existing areas.

TUTORIAL OVERVIEW
This tutorial discusses the basics of radio telephony systems, including both
analog and digital systems. Upon completion of this tutorial, you should be
able to accomplish the following:
1. describe the basic components of a cellular system
2. identify and describe digital wireless technologies

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TOPICS
1. MOBILE COMMUNICATIONS PRINCIPLES . . . . . . . . . . . . . . . . . . . . .59
2. MOBILE TELEPHONE SYSTEM USING THE CELLULAR CONCEPT . .60
3. CELLULAR SYSTEM ARCHITECTURE . . . . . . . . . . . . . . . . . . . . . . . . . .62
4. NORTH AMERICAN ANALOG CELLULAR SYSTEMS . . . . . . . . . . . . .66
5. CELLULAR SYSTEM COMPONENTS . . . . . . . . . . . . . . . . . . . . . . . . . . .68
6. DIGITAL SYSTEMS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .70
7. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .76
8. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .78

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1. MOBILE COMMUNICATIONS PRINCIPLES


Each mobile uses a separate, temporary radio channel to talk to the cell site.
The cell site talks to many mobiles at once, using one channel per mobile.
Channels use a pair of frequencies for communicationone for transmitting
from the cell site, the forward link, and one frequency for the cell site to
receive calls from the users, the reverse link. Radio energy dissipates over distance, so mobiles must stay near the base station to maintain communications. The basic structure of mobile networks includes telephone systems and
radio services. While a mobile radio service operates in a closed network and
has no access to telephone systems, mobile telephone service allows interconnection to a telephone network (see Figure 1).

Figure 1. Basic Mobile Telephone Service Network

EARLY MOBILE TELEPHONE SYSTEM ARCHITECTURE


Traditional mobile service was structured to be similar to television broadcasting: One powerful transmitter located at the highest spot in an area
would broadcast in a radius of up to 50 kilometers. The cellular concept structured the mobile telephone network in a different way. Instead of using one
powerful transmitter, many low-power transmitters were placed throughout a
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The Basics of Telecommunications


coverage area. For example, by dividing a metropolitan region into one hundred different areas (cells) with low-power transmitters using twelve conversations (channels) each, the system capacity theoretically could be increased
from twelve conversationsor voice channels using one powerful transmitterto twelve hundred conversations (channels) using one hundred lowpower transmitters. Figure 2 shows a metropolitan area configured as a traditional mobile telephone network with one high-power transmitter.

Figure 2. Early Mobile Telephone System Architecture

2. MOBILE TELEPHONE SYSTEM USING THE CELLULAR


CONCEPT
Interference problems caused by mobile units using the same channel in adjacent areas proved that all channels could not be reused in every cell. Areas
had to be skipped before the same channel could be reused. Even though this
affected the efficiency of the original concept, frequency reuse was still a
viable solution to the problems of mobile telephony systems.
Engineers discovered that the interference effects were not a result of the distance between areas but of the ratio of the distance between areas to the
transmitter power (radius) of the areas. By reducing the radius of an area by
50 percent, service providers could increase the number of potential customers in an area fourfold. Systems based on areas with a one-kilometer

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radius would have one hundred times more channels than systems with areas
ten kilometers in radius. Speculation led to the conclusion that by reducing
the radius to a few hundred meters, millions of calls could be served.
The cellular concept employs variable low-power levels, which allows cells to
be sized according to the subscriber density and demand of a given area. As
the population grows, cells can be added to accommodate that growth.
Frequencies used in one cell cluster can be reused in other cells. Conversations
can be handed off from cell to cell to maintain constant phone service as the
user moves between cells (see Figure 3).

Figure 3. Mobile Telephone System Using a Cellular Architecture


The cellular radio equipment (base station) can communicate with mobiles as
long as they are within range. Radio energy dissipates over distance, so the
mobiles must be within the operating range of the base station. Like the early
mobile radio system, the base station communicates with mobiles via a channel. The channel is made of two frequencies, one for transmitting to the base
station and one to receive information from the base station.

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3. CELLULAR SYSTEM ARCHITECTURE


Increases in demand and the poor quality of existing service led mobile service
providers to research ways to improve the quality of service and to support
more users in their systems. Because the amount of frequency spectrum available for mobile cellular use was limited, efficient use of the required frequencies was needed for better mobile cellular coverage. In modern cellular telephony, rural and urban regions are divided into areas according to specific provisioning guidelines. Deployment parameters, such as amount of cell-splitting
and cell sizes, are determined by engineers experienced in cellular system
architecture. Provisioning for each region is planned according to an engineering plan that includes cells, clusters, frequency reuse, and handoffs.

CELLS
A cell is the basic geographic unit of a cellular system. The term cellular comes
from the honeycomb shape of the areas into which a coverage region is
divided. Cells are base stations transmitting over small geographic areas that
are represented as hexagons. Each cell size varies depending on the landscape.
Because of constraints imposed by natural terrain and man-made structures,
the true shape of cells is not a perfect hexagon.

CLUSTERS
A cluster is a group of cells. No channels are reused within a cluster. Figure 4
illustrates a seven-cell cluster.

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Figure 4. A Seven-Cell Cluster

FREQUENCY REUSE
Because only a small number of radio channel frequencies were available for
mobile systems, engineers had to find a way to reuse radio channels to carry
more than one conversation at a time. The solution the industry adopted was
called frequency planning or frequency reuse. Frequency reuse was implemented by restructuring the mobile telephone system architecture into the
cellular concept.
The concept of frequency reuse is based on assigning to each cell a group of
radio channels used within a small geographic area. Cells are assigned a group
of channels that is completely different from neighboring cells. The coverage
area of cells is called the footprint. This footprint is limited by a boundary so
that the same group of channels can be used in different cells that are far
enough away from each other so that their frequencies do not interfere
(see Figure 5).

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The Basics of Telecommunications

Figure 5. Frequency Reuse


Cells with the same number have the same set of frequencies. Here, because
the number of available frequencies is 7, the frequency resue factor is 1/7.
That is, each cell is using 1/7 of available cellular channels.

CELL SPLITTING
Unfortunately, economic considerations made the concept of creating full systems with many small areas impractical. To overcome this difficulty, system
operators developed the idea of cell splitting. As a service area becomes full of
users, this approach is used to split a single area into smaller ones. In this
way, urban centers can be split into as many areas as necessary to provide
acceptable service levels in heavy-traffic regions, while larger, less expensive
cells can be used to cover remote rural regions (see Figure 6).

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Figure 6. Cell Splitting

HANDOFF
The final obstacle in the development of the cellular network involved the
problem created when a mobile subscriber traveled from one cell to another
during a call. Since adjacent areas do not use the same radio channels, a call
must either be dropped or transferred from one radio channel to another
when a user crosses the line between adjacent cells. Because dropping the call
is unacceptable, the process of handoff was created. Handoff occurs when the
mobile telephone network automatically transfers a call from radio channel to
radio channel as a mobile crosses adjacent cells (see Figure 7).

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The Basics of Telecommunications

Figure 7. Handoff between Adjacent Cells

During a call, two parties are on one voice channel. When the mobile unit
moves out of the coverage area of a given cell site, the reception becomes
weak. At this point, the cell site in use requests a handoff. The system
switches the call to a stronger frequency channel in a new site without interrupting the call or alerting the user. The call continues as long as the user is
talking, and the user does not notice the handoff at all.

4. NORTH AMERICAN ANALOG CELLULAR SYSTEMS


Originally devised in the late 1970s to early 1980s, analog systems have been
revised somewhat since that time and operate in the 800-MHz range. A group
of government, telco, and equipment manufacturers worked together as a
committee to develop a set of rules (protocols) that govern how cellular subscriber units (mobiles) can communicate with the cellular system. System
development takes into consideration many different and often opposing
requirements for the system, and often a compromise between conflicting
requirements results. Cellular development involves some basic topics:
frequency and channel assignments
type of radio modulation
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maximum power levels
modulation parameters
messaging protocols
callprocessing sequences

THE ADVANCED MOBILE PHONE SERVICE (AMPS)


AMPS was released in 1983 using the 800-MHz to 900-MHz frequency band
and the 30 kHz bandwidth for each channel as a fully automated mobile telephone service. It was the first standardized cellular service in the world and is
currently the most widely used standard for cellular communications. Designed
for use in cities, AMPS later expanded to rural areas. It maximized the cellular
concept of frequency reuse by reducing radio power output. The AMPS telephones (or handsets) have the familiar telephone-style user interface and are
compatible with any AMPS base station. This makes mobility between service
providers (roaming) simpler for subscribers. Limitations associated with AMPS
include the following:
low calling capacity
limited spectrum
no room for spectrum growth
poor data communications
minimal privacy
inadequate fraud protection
AMPS is used throughout the world and is particularly popular in the United
States, South America, China, and Australia. AMPS uses frequency modulation (FM) for radio transmission. In the United States, transmissions from
mobile to cell site use separate frequencies from the base station to the
mobile subscriber.

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The Basics of Telecommunications

NARROWBAND ANALOG MOBILE PHONE SERVICE


(NAMPS)
Since analog cellular was developed, systems have been implemented extensively throughout the world as first-generation cellular technology. In the second generation of analog cellular systems, NAMPS was designed to solve the
problem of low calling capacity. NAMPS is now operational in 35 U.S. and
overseas markets. NAMPS is a U.S. cellular radio system that combines existing voice processing with digital signaling, tripling the capacity of todays
AMPS systems. The NAMPS concept uses frequency division to get three
channels in the AMPS 30-kHz single-channel bandwidth. NAMPS provides
three users in an AMPS channel by dividing the 30-kHz AMPS bandwidth
into three 10-kHz channels. This increases the possibility of interference
because channel bandwidth is reduced.

5. CELLULAR SYSTEM COMPONENTS


The cellular system offers mobile and portable telephone stations the same
service provided fixed stations over conventional wired loops. It has the
capacity to serve tens of thousands of subscribers in a major metropolitan
area. The cellular communications system consists of the following four
major components that work together to provide mobile service to subscribers (see Figure 8):
public switched telephone network (PSTN)
mobile telephone switching office (MTSO)
cell site with antenna system
mobile subscriber unit (MSU)

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Figure 8. Cellular System Components

PSTN
The PSTN is made up of local networks, the exchange area networks, and the
long-haul network that interconnects telephones and other communication
devices on a worldwide basis. The mobile telephone switching office (MTSO)
is the central office for mobile switching. It houses the mobile switching center (MSC), field monitoring, and relay stations for switching calls from cell
sites to wireline central offices (PSTN). In analog cellular networks, the MSC
controls the system operation. The MSC controls calls, tracks billing information, and locates cellular subscribers.

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THE CELL SITE


The term cell site is used to refer to the physical location of radio equipment
that provides coverage within a cell. A list of hardware located at a cell site
includes power sources, interface equipment, radio frequency transmitters
and receivers, and antenna systems.

MOBILE SUBSCRIBER UNITS (MSUs)


The mobile subscriber unit consists of a control unit and a transceiver that
transmits and receives radio transmissions to and from a cell site. Three types
of MSUs are available:
the mobile telephone (typical transmit power is 4.0 watts)
the portable (typical transmit power is 0.6 watts)
the transportable (typical transmit power is 1.6 watts)
The mobile telephone is installed in the trunk of a car, and the handset is
installed in a location convenient to the driver. Portable and transportable
telephones are hand-held and can be used anywhere. The use of portable and
transportable telephones is limited to the charge life of the internal battery.

6. DIGITAL SYSTEMS
As demand for mobile telephone service increased, service providers found
that basic engineering assumptions borrowed from wireline (landline) networks did not hold true in mobile systems. While the average landline phone
call lasts at least ten minutes, mobile calls usually run ninety seconds.
Engineers who expected to assign fifty or more mobile phones to the same
radio channel found that by doing so they increased the probability that a
user would not get dial tonewhich is known as call-blocking probability. As
a consequence, the early systems quickly became saturated, and the quality of
service decreased rapidly. The critical problem was capacity. The general
characteristics of TDMA, GSM, PCS1900, and CDMA promise to significantly increase the efficiency of cellular telephone systems to allow a greater
number of simultaneous conversations. Figure 9 shows the components of a
typical digital cellular system.
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Cellular Communications

Interface
Microwave
Fiber Optic

Radios

Antenna

Microwave
Fiber Optic

Digital
Switch

Radio
Controller

Figure 9. Digital Cellular System


The advantages of digital cellular technologies over analog cellular networks
include increased capacity and security. Technology options such as TDMA
and CDMA offer more channels in the same analog cellular bandwidth and
encrypted voice and data. Because of the enormous amount of money that
service providers have invested in AMPS hardware and software, providers
look for a migration from AMPS to DAMPS by overlaying their existing networks with TDMA architectures.

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Analog

Digital

EIA553 (AMPS)

IS-54 (TDMA + AMPS)

824 MHz to 891 MHz

824 MHz to 891 MHz

30 kHz

30 kHz

21 CC/395 VC

21 CC/395 VC

3 or 6

40 to 50
conversations per cell

125 to 300
conversations per cell

continuous

time-shared bursts

constant phase

variable frequency

variable phase

mobile/base

mobile slaved to base

authority shared
cooperatively

poor

better/easily scrambled

poor

high

ESN plus optional


password (PIN)

ESN plus optional password (PIN)

Standard

Spectrum
Channel bandwidth
Channels
Conversations per
channel
Subscriber capacity

TX/RCV type
Carrier type
Constant frequency
Relationship

Privacy
Noise immunity
Fraud detection

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Cellular Communications

AMPS/DAMPS COMPARISON
TIME DIVISION MULTIPLE ACCESS (TDMA)
North American digital cellular (NADC) is called DAMPS and TDMA.
Because AMPS preceded digital cellular systems, DAMPS uses the same setup
protocols as analog AMPS. TDMA has the following characteristics:
IS54 standard specifies traffic on digital voice channels.
Initial implementation triples the calling capacity of AMPS systems.
Capacity improvements of 6 to 15 times that of AMPS are possible.
Uses many blocks of spectrum in 800 MHz and 1900 MHz.
All transmissions are digital.
TDMA/FDMA application 7. 3 callers per radio carrier (6 callers on half rate
later), providing three times the AMPS capacity.
TDMA is one of several technologies used in wireless communications.
TDMA provides each call with time slots so that several calls can occupy one
bandwidth. Each caller is assigned a specific time slot. In some cellular systems, digital packets of information are sent during each time slot and
reassembled by the receiving equipment into the original voice components.
TDMA uses the same frequency band and channel allocations as AMPS. Like
NAMPS, TDMA provides three to six times as many channels in the same
bandwidth as a single AMPS channel. Unlike NAMPS, digital systems have
the means to compress the spectrum used to transmit voice information by
compressing idle time and redundancy of normal speech. TDMA is the digital
standard and has 30kHz bandwidth. Using digital voice encoders, TDMA is
able to use up to six channels in the same bandwidth where AMPS uses one
channel.

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EXTENDED TIME DIVISION MULTIPLE ACCESS (ETDMA)


The extended TDMA (ETDMA) standard claims a capacity of fifteen times
that of analog cellular systems. This capacity is achieved by compressing
quiet time during conversations. ETDMA divides the finite number of cellular frequencies into more time slots than TDMA. This allows the system to
support more simultaneous cellular calls.

FIXED WIRELESS ACCESS (FWA)


Fixed wireless access (FWA) is a radio-based local exchange service in which
telephone service is provided by common carriers (see Figure 10). It is primarily a rural applicationthat is, it reduces the cost of conventional wireline.
FWA extends telephone service to rural areas by replacing a wireline local
loop with radio communications. Other labels for wireless access include
fixed loop, fixed radio access, wireless telephony, radio loop, fixed wireless,
radio access, and Ionica. FWA systems employ TDMA or CDMA access
technologies.

Figure 10. Fixed Wireless Access

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Cellular Communications

PERSONAL COMMUNICATIONS SERVICES (PCS)


The future of telecommunications includes personal communications services.
PCS at 1900 MHz (PCS1900) is the North American implementation of
DCS1800 (global system for mobile communications, or GSM). Trial networks were operational in the United States by 1993, and in 1994, the Federal
Communications Commission (FCC) began spectrum auctions. As of 1995,
the FCC auctioned commercial licenses. In the PCS frequency spectrum, the
operators authorized frequency block contains a definite number of channels.
The frequency plan assigns specific channels to specific cells, following a
reuse pattern which restarts with each Nth cell. The uplink and downlink
bands are paired mirror images. As with AMPS, a channel number implies
one uplink and one downlink frequency. For example, channel 512 = 1850.2
MHz uplink paired with 1930.2 MHz downlink.

CODE DIVISION MULTIPLE ACCESS (CDMA)


Code division multiple access (CDMA) is a digital air interface standard
claiming eight to fifteen times the capacity of analog. It employs a commercial adaptation of military spread-spectrum single-sideband technology. Based
on spread spectrum theory, it is essentially the same as wireline service; the
primary difference is that access to the local exchange carrier (LEC) is provided via wireless phone. Because users are isolated by code, they can share the
same carrier frequency, eliminating the frequency reuse problem encountered
in AMPS and DAMPS. Every CDMA cell site can use the same 1.25 MHz
band, so with respect to clusters, n = 1. This greatly simplifies frequency
planning in a fully CDMA environment.
CDMA is an interference-limited system. Unlike AMPS/TDMA, CDMA has a
soft capacity limit; however, each user is a noise source on the shared channel
and the noise contributed by users accumulates. This creates a practical limit
to how many users a system will sustain.
Mobiles that transmit excessive power increase interference to other mobiles.
For CDMA, precise power control of mobiles is critical in maximizing the
systems capacity and increasing battery life of the mobiles. The goal is to
keep each mobile at the absolute minimum power level that is necessary to
ensure acceptable service quality. Ideally, the power received at the base station from each mobile should be the same (minimum signal to interference).
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7. SELF-TEST
Multiple Choice
1. Interference effects in cellular systems are due to

a. the distance between areas


b. the power of the transmitters
c. the ratio of the distance between areas to the transmitter power of the
areas
d. the height of the antennas
2. Larger cells are more useful in

a. densely populated urban areas


b. rural areas
c. lightly populated urban areas
d. mountainous areas
3. The most widely used standard for cellular communications is

a. the advanced mobile phone service (AMPS)


b. the mobile subscriber unit (MSU)
c. the mobile telephone switching office (MTSO)
d. the mobile subscriber unit (MSU)
4. How many conversations per channel can digital cellular carry?
a. 1
b. 2
c. 3
d. 10
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5. Which of the following is NOT a limitation of AMPS?
a. low calling capacity
b. poor privacy protection
c. limited spectrum
d. wide coverage area

True or False
6. Digital cellular technologies offer increased capacity and security.
a. true
b. false
7. TDMA, a digital air interface standard, has twice the capacity of analog.
a. true
b. false
8. Cells are always hexagonal in shape.
a. true
b. false
9. Frequency reuse was maximized by increasing the size of cells.
a. true
b. false
10. Fixed wireless access is primarily a rural application.
a. true
b. false

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8. ACRONYM GUIDE
AMPS

advanced mobile phone service; another acronym for ana


log cellular radio

BTS

base transceiver station; used to transmit radio frequency


over the air interface

CDMA

code division multiple access; a form of digital cellular


phone service that is a spread spectrum technology that
assigns a code to all speech bits, sends scrambled transmis
sion of the encoded speech over the air, and reassembles the
speech to its original format

DAMPS

digital advanced mobile phone service; a term for digital


cellular radio in North America

DCS

digital cellular system

ESN

electronic serial number; an identity signal that is sent from


the mobile to the MSC during a brief registration transmission

ETDMA

extended TDMA; developed to provide fifteen times the


capacity of analog systems by compressing quiet time during
conversations

FCC

Federal Communications Commission; the government


agency responsible for regulating telecommunications in the
United States

FCCH

frequency control channel

FDMA

frequency division multiple access; used to separate multi


ple transmissions over a finite frequency allocation; refers to
the method of allocating a discrete amount of frequency
bandwidth to each user to permit many simultaneous conversations

FM

frequency modulation; a modulation technique in which the


carrier frequency is shifted by an amount proportional to the
value of the modulating signal

FRA

fixed radio access

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GSM

global system for mobile communications; standard digital


cellular phone service in Europe and Japan; to ensure interoperability between countries, standards address much of the
network wireless infrastructure, including radio interfaces,
switching, signaling, and intelligent networks

Hz
(hertz)

a measurement of electromagnetic energy, equivalent to


one wave or cycle per second

kHz
(kilohertz)

thousands of hertz

MHz
(Megahertz)

millions of hertz

MS

mobile station

MSSC

mobile services switching center; a switch that provides


services and coordination between mobile users in a network
and external networks.

MSU

mobile station unit; handset carried by the subscriber

MTSO

mobile telephone switching office; the central office for


the mobile switch, which houses the field monitoring and
relay stations for switching calls from cell sites to wireline
central offices (PSTN)

MTX

mobile telephone exchange

NADC

North American digital cellular (also called United States


digital cellular or USDC); a time division multiple access
(TDMA) system that provides three to six times the capacity
of AMPS

NAMPS

narrowband advanced mobile phone service; NAMPS was


introduced as an interim solution to capacity problems;
NAMPS provides three times the AMPS capacity to extend
the usefulness of analog systems

PCS

personal communications service; a lower-powered, higherfrequency competitive technology that incorporates wireline
and wireless networks and provides personalized features

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PSTN

public switched telephone network; a PSTN is made of


local networks, the exchange area networks, and the longhaul network that interconnects telephones and other communication devices on a worldwide basis

RF

radio frequency; electromagnetic waves operating between


10 kHz and 3 MHz propagated without guide (wire or cable)
in free space

SIM

subscriber identity module; a smart card, which is inserted


into a mobile phone to get it going

SNSE

super node size enhanced

TDMA

time division multiple access; used to separate multiple


conversation transmissions over a finite frequency allocation
of through-the-air bandwidth; used to allocate a discrete
amount of frequency bandwidth to each user; to permit
many simultaneous conversations, each caller is assigned a
specific time slot for transmission.

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Fiber-Optic Technology
DEFINITION
Fiber-optic communications is based on the principle that light in a glass
medium can carry more information over longer distances than electrical signals can carry in a copper or coaxial medium. The purity of today's glass
fiber, combined with improved system electronics, enables fiber to transmit
digitized light signals well beyond 100 km (60 miles) without amplification.
With few transmission losses, low interference, and high bandwidth potential, optical fiber is an almost ideal transmission medium.

TUTORIAL OVERVIEW
The advantages provided by optical fiber systems are the result of a continuous stream of product innovations and process improvements. As the requirements and emerging opportunities of optical fiber systems are better understood, fiber is improved to address them. This tutorial provides an extensive
overview of the history, construction, operation, and benefits of optical fiber,
with particular emphasis on outside vapor deposition (OVD) process.

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TOPICS
1. FROM THEORY TO PRACTICAL APPLICATION:
A QUICK HISTORY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
2. HOW FIBER WORKS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
3. OUTSIDE VAPOR DEPOSITION (OVD) PROCESS . . . . . . . . . . . . . . . . 87
4. OVD BENEFITS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
5. FIBER GEOMETRY: A KEY FACTOR IN SPLICING
AND SYSTEM PERFORMANCE. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
6. HOW TO CHOOSE OPTICAL FIBER. . . . . . . . . . . . . . . . . . . . . . . . . . . 94
7. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
8. ACRONYM GUIDE. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102

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1. FROM THEORY TO PRACTICAL APPLICATION: A QUICK


HISTORY
An important principle in physics became the theoretical foundation for optical fiber communications: light in a glass medium can carry more information
over longer distances than electrical signals can carry in a copper or coaxial
medium.
The first challenge undertaken by scientists was to develop a glass so pure
that one percent of the light would be retained at the end of one kilometer
(km), the existing unrepeatered transmission distance for copper-based telephone systems. In terms of attenuation, this one-percent of light retention
translated to 20 decibels per kilometer (dB/km) of glass material.
Glass researchers all over the world worked on the challenge in the 1960s, but
the breakthrough came in 1970, when Corning scientists Drs. Robert Maurer,
Donald Keck, and Peter Schultz created a fiber with a measured attenuation
of less than 20 dB per km. It was the purest glass ever made.
The three scientists' work is recognized as the discovery that led the way to
the commercialization of optical fiber technology. Since then, the technology
has advanced tremendously in terms of performance, quality, consistency, and
applications.
Working closely with customers has made it possible for scientists to understand what modifications are required, to improve the product accordingly
through design and manufacturing, and to develop industry-wide standards
for fiber.
The commitment to optical fiber technology has spanned more than 30 years
and continues today with the endeavor to determine how fiber is currently
used and how it can meet the challenges of future applications. As a result of
research and development efforts to improve fiber, a high level of glass purity
has been achieved. Today, fiber's optical performance is approaching the theoretical limits of silica-based glass materials. This purity, combined with
improved system electronics, enables fiber to transmit digitized light signals
well beyond 100 km (more than 60 miles) without amplification. When compared with early attenuation levels of 20 dB per km, today's achievable levels
of less than 0.35 dB per km at 1310 nanometers (nm) and 0.25 dB per km at
1550 nm, testify to the incredible drive for improvement.

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2. HOW FIBER WORKS


The operation of an optical fiber is based on the principle of total internal
reflection. Light reflects (bounces back) or refracts (alters its direction while
penetrating a different medium), depending on the angle at which it strikes a
surface.
One way of thinking about this concept is to envision a person looking at a
lake. By looking down at a steep angle, the person will see fish, rocks, vegetation, or whatever is below the surface of the water (in a somewhat distorted
location due to refraction), assuming that the water is relatively clear and
calm. However, by casting a glance farther out, thus making the angle of sight
less steep, the individual is likely to see a reflection of trees or other objects
on an opposite shore. Because air and water have different indices of refraction, the angle at which a person looks into or across the water influences the
image seen.
This principle is at the heart of how optical fiber works. Lightwaves are guided through the core of the optical fiber in much the same way that radio frequency (RF) signals are guided through coaxial cable. The lightwaves are
guided to the other end of the fiber by being reflected within the core.
Controlling the angle at which the light waves are transmitted makes it possible to control how efficiently they reach their destination. The composition
of the cladding glass relative to the core glass determines the fiber's ability to
reflect light. The difference in the index of refraction of the core and the
cladding causes most of the transmitted light to bounce off the cladding glass
and stay within the core. In this way, the fiber core acts as a waveguide for
the transmitted light.
The Design of Fiber
Core, Cladding, and Coating
An optical fiber consists of two different types of highly pure, solid glass,
composed to form the core and cladding. A protective acrylate coating (see
Figure 1) then surrounds the cladding. In most cases, the protective coating is
a dual layer composition.

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Figure 1. Core, Cladding, and Coating


A protective coating is applied to the glass fiber as the final step in the manufacturing process. This coating protects the glass from dust and scratches that
can affect fiber strength. This protective coating can be comprised of two layers: a soft inner layer that cushions the fiber and allows the coating to be
stripped from the glass mechanically and a harder outer layer that protects
the fiber during handling, particularly the cabling, installation, and termination processes.
Single-Mode and Multimode Fibers
There are two general categories of optical fiber: single-mode and multimode
(see Figure 2).

Figure 2. Single-Mode and Multimode Fibers


Multimode fiber was the first type of fiber to be commercialized. It has a
much larger core than single-mode fiber, allowing hundreds of modes of light
to propagate through the fiber simultaneously. Additionally, the larger core
diameter of multimode fiber facilitates the use of lower-cost optical transmitters (such as light emitting diodes [LEDs] or vertical cavity surface emitting
lasers [VCSELs]) and connectors.
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Single-mode fiber, on the other hand, has a much smaller core that allows
only one mode of light at a time to propagate through the core. While it
might appear that multimode fibers have higher capacity, in fact the opposite
is true. Single-mode fibers are designed to maintain spatial and spectral
integrity of each optical signal over longer distances, allowing more information to be transmitted.
Its tremendous information-carrying capacity and low intrinsic loss have
made single-mode fiber the ideal transmission medium for a multitude of
applications. Single-mode fiber is typically used for longer-distance and higher-bandwidth applications (see Figure 3). Multimode fiber is used primarily in
systems with short transmission distances (under 2 km), such as premises
communications, private data networks, and parallel optic applications.
Optical Fiber Sizes
The international standard for outer cladding diameter of most single-mode
optical fibers is 125 microns (m) for the glass and 245 m for the coating.
This standard is important because it ensures compatibility among connectors, splices, and tools used throughout the industry.
Standard single-mode fibers are manufactured with a small core size, approximately 8 to 10 m in diameter. Multimode fibers have core sizes of 50 to 62.5
m in diameter (see Figure 3).

Figure 3. Optical Fiber Sizes

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Fiber-Optic Technology

3. OUTSIDE VAPOR DEPOSITION (OVD) PROCESS


Basic OVD optical fiber manufacturing consists of three steps: laydown, consolidation, and draw.
Laydown
In the laydown step, a soot preform is made from ultrapure vapors as they
travel through a traversing burner and react in the flame to form fine soot
particles of silica and germania (see Figure 4).

Figure 4. OVD Laydown Process


The OVD process is distinguished by the method of depositing the soot.
These particles are deposited on the surface of a rotating target rod. The core
material is deposited first, followed by the pure silica cladding. As both core
and cladding raw materials are vapor-deposited, the entire preform becomes
totally synthetic and extremely pure.
Consolidation
When deposition is complete, the bait rod is removed from the center of the
porous preform, and the preform is placed into a consolidation furnace.
During the consolidation process, the water vapor is removed from the preform. This high-temperature consolidation step sinters the preform into a
solid, dense, and transparent glass.
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The Draw
The finished glass preform is then placed on a draw tower and drawn into
one continuous strand of glass fiber (see Figure 5).

Figure 5. Optical Fiber Drawing Process


First, the glass blank is lowered into the top of the draw furnace. The tip of
the blank is heated until a piece of molten glass, called a gob, begins to fall
from the blankmuch like hot taffy. As the glob falls it pulls behind it a thin
strand of glass, the beginning of an optical fiber.

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Fiber-Optic Technology
The gob is cut off, and the fine fiber strand is threaded into a computer-controlled tractor assembly and drawn. Then, as the diameter is monitored, the
assembly speeds up or slows down to precisely control the size of the fiber's
diameter.
The fiber progresses through a diameter sensor that measures the diameter
hundreds of times per second to ensure specified outside diameter. Next, the
primary and secondary coatings are applied and cured, using ultraviolet
lamps.
At the bottom of the draw, the fiber is wound on spools for further processing. Fiber from these spools is proof-tested to ensure the minimal proof-test
of each fiber and then measured for performance of relevant optical and geometrical parameters. Each fiber has a unique identification number that can
be traced to all relevant manufacturing data (including raw materials and
manufacturing equipment). Each fiber reel is then placed into protective shipping containers and prepared for shipment to customers worldwide.

4. OVD BENEFITS
Fiber produced using the OVD process is purely synthetic, exhibits enhanced
reliability, and allows for precise geometrical and optical consistency. The
OVD process produces a very consistent matched-clad fiber.
OVD fibers are made of a core and cladding glass, each with slightly different
compositions. The manufacturing process provides the relationship between
these two glasses. A matched-clad, single-mode fiber design allows for a consistent fiber (see Figure 6).

Figure 6. Index Profile of a Matched-Clad Fiber Design


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The OVD process produces well-controlled fiber profiles and geometry, both
of which lead to a more consistent fiber. Fiber-to-fiber consistency is especially important when fibers from different manufacturing periods are joined,
through splicing and connectorization, to form an optical system.
Depressed-Clad Fiber Profile
The inside vapor deposition (IVD) or modified chemical vapor deposition
(MCVD) process produces what is called depressed-clad fiber because of the
shape of its refractive index profile.

Figure 7. Index Profile of a Depressed-Clad Fiber Design


Depressed-clad fibers are made with two different cladding glasses that form
an inner and an outer cladding region. The inner cladding region adjacent to
the fiber core has an index of refraction that is lower than that of pure silica,
while the outer cladding has an index equal to that of pure silica. Hence, the
index of the glass adjacent to the core is depressed.
Questions of Strength
One common misconception about optical fiber is that it must be fragile
because it is made of glass. In fact, research, theoretical analysis, and practical
experience prove that the opposite is true. While traditional bulk glass is brittle, the ultra-pure glass of optical fibers exhibits both high tensile strength and
extreme durability.

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How strong is fiber? Figures like 600 or 800 thousand pounds per square inch
are often cited, far more than copper's capability of 100 pounds per square
inch. That figure refers to the ultimate tensile strength of fiber produced
today. This is fiber's real, rather than theoretical, strength is 2 million pounds
per square inch.
ABCs of Fiber Strength
The depth of inherent microscopic flaws on its surface determines the actual
strength of optical fiber. These microscopic flaws exist in any fiber. As in a
length of chain, the weakest link (or, in fiber's case, the deepest flaw) determines the ultimate strength of the entire length of the chain.
Many fiber manufacturers tensile-load, or proof-test, fiber after production.
This process eliminates proof-test size flaws and larger, thereby ensuring that
the flaws of most concern are removed.
Life Expectancy
Fiber is designed and manufactured to provide a lifetime of service, provided
it is cabled and installed according to recommended procedures. Life
expectancy can be extrapolated from many tests. These test results, along
with theoretical analysis, support the prediction of long service life.
Environmental issues are also important to consider when evaluating a fiber's
mechanical and reliability performance.
Bending Parameters
Optical fiber and cable are easy to install because it is lightweight, small in
size, and flexible. Nevertheless, precautions are needed to avoid tight bends,
which may cause loss of light or premature fiber failure.
Experience and testing show that bare fiber can be safely looped with bend
diameters as small as two inches, the recognized industry standard for minimum-bend diameter. Splice trays and other fiber-handling equipment, such as
racks, are designed to prevent fiber-installation errors such as this.

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5. FIBER GEOMETRY: A KEY FACTOR IN SPLICING AND


SYSTEM PERFORMANCE
As greater volumes of fiber in higher fiber-count cables are installed, system
engineers are becoming increasingly conscious of the impact of splicing on
their systems. Splice yields and system losses have a profound impact on the
quality of system performance and the cost of installation.
Glass geometry, the physical dimensions of an optical fiber, has been shown
to be a primary contributor to splice loss and splice yield in the field. Early
on, one company recognized the benefit provided by tightly controlled fiber
geometry and has steadily invested in continuous improvement in this area.
The manufacturing process helps engineers reduce systems costs and support
the industry's low maximum splice-loss requirement, typically at around 0.05
dB.
Fiber that exhibits tightly controlled geometry tolerances will not only be easier and faster to splice but will also reduce the need for testing by ensuring
predictable, high-quality splice performance. This is particularly true when
fibers are spliced by passive, mechanical, or fusion techniques for both single
fibers and fiber ribbons. In addition, tight geometry tolerances lead to the
additional benefit of flexibility in equipment choice.
The benefits of tighter geometry tolerances can be significant. In today's
fiber-intensive architectures, it is estimated that splicing and testing can
account for more than 30 percent of the total labor costs of system installation.
Fiber Geometry Parameters
The three fiber geometry parameters that have the greatest impact on splicing
performance include the following:
* cladding diameterthe outside diameter of the glass
* core/clad concentricity (or core-to-cladding offset)how well the core is
centered in the cladding glass region
* fiber curlthe amount of curvature over a fixed length of fiber

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These parameters are determined and controlled during the fiber-manufacturing process. As fiber is cut and spliced according to system needs, it is important to be able to count on consistent geometry along the entire length of the
fiber and between fibers and not to rely solely on measurements made.
Cladding Diameter
The cladding diameter tolerance controls the outer diameter of the fiber, with
tighter tolerances ensuring that fibers are almost exactly the same size.
During splicing, inconsistent cladding diameters can cause cores to misalign
where the fibers join, leading to higher splice losses.
The drawing process controls cladding diameter tolerance. Some manufacturers are able to control the tolerance of the cladding to a level of 125.01.0 m.
Once the cladding diameter tolerance is tightened to this level, core/clad concentricity becomes the single largest geometry contributor to splice loss.
Core/Clad Concentricity
Tighter core/clad concentricity tolerances help ensure that the fiber core is
centered in relation to the cladding. This reduces the chance of ending up
with cores that do not match up precisely when two fibers are spliced together. A core that is precisely centered in the fiber yields lower-loss splices more
often.
Core/clad concentricity is determined during the first stages of the manufacturing process, when the fiber design and resulting characteristics are created.
During these laydown and consolidation processes, the dopant chemicals that
make up the fiber must be deposited with precise control and symmetry to
maintain consistent core/clad concentricity performance throughout the entire
length of fiber.

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Fiber Curl
Fiber curl is the inherent curvature along a specific length of optical fiber that
is exhibited to some degree by all fibers. It is a result of thermal stresses that
occur during the manufacturing process. Therefore, these factors must be rigorously monitored and controlled during fiber manufacture. Tighter fiber-curl
tolerances reduce the possibility that fiber cores will be misaligned during
splicing, thereby impacting splice loss.
Some mass fusion splicers use fixed v-grooves for fiber alignment, where the
effect of fiber curl is most noticeable.

Figure 8. Cladding Diameter, Core/Clad Concentricity, and Fiber Curl

6. HOW TO CHOOSE OPTICAL FIBER


Single-Mode Fiber Performance Characteristics
The key optical performance parameters for single-mode fibers are attenuation, dispersion, and mode-field diameter.
Optical fiber performance parameters can vary significantly among fibers
from different manufacturers in ways that can affect your system's performance. It is important to understand how to specify the fiber that best meets
system requirements.

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Attenuation
Attenuation is the reduction of signal strength or light power over the length
of the light-carrying medium. Fiber attenuation is measured in decibels per
kilometer (dB/km).
Optical fiber offers superior performance over other transmission media
because it combines high bandwidth with low attenuation. This allows signals to be transmitted over longer distances while using fewer regenerators or
amplifiers, thus reducing cost and improving signal reliability.
Attenuation of an optical signal varies as a function of wavelength (see Figure
9). Attenuation is very low, as compared to other transmission media (i.e.,
copper, coaxial cable, etc.), with a typical value of 0.35 dB/km at 1300 nm.
Attenuation at 1550 nm is even lower with a typical value of 0.25 dB/km.
This gives an optical signal, transmitted through fiber, the ability to travel
more than 100 km without regeneration or amplification.
Attenuation is caused by several different factors, but scattering and absorption primarily cause it. The scattering of light from molecular level irregularities in the glass structure leads to the general shape of the attenuation curve
(see Figure 9). Further attenuation is caused by light absorbed by residual
materials, such as metals or water ions, within the fiber core and inner
cladding. It is these water ions that cause the water peak region on the
attenuation curve, typically around 1383 nm. The removal of water ions is of
particular interest to fiber manufacturers as this water peak region has a
broadening effect and contributes to attenuation loss for nearby wavelengths.
Light leakage due to bending, splices, connectors, or other outside forces are
other factors resulting in attenuation.

Figure 9. Typical Attenuation vs. Wavelength


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Dispersion
Dispersion is the time distortion of an optical signal that results from the
many discrete wavelength components traveling at different rates and typically result in pulse broadening (see Figure 10). In digital transmission, dispersion
limits the maximum data rate, the maximum distance, or the information-carrying capacity of a single-mode fiber link. In analog transmission, dispersion
can cause a waveform to become significantly distorted and can result in
unacceptable levels of composite second-order distortion (CSO).

Figure 10. Impact of Dispersion


Dispersion vs. Wavelength
Fiber dispersion varies with wavelength and is controlled by fiber design (see
Figure 11). The wavelength at which dispersion equals zero is called the zerodispersion wavelength (l 0). This is the wavelength at which fiber has its
maximum information-carrying capacity. For standard single-mode fibers, this
is in the region of 1310 nm. The units for dispersion are also shown in Figure

11.
Figure 11. Typical Dispersion vs. Wavelength Curve
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Chromatic dispersion consists of two kinds of dispersion. Material dispersion
refers to the pulse spreading caused by the specific composition of the glass.
Waveguide dispersion results from the light traveling in both the core and the
inner cladding glasses at the same time but at slightly different speeds. The
two types can be balanced to produce a wavelength of zero dispersion anywhere within the 1310 nm to 1650 nm operating window.
Transmission in the 1550 nm Window
Optical fibers also can be manufactured to have the zero dispersion wavelength in the 1550-nm region, which is also the point where silica-based
fibers have inherently minimal attenuation. These fibers are referred to as dispersion-shifted fibers and are used in long-distance applications with high bit
rates. For applications utilizing multiple wavelengths, it is undesirable to have
the zero dispersion point within the operating wavelength range and fibers
known as nonzero dispersion-shifted fiber (NZDSF) are most applicable.
NZDSF fibers with large effective areas are used to obtain greater capacity
transmission over longer distance than would be possible with standard single-mode fibers. These fibers are able to take advantage of the optical amplifier technology available in the 1530 to 1600+ nm operating window while
mitigating nonlinear effects that can be troublesome at higher power levels.
For applications such as the interconnection of headends, delivery of programming to remote node sites, high-speed communication networks, and
regional and metropolitan rings (used primarily for competitive access applications), NZDSF fiber can improve system reliability, increase capacity, and
lower system costs.
Mode-Field Diameter
Mode-field diameter (MFD) describes the size of the light-carrying portion of
the fiber. This region includes the fiber core as well as a small portion of the
surrounding cladding glass. MFD is an important parameter for determining a
fiber's resistance to bend-induced loss and can affect splice loss as well. MFD,
rather than core diameter, is the functional parameter that determines optical
performance when a fiber is coupled to a light source, connectorized, spliced,
or bent. It is a function of wavelength, core diameter, and the refractive-index
difference between the core and the cladding. These last two are fiber design
and manufacturing parameters.

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Cutoff Wavelength
Cutoff wavelength is the wavelength above which a single-mode fiber supports and propagates only one mode of light. An optical fiber that is singlemoded at a particular wavelength may have two or more modes at wavelengths lower than the cutoff wavelength.
The effective cutoff wavelength of a fiber is dependent on the length of fiber
and its deployment and the longer the fiber, the lower the effective cutoff
wavelength. Or the smaller the bend radius of a loop of the fiber is, the lower
the effective cutoff wavelength will be. If a fiber is bent in a loop, the cutoff
is lowered.
Environmental Performance
While cable design and construction play a key role in environmental performance, optimum system performance requires the user to specify fiber that
will operate without undue loss from microbending.
Microbends are small-scale perturbations along the fiber axis, the amplitude
of which are on the order of microns. These distortions can cause light to leak
out of a fiber. Microbending may be induced at very cold temperatures
because the glass has a different coefficient of thermal expansion from the
coating and cabling materials. At low temperatures, the coating and cable
become more rigid and may contract more than the glass. Consequently,
enough load may be exerted on the glass to cause microbends. Coating and
cabling materials are selected by manufacturers to minimize loss due to
microbending.
Specification Examples of Uncabled Fiber
To ensure that a cabled fiber provides the best performance for a specific
application, it is important to work with an optical fibercable supplier to
specify the fiber parameters just reviewed as well as the geometric characteristics that provide the consistency necessary for acceptable splicing and connectorizing.

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Splicers and Connectors
As optical fiber moves closer to the customer, where cable lengths are shorter
and cables have higher fiber counts, the need for joining fibers becomes
greater. Splicing and connectorizing play a critical role both in the cost of
installation and in system performance.
The object of splicing and connectorizing is to precisely match the core of
one optical fiber with that of another in order to produce a smooth junction
through which light signals can continue without alteration or interruption.
There are two ways that fibers are joined:
* splices, which form permanent connections between fibers in the system
* connectors, which provide remateable connections, typically at termination
points
Fusion Splicing
Fusion splicing provides a fast, reliable, low-loss, fiber-to-fiber connection by
creating a homogenous joint between the two fiber ends. The fibers are melted or fused together by heating the fiber ends, typically using an electric arc.
Fusion splices provide a high-quality joint with the lowest loss (in the range
of 0.01 dB to 0.10 dB for single-mode fibers) and are practically nonreflective.
Mechanical Splicing
Mechanical splicing is an alternative method of making a permanent connection between fibers. In the past, the disadvantages of mechanical splicing
have been slightly higher losses, less-reliable performance, and a cost associated with each splice. However, advances in the technology have significantly
improved performance. System operators typically use mechanical splicing for
emergency restoration because it is fast, inexpensive, and easy. (Mechanical
splice losses typically range from 0.050.2 dB for single-mode fiber.)
Connectors
Connectors are used in applications where flexibility is required in routing an
optical signal from lasers to receivers, wherever reconfiguration is necessary,
and in terminating cables. These remateable connections simplify system
reconfigurations to meet changing customer requirements.

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7. SELF-TEST
1. A physics principle that became the theoretical foundation of optical fiber communications holds that ___________ in a __________ medium can carry more information
over longer distances.
a. light; coaxial
b. electrical signals; glass
c. light; glass
d. electrical signals; copper

2. Single-mode fiber was the first type of fiber to be commercialized.


a. true
b. false

3. What are the three primary steps in the optical fiber manufacturing process?
a. laydown, compilation, and draw
b. assembly, consolidation, and pull
c. laydown, consolidation, and draw
d. assembly, compilation, and pull

4. Which of the following is not a characteristic of the fiber produced by the OVD
process?
a. less attenuation
b. infinite flexibility
c. exceptional strength
d. fewer flaws
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5. Fiber curl is exhibited by all fibers.
a. true
b. false
6. Attenuation is which of the following?
a. the inherent curvature along a specific length of optical fiber
b. the wavelength above which a single-mode fiber supports only
one mode or ray of light
c. the reduction of signal strength over the length of the
light-carrying medium
d. smearing an optical signal that results from the many discrete
wavelength components traveling at different rates
7. Controlling the ____________ at which light waves are transmitted makes it possible to control how efficiently they reach their destination.
a. speed
b. angle
c. time
d. rate
8. Dispersion is which of the following?
a. the inherent curvature along a specific length of optical fiber
b. the wavelength above which a single-mode fiber supports only
one mode or ray of light
c. the reduction of signal strength over the length of the
light-carrying medium
d. smearing an optical signal that results from the many discrete
wavelength components traveling at different rates

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9. The longer the fiber is, the shorter the effective cutoff wavelength will be.
a. true
b. false

10. Mechanical splicing is the predominant choice of operators for joining fibers.
a. true
b. false

8. ACRONYM GUIDE
CNR

carrier-to-noise ratio

CSO

composite second-order distortion

IVD

inside vapor deposition

MCVD modified chemical vapor deposition


MFD

mode-field diameter

NZDSF nonzero dispersion-shifted fiber


OVD

outside vapor deposition

RF

radio frequency

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Telecommunications
TUTORIAL OVERVIEW
Sometimes, when attending a class, it is okay to miss the first half-hour. After
all, we do know something about the subject, and that first half-hour is likely to
carry little more than introductory information. The same holds true when picking up a textbook on a particular subject. Maybe the first chapter can be skipped;
again, it contains introductory material that we already know.
But often this is not the case. We do not know as much as we thought we
knew, and that first half-hour, or that first chapter, contains material that may
well be prerequisite material. That is what this Web ProForum tutorial is all
about. It provides the equivalent of that first half-hour, or that first chapter, and
may well be valuable in understanding the other tutorials in this series. The tutorial will cover the fundamentals of telephony, from its inception in Alexander
Graham Bells laboratory to todays emerging technologies.

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TOPICS
1. HISTORY AND REGULATION OF THE TELEPHONE INDUSTRY......105
2. NETWORK..................................................................................................108
3. SWITCHING TECHNOLOGY ..................................................................112
4. TRANSMISSION MEDIA ..........................................................................115
5. TRANSMISSION TECHNOLOGY............................................................118
6. BROADBAND ACCESS AND SERVICE...................................................120
7. WIRELESS....................................................................................................121
8. SELF-TEST ...................................................................................................125
9. ACRONYM GUIDE....................................................................................127

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1. HISTORY AND REGULATION OF THE TELEPHONE INDUSTRY


Mr. Watson, come here, I want you. With these historic words Alexander
Graham Bell called to his assistant Thomas Augustus Watson over the socalled telephone, and an industry was born.
The place: 5 Exeter Place, Boston, Massachusetts
The time: evening, March 10, 1876
As with all inventions, the road had not been smooth. For years, Graham Bell
(as he liked to be called) had been experimenting with a harmonic telegraph. It should be possible, he reasoned, to send six tones over the same
wire at the same time and cause six reeds attached to the receiving end to be
operated. Furthermore, if all worked well, varied combinations of these six
pitches could reproduce human speech.
Simultaneously, he was working on a scheme that utilized the varying resistance of a wire. A diaphragm, which would be vibrated by the human voice,
was attached to a wire that was dipped into a mixture of acid and water. In
theory, as the diaphragm moved downward, forcing more wire into the acid,
the resistance of the wire would be decreased. As the diaphragm moved
upward, the wire would be withdrawn from the conducting liquid, and its
resistance would be increased. It was this device that was ultimately successful and that formed the basis for the telephone industry for many years.
A year later, on July 9, 1877, the Bell Telephone Company was formed, and
Alexander Graham Bell became the companys electrician, at a salary of
$3,000, and Watson became superintendent in charge of research and manufacturing. Unfortunately for Bell, the basic patents were due to run out in
1893 and 1894. But by this time Theodore Newton Vail had been brought in
as general manager, and he immediately set about establishing an organization strong enough to survive without a monopoly. What we wanted to do
was get possession of the field in such a way that, patent or no patent, we
could control it, Vail said. The first step was to obtain a captive manufacturing facility, and this was accomplished in 1881 with the purchase of Western
Electric Company.
Vail also sent his salesmen into the field to set up telephone exchanges in virgin territory. Generally, local promoters were encouraged to organize a local
telephone company and sell stock. Thus, by 1885, Vail had established a vertically integrated supply division, a network of companies licensed by the
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parent, and a strong research and development arm. The expiration of Bells
basic patents in 1893 and 1894 was the starting signal for open competition.
Independent telephone operating companies sprang up throughout the country; by the turn of the century there were approximately 6,000 of them, and
these 6,000 provided service to some 600,000 subscribers. Through the years,
mergers and acquisitions took their toll; at the present time there are approximately 1,300 local exchange carriers.
Unfortunately for the general public, all these telephones were not interconnected. Therefore it was necessary for a subscriber to have two or three
instruments in order to communicate with the total population of the city.
But the great asset of AT&T, which became the official name of the company
at the end of 1899, was the control of all the long-distance circuits and its
steadfast refusal to interconnect any other company to it.
This would never do, and the Justice Department filed suit in 1912. The
world was angry with AT&T, and an AT&T vice presidentNathan C.
Kingsburyrealized it. He recognized that the best demonstration of AT&T
not being in a monopoly position was to point to thousands of independents
apparently operating in harmony. To this end, AT&T agreed to provide interconnection arrangements to all independents. This 1913 agreement was
henceforth called the Kingsbury Commitment.
By 1934 telecommunications had become so important to the country that
Congress passed a Communications Act and, simultaneously, created the
Federal Communications Commission (FCC). The section of this Act that has
turned out to be most important has to do with what we now call universal
service. It said: For the purpose of regulating interstate and foreign commerce in communication by wire and radio so as to make available, so far as
possible, to all the people of the United States a rapid, efficient, nationwide,
and worldwide wire and radio communication service with adequate facilities
at reasonable charges.
As a result of this principle, a support structure has been established whereby
certain groups of subscribers (e.g., long-distance users, business subscribers,
subscribers in locations where telephone service can be provided with relative
ease, etc.) will pay more than true costs, and other groups of subscribers (e.g.,
subscribers in rural and other high-cost locations) will pay less than true costs.
In 1949 the Justice Department again filed suit against AT&T, claiming that
Western Electric charged inordinately high prices from their customers (i.e.,
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the operating telephone companies owned by AT&T), thus making it possible
for the operating telephone companies to charge their subscribers inappropriately high rates. The suit dragged on, and a consent decree was reached in
1956. AT&T won; Western Electric need not be divested from AT&T, the Bell
System would engage only in telecommunications business, and nonexclusive
licenses would be granted to any applicant on fair terms. This was the final
judgment. The eventual breakup of the Bell System in 1984 was accomplished through a modification of this final judgment, hence the
Modification of Final Judgment (MFJ).
Although the Bell System appeared to be the winner in this 1956 suit, over the
next two decades they would lose battles, one at a time. There was the HushA-Phone case in 1955; the Carterfone case in 1968; MCIs above 890 case in
1959; and the MCI case dealing with a long-distance route from Chicago to St.
Louis in 1969. In November, 1974, the Justice Department once again filed suit
to break up the Bell System. The case trudged on until 1978, when Judge
Harold Greene took over. He moved things quickly, and on January 4, 1982, a
terse announcement was issued by the Justice Department and AT&T saying
that negotiations had been reopened. Then, on January 8, 1982, the news
broke; AT&T had agreed to break up its $136.8 billion empire. It was agreed
that AT&T would divest the local parts of the Bell operating telephone companies. It would keep its manufacturing facilities, and its long-distance network.
The agreement would take effect on January 1, 1984.
The twenty-two regional Bell operating companies (RBOCs) agreed to form
seven regional holding companies (Bell Atlantic, NYNEX, BellSouth,
Ameritech, U S West, Pacific Telesis, and Southwestern Bell). The agreement
also said that the Bell operating companies would not be allowed to manufacture nor would they be allowed to get in the long-distance business within
their territories. AT&T would not be allowed to get in the local-exchange
business nor to acquire the stock or assets of any RBOC.
That remained the state of affairs until the passage of the 1996 Telecommunications
Act. This Act threw most of the rules established in 1984 out the window and left
the implementation of the Act to the FCC. There have been problems ever since.
What did the Congress mean by promote competition? Should AT&T be
allowed to get in the local-exchange business? (Answer: yes.) Should the
RBOCs be allowed to get in the long-distance business? (Answer: yes, but
only after passing a fourteen-point checklist.) What did expanded universal
service mean? Should the RBOCs be allowed to merge? (Answer: yes. Bell
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Atlantic has merged with NYNEX; Southwestern Bell [SBC] has merged with
Pacific Telesis and is planning to merge with Ameritech; Bell Atlantic intends
to merge with GTE. If all of these are ultimately approved, there will remain
four RBOCs). To date, many questions remain, and there is no assurance that
they will be answered in the foreseeable future.

2. NETWORK
If there were only three or four telephones in a locale, it would make sense
to connect each phone to all other phones and find a simple method of
selecting the desired one. However, if there are three or four thousand
phones in a locale, such a method is out of the question. Then it is appropriate to connect each phone to some centrally located office and perform
switching there. This switching could be a simple manual operation using
plugs and sockets or could be done with electromechanical devices or with
electronics. In any case, this central office solution is the one that has been
chosen by the telecommunications industry.
As we connect each of these thousands of telephones to the central office, we
have what is called a star configuration; all lines are particular to one and only
one station, and all terminate on the nucleus of this starthe central office (CO).
These connections are called the local exchange plant, and the telephone
company handling this function is called the local exchange carrier (LEC).
The connections themselves are often called the local loop; at other times
we refer to them as the last mile. In more technical terms, the section closest to the customers premises is called the distribution plant and that section
closest to the central office, the feeder plant (see Figure 1).

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Network Interface Device

Figure 1: Particular Names Are Applied to the Various Parts of the


PSTN; End Offices Are Class-5 Offices; Toll Tandem Offices Are
Generally Class-4 Offices
(Note: This is certainly a generalization, as will be much that follows.
Although the distribution plant usually consists of one or more cables leading to some point of demarcation [a terminal box or an enclosure] after
which the lines are spread out going in many smaller cables to the customer
premises [the feeder plant], there are cases where there is no need for a point
of demarcation. Then what do we call the plant? We will not struggle with
such semantic difficulties here.)
But what if a particular telephone call is not originated and terminated within the particular central offices geographic coverage? How do we get to
another city, another state, or even another country?
The answer, of course, is to connect these central offices to a higher echelon
central office (see Figure 2). We apply numbers to these levels of offices; the
local office, also called the end office, is called a Class-5 office. The office to
which it connects is called the Class-4 office. And so on, with the top level,
the Class-1 office, appearing in only a few places in the country. Please note
that the only office that has consumers as its subscribers is the Class-5
office. The other offices in this hierarchy have lower-level central offices as
their subscribers. Those lines connecting switching offices to switching
offices, rather than to subscribers, are called trunks.
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Figure 2: The Hierarchy of Switching Systems in Its Most Basic Form


Consists of Five Classes of Offices
This section of the telephone infrastructurethe section leading upward
from the Class-5 officesis handled not by the LECs, but by the interexchange carriers (IXCs), the long-distance carriers. This entire structure has
been titled the hierarchy of switching systems. The total network is called
the public switched telephone network (PSTN).
In days of old, there was only one long-distance carrierAT&T. Hence any
time a telephone number was dialed with an area code up-front, the LEC
knew that it must be handed off to AT&T. But then came MCI, Sprint, and
hundreds of other long-distance carriers. What was an LEC to do with a particular long-distance call? To whom should it be handed off? This was and is
a technical challenge. In political terms it was called equal access, which
means that a requesting long-distance carrier could require that the LEC
examine the number and handoff the call to the proper long-distance carrier.
This handoff was from the CO of the LEC to the point of presence (PoP) of
the IXC. This PoP could be in a building adjacent to the telcos CO, or it
could be in some convenient site in the suburbs where it could serve several
of the telcos COs. The very pure hierarchy of switching systems was
becoming somewhat corrupted; new hierarchies in the long-distance part of
the network were being applied on top of the old one.
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Although it is not pertinent to the topology of this network, it should be recognized that the interconnections among these various COs can be twisted
copper pair carrier systems utilizing copper pairs (e.g., T1), microwave, satellites, and certainly fiber.
But this hierarchical network is not the only network in the telephone
system of today. There are many others:
A local area network (LAN) is a limited-distance network connecting a
defined set of terminals. It could connect workstations in an office,
offices in a building, or buildings on a campus.
A wide area network (WAN) links metropolitan or local networks,
usually over common carrier facilities.
The intelligent network is a concept that centralizes a significant amount
of intelligence rather than installing this intelligence in individual COs.
For instance, how does a particular CO know which long-distance carrier
is to receive a particular call?
The synchronous optical network (SONET) is a particular set of standards
that allows the interworking of products from different vendors. It usually
embodies a fiber-optic ring that will permit transmission in both directions.
The Internet is really quite different from the network we have been
describing. It is a packet network (rather than a circuit-switched network), but, as has been discussed, it is an overlay network.
The common channel signaling network is especially important; it
works closely with the PSTN. We also apply the term out-of-band signaling. In the original PSTN, signaling (e.g., call setup) and talking utilized the same common trunk from the originating switching system to
the terminating switching system. This process seized the trunks in all of
the switching systems involved. Hence if the terminating end was busy,
all of the trunks were set up unnecessarily. In the mid-1970s, the common channel signaling network was established; it utilizes the protocol
called signaling system 7 (SS7). With this system, a talking path was not
assigned until all signaling had been satisfactorily completed. This network, incidentally, was and is a packet network rather than a circuitswitched network.

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The PSTN we have been describing utilizes a star configuration. But this is
not the only configuration being applied in todays telecommunications
world. The cable television (CATV) companies, for instance, use a tree-andbranch technology. In this case the headend (equivalent to the CO) receives
programming from satellites and sends all signals downstream, out on the
trunk. At various points along the way, branches extend outward, toward
various neighborhoods. These branches are split several more times before
the coaxial cable (the media of choice in past CATV systems) reaches the
customers premises. Frequently the signals must be amplified along the way,
and therefore power must be sent along with the TV signal. In any case,
because the intent of the CATV system is broadcastthat is, to send the signal to everyonethere is no need to send an individual and distinct wire to
each and every subscriber, as was the case with the telephone system.
However, this methodology has proven to be disadvantageous to the CATV
companies because it is extremely difficult to send signals upstream. Of
course, in a telephone system, signals (voices) must be sent in both directions.
CATV companies are spending billions of dollars to upgrade their systems, not
only by utilizing fiber instead of the coax, but by adding electronics to the
many nodes that permit both upstream and downstream transmission.

3. SWITCHING TECHNOLOGY
The PSTN we have been describing has a star configuration. Local loops (usually one per subscriber) terminate in a CO. This CO completes connections
from one local loop to another local loop, or from one local loop to a trunk
that terminates on some other CO. This central office has gone through a
number of fundamental technological changes (see Table 1).

Table 1: Types of End-Office Switching and Their Evolution


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The manual system required, of course, constant attention from operators (see
Figure 3). In the late 1800s, telephone calls were connected manually at the CO.
When a call came in, an attendant would plug into a horizontal bar line. He
then would yell to the operator who handled the customer being called, and
that second operator would connect to the bar and finish setting up the call.
When the call was completed, another operator would yell to all in the room
that the line was clear again. The step-by-step system, which is still in operation in many parts of the country, utilized what is known as the Strowger
switch. The intelligence in the system was located in relays mounted on each
switch. The switch itself responded to the dial pulses of the rotary dial.

Figure 3: A Depiction of an Early CO


The crossbar system was still electromechanical in nature, but the intelligence
of the system was separated from the actual switch. Thus, this common control could be used repeatedly to set up and tear down calls and never sit idle.
When electronics came along, the electromechanical control of the common
control system was replaced with electronics, and the network, or matrix, was
usually replaced with tiny glass-encapsulated reed switches. Hence, only a part
of the switch was electronic. In the next generation, the stored program operation of a digital computer was applied to the switch, although the network
remained a complex of reed switches. In the final generation, called a digital
switch, the talking path was no longer an electrically continuous circuit; rather
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the speech being carried was digitized into a stream of 1s and 0s. Notice
that this final generation depicted a significant change from the previous generations in that there was no longer an electrical talking path through the
switch. We were, in fact, operating in a digital (rather than analog) domain.
However, whether the system was analog or digital, one thing must be recognized: There was an actual talking patha circuitfrom the calling party to
the called party. This talking path was established at the beginning of a call
and held for the duration of a call. We call it circuit switching. This system is
not very efficient, really. When I am talking, you are listening, and the circuit
is being used in only one directionthat is, 50 percent. When you are talking
and I am listening, it is still 50 percent. When neither of us is talking, or when
there is silence between words, the efficiency is 0 percent.
There is, however, a different kind of connection, and we see it today in a
number of applications:
credit-card verification
automated teller machine
SS7
Internet and the World Wide Web
This system is called packet switching (as opposed to circuit switching). In a
packet switching system, the information being transmitted (be it data or digitized voice) is not sent in real time over a dedicated circuit; rather it is stored
in a nearby computer until a sufficiently sized packet is on hand. Then a very
smart computer seizes a channel heading in the general direction of the destination, and that packet of data is transmitted at very high speeds. Then the
channel is released. So, except for some necessary supervisory information
(destination, error checking codes, etc.) the channel is 100 percent efficient.
When the distant station gets that message no more than a few milliseconds
later, it responds with the necessary handshaking informationagain, by
accumulating a packet of data, seizing a channel, and bursting the information
out over that channel. Again, 100 percent efficient.
As mentioned earlier, the packet networks in the world (actually overlay networks to the PSTN) are being used extensively for data; only recently are we
seeing them being used for voice. As systems are perfected, this also will change.

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4. TRANSMISSION MEDIA
There are four types of media that can be used in transmitting information in
the telecommunications world:
copper wire
coaxial cable (actually an adaptation of copper wire)
fiber
wireless
In days of old, copper wire was the only means of transmitting information.
Technically known as unshielded twisted pair (UTP), this consisted of a large
number of pairs of copper wire of varying size in a cable. The cable did not
have a shield and therefore the signalprimarily the high-frequency part of
the signalwas able to leak out. Also, the twisting on the copper pair was
very casual, designed as much to identify which wires belonged to a pair as
to handle transmission problems. However, this is the way it was done, and
for voice communications it was quite satisfactory. Consequently, there are
millions of miles of copper in the PSTNmiles that must be used.
Not only did the copper cable itself have limitations, but things were done to
this cable to make it even more unsuitable for high-speed data transmission.
These actions primarily took two forms:
Loading
Load coils were frequently added to loops longer than 18,000 feet. These load
coils were essentially low-pass filters. That is, they passed without attenuation all voice frequencies but effectively blocked frequencies above the voice
band. This is disastrous for data communications, which depend on high frequencies to achieve the desired speed of transmission.
Bridge Taps
A bridge tap is any unterminated portion of a loop not in the direct talking
path. A bridge tap may be a used cable pair connected at an intermediate
point or an extension beyond the customer. For example, a drop wire that
provided a second line to a home is left in place even after the second set of
customer premises equipment (CPE) is removed. Records of this were not
always kept and assigning a particular copper pair to a high-speed data circuit
is far from a sure thing. Bridge taps do nasty things to data transmission.
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Coaxial cable consists of a single strand of copper running down the axis of
the cable. This strand is separated from the outer shielding by an insulator
made of foam or other dielectrics. Covering the cable is a conductive shield.
Usually an outer insulating cover is applied to the overall cablethis has
nothing to do with the carrying capacity of the cable. Because of the construction of the cable, obviously coaxial in nature, very high frequencies can
be carried without leaking out. In fact, dozens of TV channels, each 6 MHz
wide, can be carried on a single cable.
The fact that a coaxial cableor coaxcan support a tremendous bandwidth
has not been lost on the cable TV folk. A leader of the CATV industry said,
some years ago, We have more bandwidth by accident than the telephone
people have on purpose. Indeed, that is correct; piggybacking a telephone
channel on a coax cable is no challenge at all.
Fiber is the third transmission medium, and it is unquestionably the transmission medium of choice. Whereas transmission over copper utilizes frequencies in the megahertz range, transmission over fiber utilizes frequencies a million times higher. This is another way of saying that the predominant difference between electromagnetic waves and light waves is the frequency. This
difference, in turn, permits transmission speeds of immense magnitudes.
Transmission speeds as high as 9.9 Gbps have become commonplace in the
industry today. At this speed, the entire fifteen-volume set of Encyclopedia
Britannica can be transmitted in well under one second.
Laying fiber, on a per-mile basis, still costs somewhat more than laying copper. However, on a per-circuit basis there is no contest; fiber wins hands
down. However, if a local loop is being laid to a residence, there is little justification to installing fiberthere will never be a need for more than one or
two or three circuits. This realization has led to a transition in our thinking.
Shortly after the commercialization of fiber, we talked about fiber-to-thehome (FTTH). It was then realized that there was little need to install fiber
for a final several hundred yards, so the industry shied away from fiber-tothe-curb (FTTC). In such a system, fiber would carry a plurality of channels to
the curb, whereupon they would be broken down and applied to the copper drop leading to the home. In many cases even this was overkill, and fiberto-the-neighborhood (FTTN) is now being used. The message is clear: Apply
fiber when it is economical to do so, and otherwise rely on copper.
One final approach is being used in many areas, and it often proves workable.
This is a combination of fiber and coax or, as it is known, hybrid fiber/coax
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(HFC). As we have seen, coax has a greater bandwidth than copper but a
smaller one than fiber. Also, in some 60 percent of the homes in the United
States, coax in the form of cable TV goes to the home; tying fiber to coax for
the final several hundred yards makes technological sense.
Fiber comes in several forms; the two predominant ones are multimode and
single-mode (see Figure 4). As can be seen, the total strand diameter for both
is about 125 microns (a micron is a millionth of a meter). However, the ultrapure glass that forms the core transmission medium is between 50 and 62.5
microns for the multimode fiber and about 810 microns for the single-mode
fiber. One would think that the multimode fiber would have a greater carrying capacity; however, just the opposite is true. With single-mode fiber, only
one ray or mode can travel down the strand, and this makes for a simpler job
in regenerating the signal at points along the span. In fact, single-mode fiber
makes up the majority of todays long-distance network.

Figure 4: Optical Fiber Sizes


The tremendous capacity of fiber certainly makes for more efficient communication; however, placing so much traffic on a single strand makes for greater
vulnerability. Most of the disruptions in the long-distance network are a result
of physical interruption of a fiber run. It is called backhoe fade.
Wireless communications is the final option as a transmission medium. This
can take several forms: microwave, synchronous satellites, low-earth-orbit
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satellites, cellular, personal communications service (PCS), etc. Some of these
will be described in more detail later. In every case, however, a wireless system obviates the need for a complex wired infrastructure. In the case of synchronous satellites, transmission can take place across oceans or deserts. With
microwave there is no need to plant cable, and in mountainous territories this
is a significant advantage. Cellular and PCS afford mobility. There are advantages and disadvantages to each.

5. TRANSMISSION TECHNOLOGY
Most transmissionat least most transmission in the local exchange plant
is analog in nature. That is, the signal being transmitted varies continuously,
both in frequency and in amplitude. A high-pitched voice mostly contains
high frequencies; a low-pitched voice, low frequencies. A loud voice, a highamplitude signal; a soft voice, a low-amplitude signal.
In the long-distance network, and more and more in the local exchange plant,
digital transmission is being used. A digital signal is comprised of a stream of
1s and 0s that portray the analog voice signal by means of a code.
Analog signals can be combined, i.e., multiplexed, by combining them with a
carrier frequency. When there is more than one channel, this is called frequency division multiplexing (FDM). FDM was used extensively in the past,
but now has generally been replaced with the digital equivalenttime
division multiplexing (TDM). The most popular TDM system is known as
T1. In a T1 system, an analog voice channel is sampled 8,000 times per second, and each sample is encoded into a seven-bit byte. Twenty-four such
channels are mixed on these two copper pairs and transmitted at a bit rate of
1.544 Mbps. T1 remains an important method of transmitting voice and data
in the PSTN (see Figure 5).

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Figure 5: Pulse Code Modulation (PCM) Sampling, Quantizing, and


Encoding Process: In This Example, a 3-Bit Encoding Scheme Is Used for
Quantizing the Total Amplitude
Such a digital transmission scheme (and certainly there are modifications of it
that improve efficiency, capacity, or quality, etc.) works hand in glove with
the digital-switching schemes we talked about previously. Those 1s and 0s
need not be transmitted through an actual circuit in that switch, but rather
one can simply turn on and off the various electronic devices that make up
that switch.
Thus, a talking path (i.e., a switched circuit) in the PSTN can be either analog
or digital or a combination thereof. In fact, a digital signal can be transmitted
over a packet-switched network as easily as a circuit-switched network. Now
if we consider the next step, we see that digitized voice is little different from
data, and if data can be transmitted over a packet network, then so can digitized voice. This, of course, is now known as voice over the Internet. The
challenge, of course, is to get the transmitted signal to the destination fast
enough. After all, this may well be a time-sensitive voice conversation. A second challenge is to get each packet, which is a small piece of a voice conversation, to the destination in the proper order. Progress is being made, and we
can well believe that packet switching will play an important role in the
PSTN of tomorrow.

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6. BROADBAND ACCESS AND SERVICE


Transmission in the telecommunications networks of today is, more and
more, digital in nature, and the transmission medium of choice is fiber.
Digital, however, does no more than imply a string of 1s and 0s racing
through the network. But how are these 1s and 0s to be arranged? At what
speed are they to travel? What route should they take? Answers to questions
such as these have taken many forms and have made for the most complicated aspect of the telecommunications business.
There has never been a scarcity of coding schemes in the industry. Starting
with Morse code, going to the Baudot code, then the ASCII code, we have
seen each providing for better transmission and higher quality. In this section
we will discuss three codes: the most popular and important three, but certainly not the only three being used.
SONET
Synchronous optical network (SONET) is a standard for optical telecommunications transport. The SONET standard is expected to provide the transport infrastructure for worldwide telecommunications for at least the next two or three
decades. It defines a technology for carrying many signals of different capacities
through a synchronous optical hierarchy. The standard specifies a byte-interleaved multiplexing scheme. The synchronous optical hierarchy mentioned is
shown in Table 2.

Table 2: SONET Hierarchy


The SONET standards govern not only rates, but also interface parameters; formats; multiplexing methods; and operations, administration, maintenance, and
provisioning for high-speed transmission. We most often hear of SONET rings,
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tion. The system is designed so that transmission can take place in either direction; should there be a fault at any one location, transmission will immediately
take place in the opposite direction. That is, the system is self-healing.
ATM
Asynchronous transfer mode (ATM) is a high-performance switching and
multiplexing technology that utilizes fixed-length packets to carry different
types of traffic. Information is formatted into fixed-length cells consisting of
forty-eight bytes (eight bits per byte) of payload and five bytes of cell header.
The fixed cell size guarantees that time-critical information (e.g., voice or
video) is not adversely affected by long data frames or packets. Of course, if
the cells were longer in length, the system would be more efficient because
the header would take up a smaller percentage of the total cell.
Multiple streams of traffic can be multiplexed on each physical facility and
can be managed so as to send the streams to many different destinations.
This enables cost savings through a reduction in the number of interfaces and
facilities required to construct a network.
ADSL
Asymmetric digital subscriber line (ADSL) is, essentially, a modem that
employs a sophisticated coding scheme. This coding scheme permits transmission over copper pairs at rates as high as 6Mbps for distances of 9,000 to
12,000 feet. Speeds of this magnitude bring to mind television signals; a 6
Mbps channel can easily handle a television movie.
ADSL succeeds because it takes advantage of the fact that most of its target
applications (video-on-demand, home shopping, Internet access, etc.) function
perfectly well with a relatively low upstream data rate. Hence the word
asymmetric. LECs are now using ADSL as an access technology for their television businesses and for Internet access.

7. WIRELESS
The first commercially available radio and telephone system, known as
improved mobile telephone service (IMTS), was put into service in 1946. This
system was quite unsophisticatedbut then there were no solid state electronics available.

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With IMTS, a tall transmitter tower was erected near the center of a metropolitan area. Several assigned channels were transmitted and received from
the antenna atop this tower. Any vehicle within range could attempt to seize
one of those channels and complete a call. Unfortunately, the number of
channels made available did not come even close to satisfying the need. To
make matters worse, as the metropolitan area grew, more power was applied
to the transmitter or receiver, the reach was made greater, and still more erstwhile subscribers were unable to get dial tone.
The solution to this problem was cellular radio. Metropolitan areas were
divided into cells of no more than a few miles in diameter, and each cell operated on a set of frequencies (send and receive) that differed from the frequencies of the adjacent cells. Because the power of the transmitter in a particular
cell was kept at a level just high enough to serve that cell, these same sets of
frequencies could be used at several places within the metropolitan area.
Beginning in 1983, two companies, one called a wireline company and the
other called a nonwireline carrier, were given a franchise to operate in each
major territory.
Two characteristics of cellular systems were important to their usefulness.
First, the systems controlled handoff. As subscribers drove out of one cell and
into another, their automobile radios, in conjunction with sophisticated electronic equipment at the cell sites (also known as base stations) and the telephone switching offices (also known as mobile telephone switching offices
[MTSOs] ), transferred from one frequency set to another with no audible
pause. Second, systems were also designed to locate particular subscribers by
paging them in each of the cells. When the vehicle in which a paged subscriber was riding was located, the equipment assigned sets of frequencies to
it and conversation could begin (see Figure 6).

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Figure 6: In a Cellular System, Vehicle-to-Cell-Site Communications Take


Place Using FDMA, TDMA, CDMA, GSM, etc.; Communications between
the Cell Site and the MTSO Utilize Conventional Techniques
The initial transmission technology used between the vehicle and the cell site
was analog in nature. It is known as advanced mobile phone service (AMPS).
The analog scheme used was called frequency division multiple access
(FDMA).
But the age of digital transmission was upon us, and many companies operating in this arena concluded that a digital transmission scheme would be preferred. The result was time division multiple access (TDMA). In Europe, the
selected scheme was an adaptation of the TDMA used in the United States,
and it was called groupe special mobile. Since then, the name has been
changed to global system for mobile communications (GSM).
As if that were not enough, a third group of companies determined that a
special spread-spectrum, or frequency-hopping, scheme would be even better,
and this also was developed and trialed. This is called code division multiple
access (CDMA). Thus, there are at least four schemes that may be used for
communications between a vehicle and the cell site. Communications
between the cell site and the MTSO utilize more conventional techniques,
such as microwave, copper pairs, or fiber optics.

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The continuing growth of cellular communications (there are presently about
20,000 new subscribers signing on each day), led government and industry in
the United States to search for additional ways to satisfy the obvious need
not only for ordinary telephone service but also for special services and features, smaller telephones, and cellular-phone use. This search led to the PCS
industry. Additional frequency bands were allocated for their use, and rather
than assign them to the first comers or by way of a lottery, the FCC auctioned them off through a sophisticated bidding contest that brought the U.S.
treasury billions of dollars.
Geosynchronous satellites represent yet another way of providing wireless
communications. These satellites, located 22,300 miles above the earth,
revolve around the earth once each twenty-four hoursthe same as the earth
itself. Consequently they appear to be stationary. Communications between
two places on earth can take place by using these satellites; one frequency
band is used for the uplink, and another for the downlink. Such satellite systems are excellent for the transmission of data, but they leave something to
be desired for voice communications. This is a result of the vast distance and
the time it takes for an electrical signal to make an earth-satellite-earth round
trip. That time amounts to one quarter of a second. A reply from the called
subscriber takes another quarter of a second, and the resultant half a second is
definitely noticeable. Consequently, voice communications are seldom carried
via geosynchronous satellites.
Yet another wireless telecommunications technology is the low earth orbit
(LEO) satellite system. LEOs are satellites that communicate directly with
handheld telephones on earth. Because these satellites are relatively lowless
than 900 milesthey move across the sky quite rapidly.
In a LEO system, the communications equipment on a satellite acts much like
the cell site of a cellular system. It catches the call from earth and usually
passes it to an earth-based switching system. Because of the speed of the
satellite, it is frequently necessary to hand off a particular call to a second
satellite just rising over the horizon. This is akin to a cellular system, except
that in this case it is the cell site that is moving rather than the subscriber.
Several systems are now in the planning stage, and in fact many satellites
have already been launched. The most noted is Iridium, created by Motorola,
which would utilize sixty-six satellites. A second system, called Globalstar,
would employ forty-eight satellites. There are at least two or three others
that are well along in preparations to launch.
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8. SELF-TEST
1. Alexander Graham Bell called his first telephone a synchronous telegraph.
a. true
b. false

2. The expiration of Bells basic patents in 1893 and 1894 was the starting signal for
open competition.
a. true
b. false

3. In 1946, Congress passed a Communications Act and established the Federal


Communications Commission.
a. true
b. false

4. Each telephone subscriber is connected to several central offices.


a. true
b. false

5. The local loop, or the connections between individual subscribers and central offices,
is also known as the last mile.
a. true
b. false

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6. The ________________ is a particular set of standards that allows the interworking of products from different vendors. It usually embodies a fiber-optic ring that will
permit transmission in both directions.
a. local-area network (LAN)
b. wide-area network (WAN)
c. synchronous optical network (SONET)
d. common channel signaling network

7. Packet switching is used for which of the following?


a. credit-card verification
b. automated teller machines
c. SS7
d. the Internet and the World Wide Web
e. all of the above

8. The types of media that can transmit information in the telecommunications world
are the following:
a. copper wire, coaxial cable, fiber, and wireless
b. hybrid fiber/coax and copper wire
c. wireless and copper wire
d. copper wire, coaxial cable, fiber, and hybrid fiber/coax

9. Analog signals can be ________ by combining them with a carrier frequency.


a. carried
b. transported
c. multiplexed
d. mixed
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10. ______________ is a high-performance switching and multiplexing technology
that utilizes fixed-length packets to carry different types of traffic.
a. asynchronous transfer mode (ATM)
b. asymmetric digital subscriber line (ADSL)
c. synchronous optical network (SONET)
d. none of the above

9. ACRONYM GUIDE
ADSL

asymmetric digital subscriber line

AMPS

advanced mobile phone service

ATM

asynchronous transfer mode

CDMA

code division multiple access

CO

central office

CPE

customer premises equipment

FCC

Federal Communications Commission

FDM

frequency division multiplexing

FDMA

frequency division multiple access

FTTC

fiber-to-the-curb

FTTH

fiber-to-the-home

FTTN

fiber-to-the-neighborhood

GSM

global system for mobile communications

HFC

hybrid fiber/coax

IMTS

improved mobile telephone service

IXC

interexchange carrier

LAN

local-area network

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LEC

local exchange carrier

LEO

low earth orbit

MFJ

modification of final judgement

MTSO

mobile telephone switching office

OC

optical carrier

PCM

pulse code modulation

PCS

personal communications service

PoP

point of presence

PSTN

public switched telephone network

RBOC

regional Bell operating company

SONET

synchronous optical network

SS7

signaling system 7

TDM

time division multiplexing

TDMA

time division multiple access

UTP

unshielded twisted pair

WAN

wide-area network

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DEFINITION
An intelligent network (IN) is a service-independent telecommunications network. That is, intelligence is taken out of the switch and placed in computer
nodes that are distributed throughout the network. This provides the network operator with the means to develop and control services more efficiently. New capabilities can be rapidly introduced into the network. Once introduced, services are easily customized to meet individual customer needs.

TUTORIAL OVERVIEW
This tutorial discusses how the network has evolved from one in which
switch-based service logic provided services to one in which service-independent advanced intelligent network (AIN) capabilities allow for service creation
and deployment.
As the IN evolves, service providers will be faced with many opportunities
and challenges. While the IN provides a network capability to meet the everchanging needs of customers, network intelligence is becoming increasingly
distributed and complicated. For example, third-party service providers will
be interconnecting with traditional operating company networks. Local number portability (LNP) presents many issues that can only be resolved in an IN
environment to meet government mandates. Also, as competition grows with
companies offering telephone services previously denied to them, the IN provides a solution to meet the challenge.

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TOPICS
1.

NETWORK EVOLUTION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .131

2.

INTRODUCTION OF INTELLIGENT NETWORKS . . . . . . . . . . . . . .133

3.

BENEFITS OF INTELLIGENT NETWORKS . . . . . . . . . . . . . . . . . . . .135

4.

AIN RELEASES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .137

5.

AIN RELEASE 1 ARCHITECTURE . . . . . . . . . . . . . . . . . . . . . . . . . . .138

6.

THE CALL MODEL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .140

7.

AIN RELEASE 0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .141

8.

AIN RELEASE 0.1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .143

9.

AIN RELEASE 0.2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .145

10. AIN SERVICE CREATION EXAMPLES . . . . . . . . . . . . . . . . . . . . . . . .146


11. OTHER AIN SERVICES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .151
12. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .156
13. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .158

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1. NETWORK EVOLUTION
Plain Old Telephone Service (POTS)
Prior to the mid-1960s, the service logic, as shown in Figure 1, was hard-wired
in switching systems. Typically, network operators met with switch vendors,
discussed the types of services customers required, negotiated the switching
features that provided the services, and finally agreed upon a generic release
date for feature availability. After this, the network operator planned for the
deployment of the generic feature/service in the switching network fabric.

Figure 1: Plain Old Telephone Service


This process was compounded for the network operator with switching systems from multiple vendors. As a result, services were not offered ubiquitously across an operators serving area. So, a customer in one end of a city, county, or state may not have had the same service offerings as a person in another part of the area.
Also, once services were implemented, they were not easily modified to meet
individual customer requirements. Often, the network operator negotiated
the change with the switch vendor. As a result of this process, it took years to
plan and implement services.
This approach to new service deployment required detailed management of
calling patterns and providing new trunk groups to handle calling patterns. As
customer calling habits changed (i.e., longer call lengths, larger calling areas,
and multiple lines in businesses and residences) the demand on network operators increased.

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Stored Program Control (SPC)
In the mid-1960s, stored program control (SPC) switching systems were introduced. SPC was a major step forward because now service logic was programmable where, in the past, the service logic was hard-wired. As a result, it
was now easier to introduce new services. Nevertheless, this service logic
concept was not modular. It became increasingly more complicated to add
new services because of the dependency between the service and the servicespecific logic. Essentially, service logic that was used for one service could not
be used for another service. As a result, if customers were not served by an
SPC switching system, new services were not available to them.

Common Channel Signaling Network (CCSN)


Another aspect of the traditional services offerings was the call set-up informationthat is, the signaling and call supervision that takes place between
switching systems and the actual call. When a call was set up, a signal and
talk path used the same common trunk from the originating switching system
to the terminating switching system. Often there were multiple offices
involved in the routing of a call. This process seized the trunks in all of the
switching systems involved. Hence, if the terminating end was busy, all of the
trunks were set up unnecessarily.
The network took a major leap forward in the mid-1970s with the introduction of the common channel signaling network (CCSN), or SS7 network for
short. Signaling system number 7 (SS7) is the protocol that runs over the
CCSN. The SS7 network consists of packet data links and packet data switching systems called signaling transfer points (STPs).
The SS7 network (see Figure 2) separates the call set-up information and talk
path from the common trunks that run between switching systems. The call
set-up information travels outside the common trunk path over the SS7 network. The type of information transferred included permission for the call
set-upthat is, whether or not the called party was busy.

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Figure 2. Common Channel Signaling


SS7 technology frees up trunk circuits between switching systems for the
actual calls. The SS7 network enabled the introduction of new services, such
as caller ID. Caller ID provides the calling partys telephone number, which is
transmitted over the SS7 network.
The SS7 network was designed before the IN concept was introduced.
However, telephone operators realized that there were many advantages to
implementing and using SS7 network capabilities.

2. INTRODUCTION OF INTELLIGENT NETWORKS


During the mid-1980s, regional Bell operating companies (RBOCs) began
requesting features that met the following objectives:
rapid deployment of services in the network
vendor independence and standard interfaces
opportunities for nonRBOCs to offer services for increased
network usage
Bell Communications Research (Bellcore) responded to this request and
developed the concept of intelligent network 1 (IN/1), shown in Figure 3.

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Figure 3: Intelligent Network (IN/1)


The introduction of the IN/1 marked the first time that service logic was external to switching systems and located in databases called service control points
(SCPs). Two services evolved that required IN/1 service logic: the 800 (or
freephone) service and the calling card verification service (or alternate billing
service [ABS]). Because of the service-specific nature of the technology, these
services required two separate SCPs. In order to communicate with the associated service logic, software was deployed in switching systems. This switching system software enabled the switching system to recognize when it was
necessary to communicate with an SCP via the SS7 network.
With the introduction of the SCP concept, new operations and management
systems became necessary to support service creation, testing, and provisioning. In the above figure, note the term service-specific management systems under
the box labeled service management system. This means that the softwaredefined hooks or triggers are specific to the associated service. For example,
an 800 service has an 800-type trigger at the switching system, an 800-service
database at the SCP, and an 800-service management system to support the
800 SCP. In this service-specific environment, the 800-service set of capabilities cannot be used for other services (e.g., 900 service). Although the service
logic is external to the switching system, it is still service-specific.

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Figure 4: AIN Architecture


At first glance, Figure 4 looks similar to Figure 3. However, there is one fundamental difference. Notice the wording service-independent management systems under the box labeled service management system. Now, following
the IN/1 800 service-specific example, the AIN service-independent software
has a three-digit trigger capability that can be used to provide a range of
three-digit services (800, 900, XXX, etc.) as opposed to 800 service-specific
logic. Likewise, the SCP service logic and the service management system are
service-independent, not service specific. AIN is a service-independent network capability!

3. BENEFITS OF INTELLIGENT NETWORKS


The main benefit of intelligent networks is the ability to improve existing
services and develop new sources of revenue. To meet these objectives,
providers require the ability to accomplish the following:
introduce new services rapidly
IN provides the capability to provision new services or modify existing
services throughout the network with physical intervention.

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provide service customization
Service providers require the ability to change the service logic rapidly
and efficiently.
Customers are also demanding control of their own services to meet
their individual needs.
establish vendor independence
A major criterion for service providers is that the software must be
developed quickly and inexpensively. To accomplish this, suppliers
must integrate commercially available software to create the applications required by service providers.
create open interfaces
Open interfaces allow service providers to introduce network elements
quickly for individualized customer services. The software must interface with other vendors products while still maintaining stringent network operations standards. Service providers are no longer relying on
one or two vendors to provide equipment and software to meet customer requirements.
AIN technology uses the embedded base of stored program-controlled
switching systems and the SS7 network. The AIN technology also
allows for the separation of service-specific functions and data from
other network resources. This feature reduces the dependency on
switching-system vendors for software development and delivery
schedules. Service providers have more freedom to create and customize services.
The SCP contains programmable service-independent capabilities (or service
logic) that are under the control of service providers. The SCP also contains
service-specific data that allows service providers and their customers to customize services. With the IN, there is no such thing as one size fits all; services are customized to meet individual needs.
As service logic is under the service providers control, it is easier to create
services in a cost-effective manner. Network providers can offer marketfocused service trials by loading service logic in an SCP and triggering capabilities in one or more switching systems.

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Accepted standards and open, well-documented interfaces provide a standard
means of communicating between switching systems and SCPs, especially in
a multivendor environment.

3A. LOCAL NUMBER PORTABILITY


The Telecommunications Act of 1996 is having a profound impact on the U.S.
telecommunications industry. One area of impact that is being felt by everyone is local number portability (LNP). For LNP, the Federal Communications
Commission (FCC) requires the nations local exchange carriers (LECs) to
allow customers to keep their telephone numbers if they switch local carriers.
The LECs must continue to maintain the quality of service and network reliability that the customer always received.
The rules required that all LECs begin a phased deployment of a long-term
service providerportability solution no later than October 1, 1997, in the
nations largest metropolitan statistical areas. This deployment had to be
completed by December 31, 1998. Incumbent service providers are required
to make a number portable within six months after a request is received.
Wireless carriers are also affected by LNP. December 31, 1998, was the date
that wireless carriers had to be able to complete a call to a ported wireline
number. By June 30, 1999, there must be full portability between wireless and
wireline, including roaming capabilities.
AIN is a logical technology to help service providers meet this mandate.
Many providers are looking to AIN LNP solutions because of the flexibility
that AIN provides without the burden of costly network additions.
For more information on the Telecommunications Act of 1996, visit the FCCs
home page on the World Wide Web at http://www.fcc.gov.

4. AIN RELEASES
The demand for AIN services far exceeded the availability of network functionality. Service providers could not wait for all the features and functionality
as described in AIN Release 1. AIN Release 1 defined all types of requirements,
which made the capability sets too large to be adapted by the industry.

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In North America, the industry agreed to develop subsets of AIN Release 1
that provided for a phased evolution to its release. AIN 0.1 was the first subset targeted for use in the industry.
Bellcore developed functionality to address the FTS 2000 requirements set
forth by the United States government. The RBOCs adapted these requirements to meet their customers immediate needs. This effort resulted in AIN
Release 0, which had a time frame before the availability of AIN 0.1.
Meanwhile, the international standards body, the International Telecommunications
Union (ITU), embraced the concepts put forth in the AIN Release 1 requirements.
The ITU developed an international IN standard called Capability Set 1, or CS1.
As with AIN Release 1 in North America, CS1 was all encompassing. To meet
the market demand, the ITU formed a subgroup called ETSI to focus on the
immediate needs. This subgroup developed the core INAP capabilities. Many
PTTs and their switch vendors have adopted the ETSI Core INAP as the standard
and are providing core INAP capabilities.
The following modules discuss the functionality of various AIN releases, as
well as the international standards.

5. AIN RELEASE 1 ARCHITECTURE


Figure 5 shows the target AIN Release 1 architecture, as defined in Bellcore
AIN Generic Requirements (GRs):

Figure 5: AIN Release 1

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The service switching point (SSP) in this diagram is an AINcapable switching
system. In addition to providing end users with access to the network and
performing any necessary switching functionality, the SSP allows access to
the set of AIN capabilities. The SSP has the ability to detect requests for
AINbased services and establish communications with the AIN service logic
located at the SCPs. The SSP is able to communicate with other network systems (e.g., intelligent peripherals) as defined by the individual services.
The service control point (SCP) provides the service control. There are two
basic parts to an SCP. One part is the application functionality in which the
service logic is installed after the services have been created. This application
functionality sits on top of the second basic SCP parta set of generic platform functionalities that are developed by SCP vendors. This platform functionality is shared among the service logic application programs in the application functionality. The platform functionality also provides the SS7 interface
to switching systems. As shown in Figure 5, the SCP is connected to SSPs by
the SS7 network.
The intelligent peripheral (IP) provides resources such as customized and concatenated voice announcements, voice recognition, and dual-tone multifrequencies (DTMF) digit collection. The IP contains a switching matrix to connect users to these resources. In addition, the IP supports flexible information
interactions between an end user and the network. It has the resource management capabilities to search for idle resources, initiate those resources, and
then return them to their idle state.
The interface between the SSP and the IP can be integrated services digital
network (ISDN), primary rate interface (PRI), and/or basic rate interface (BRI).
The IP has the switching functionality that provides the ISDN interface to the
switching system.
The adjunct shown in Figure 5 is functionally equivalent to an SCP, but it is
connected directly to an SSP. A high-speed interface supports the communications between an adjunct and an SSP. The application-layer messages are identical in content to those carried by the SS7 network between the SSP and SCP.

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6. THE CALL MODEL


The call model is a generic representation of SSP call-processing activities
required to establish, maintain, and clear a basic call. The call model consists
of points in calls (or PICs), detection points (DPs), and triggers. These are
depicted in Figure 6.

Figure 6. The Call Model: Basic Concept


PICs represent the normal switching system activities or states that a call goes
through from origination to termination. For example, the null state or the
idle state is when the SSP is actually monitoring the customers line. Other
examples of states, or PICs, are off-hook (or origination attempt), collecting
information, analyzing information, routing, alerting, etc.
Switching systems went through similar stages before AIN was developed.
However, the advent of AIN introduced a formal call model to which all
switching systems must adhere. In this new call model, trigger detection
points (TDPs) were added between the PICs. SSPs check TDPs to see if there
are any active triggers.
There are three types of triggers: subscribed or line-based triggers, groupbased triggers, and office-based triggers. Subscribed triggers are provisioned
to the customers line, so that any calls originating from or terminating to that
line would encounter the trigger. Group-based triggers are assigned to groups
of subscribers (e.g., business or Centrex groups). Any member of a softwaredefined group will encounter the trigger. Office-based triggers are available to

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everyone who is connected to the telephone switching office or has access to
the North American numbering plan. Office-based triggers are not assigned to
individuals or groups.
If an active trigger is detected, normal switching-system call processing is suspended until the SSP and SCP complete communications. For example, in
Figure 6, suppose an AIN call has progressed through the null state or PIC, the
off-hook PIC, and is currently at the collecting information PIC. Normal call
processing is suspended at the information-collected TDP because of an active
off-hook delayed trigger. Before progressing to the next (analyze information)
PIC, the SSP assembles an information-collected message and sends it to the
SCP over the SS7 network. After SCP service logic acts on the message, the
SCP sends an analyze-route message that tells the SSP how to handle the call
before going to the next PIC.
Essentially, when the SSP recognizes that a call has an associated AIN trigger,
the SSP suspends the call processing while querying the SCP for call-routing
instructions. Once the SCP provides the instruction, the SSP continues the
call-model flow until completion of the call. This is basically how a call
model works, and it is an important part of AIN. This concept differs from
the preAIN switching concept in which calls were processed from origination state to the call-termination state without call suspension.

7. AIN RELEASE 0
The AIN Release 0 call model has three trigger checkpoints (TCPs). At each
TCP, there are one or more triggers. For example, the off-hook TCP includes
the off-hook immediate trigger. If a subscribers line is equipped with this
trigger, communications with the SCP will occur if the switching system
detects an off-hook condition. For an off-hook delayed trigger, one or more
digits are dialed before triggering to the SCP. At the digit collection and analysis TCP, collected digits are analyzed before triggering. Triggering may also
occur at the routing stage of a call. This call model is shown in Figure 7.

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Figure 7: AIN Release 0 Call Model


When a switching system recognizes that a call needs AIN involvement, it
checks for overload conditions before communicating with the SCP. This
process is called code gapping. Code gapping allows the SCP to notify the
switching system to throttle back messages for certain NPAs or NPANXXs.
When code gapping is in effect, some calls may receive final treatment. For
others, a provide instruction message is sent to the SCP. Depending on the
SCP service logic, it will respond to the switching system with any of the callprocessing instructions shown in Figure 8.

Figure 8. AIN Release 0 Functions

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AIN Release 0 provided 75 announcements at the switching system. Release 0
was based on American National Standards Institute (ANSI) transaction capability application part (TCAP) Issue 1. TCAP is at Layer 7 of the SS7 protocol
stack. This means that there is only one message sent from the SSP to the
SCP, no matter what trigger is hit at any of the three TCPs.

8. AIN RELEASE 0.1


AIN 0.1 is the first subset of AIN Release 1. There are two fundamental differences between AIN Release 0 and AIN 0.1 The first is a formal call model
and the second is the messaging sets between the switching system and the
SCP. The formal call model is separated into the originating call model (originating half call) and the terminating call model (terminating half call). The
AIN Release 0 call model did not distinguish between originating and terminating. A standard or formal call model is necessary as we evolve to the
Target AIN Release 1 capability, because the capabilities will have more PICs
and TDPs. Also, there will be multiple switch types and network elements
involved. Therefore, the service logic will need to interact with every element
that will be required in the network.
AIN 0.1 includes several other major features. There are 254 announcements
at the switching system, which provides more flexible messages available to
customers. There are additional call-related and noncall-related functions as
well as three additional triggers: the N11 trigger, the 3-6-10-digit trigger, and
the termination attempt trigger. More triggers provide additional opportunities for SCP service logic to influence call processing. (Note: TCP was an AIN
Release 0 term that changed to TDP in AIN 0.1.)
There are several AIN 0.1 noncall-related capabilities. The SCP has the ability
to activate and deactivate subscribed triggers. The AIN 0.1 SCP can also monitor resources. In addition to sending a call-routing message to the switching
system, the SCP may request that the switching system monitor the busy/idle
status of a particular line and report changes. AIN 0.1 also supports standard
ISDN capabilities.
As mentioned previously, there is a distinction between the originating side and
the terminating side of a service switching point. This means that both originating and terminating triggers and service logic could influence a single call.

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Figure 9 shows a portion of the AIN 0.1 originating call model. The AIN 0.1
originating call model includes four originating trigger detection pointsorigination attempt, information collected, information analyzed, and network busy.

Figure 9. AIN 0.1 Originating Call Model


The AIN 0.1 terminating call model includes one TDPtermination attempt,
as depicted in the partial call model in Figure 10.

Figure 10. AIN 0.1 Terminating Call Model

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AIN 0.1: SSPSCP Interface
The AIN 0.1, as shown in Figure 11, is based on ANSI TCAP Issue 2, which
means that the message set is different from the message set in ANSI TCAP
Issue 1. For example, in AIN Release 0, there is only one message sent from
the SSP to the SCP no matter what trigger is hit at any of the three TCPs. In
AIN 0.1, separate messages are sent for the four originating and one terminating TDP.

Figure 11. AIN 0.1 SSPSCP Interface

9. AIN RELEASE 0.2


AIN 0.2 builds on AIN 0.1 with additional capabilities to support two service
drivers: Phase-2 personal communication service (PCS) and voice-activated
dialing (VAD). While AIN 0.2 is focused on capabilities to supports PCS and
VAD, all requirements for these capabilities are defined in a service-independent manner. AIN 0.2 capabilities will include the following:
ISDNbased SSPIP interface
busy and no-answer triggers
next event lists processing
default routing and additional functions in all operations areas (e.g.,
network testing)

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The two primary AIN 0.2 capabilities are the ISDN interface between a
switching system and an ISDNcapable device (such as an IP) and the addition of busy and no-answer triggers.
Next event lists processing is another important capability. In addition to
TDPs, AIN 0.2 includes event detection points (EDPs). With EDPs, the SCP
will have the ability to send a next event list to the SSP. This next event list is
used by the SSP to notify the SCP of events listed in the next event list. These
events may include busy, no answer, terminating resource available, etc.
AIN 0.2 also includes default routing capabilities. This means that when calls
encounter error conditions, they can be sent to a directory number, an
announcement, etc., as opposed to sending it to final treatment, as is the case
in AIN 0.1.
AIN Release 0 and AIN 0.1 assumed that the announcements were switchbased. With the introduction of 0.2, announcements can reside in an external
database, such as an IP.
If the SCP sends a send-to-resource message to the switching system to have
the IP play an announcement or collect digits, the switching system connects
the customer to the IP via the SSPIP ISDN interface. The end user exchanges
information with the IP. The IP collects the information and sends it to the
switching system. The switching-system forwards the information to the
SCP. One of the fundamental switching-system capabilities is the interworking of SS7 (SCP) messages with ISDN messages (SSPIP).
In addition, the SSP may control IP resources without SCP involvement. Voice
activated dialing is an example. A VAD subscriber could be connected to the
IP voice-recognition capabilities upon going off-hook. The VAD subscriber
says call mom, and the IP returns moms telephone number to the switching system. The switching system would recognize moms number as if the
subscriber had actually dialed the number.

10. AIN SERVICE CREATION EXAMPLES


The previous modules addressed the architecture and the theory of the AIN.
This section will discuss various aspects of service creationthe tool that
builds the representation of the call flow for each individual customer. Many

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AIN software vendors have paired service creation software with state-of-theart computer graphics software to eliminate the need for traditional programming methods. Through the use of menu-driven software, services are created
by inputting various service parameters.

Building-Block Approach
Figure 12 provides an example of a building-block approach to creating AIN
services. Play announcement, collect digits, call routing, and number translation building blocks are shown here. The SSP has the ability to play
announcement and collect digits, as does the IP. Routing the call is an SSP
function, and number translation is an SCP capability. By arranging these four
capabilities or building blocks in various combinations, services such as 800
service with interactive dialing, outgoing call screening, and area number calling can be created.

Figure 12. AIN Service Example: Building-Block Approach


Service Creation Template
Figure 13 represents what a service-creation template might look like. For an
outgoing call-screening service, the service begins with the customers telephone number. This example allows the customer to screen 900 numbers
while still having the ability to override 900 screening by entering a PIN.
Except for 703-974-1234, all non-900 calls are processed without screening.

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Figure 13. AIN Service Example: Service Creation Template

AIN Service Creation Template


Digit Extension Dialing Service
A 5-digit extension dialing service is displayed in Figure 14. It allows for
abbreviated dialing beyond central office boundaries. If an employee at location 1 wants to call an employee at location 2 by dialing the extension number 111, 2111 would be dialed. Although 2111 is not a number that a switching system can use to route the call, a customized dialing plan trigger is
encountered after 2111 is dialed and a query is sent to the SCP. Service logic
at the SCP uses the 2111 number to determine the real telephone number of
the called party.

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Figure 14. AIN Service Example: Digit Extension Dialing Service

5-Digit Extension Dialing Service


Disaster-Recovery Service
Figure 15 illustrates a disaster-recovery service. This service allows businesses
to have calls routed to one or more alternate locations based on customer service logic at the SCP. Calls come into the switching system served by the normal location. After triggering, communication with the SCP occurs. Based on
the service logic, the call could be either routed to the normal business location or to one or more alternate business locations.

Figure 15. Disaster-Recovery Service


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Area Number Calling Service
An area number calling (ANC) service is shown in Figure 16. This service is
useful for companies or businesses that want to advertise one telephone number but want their customers calls routed to the nearest or most convenient
business location. The SCP service logic and data (e.g., zip codes) are used to
make a match between the calling partys telephone number and their geographical location. The call is then routed to the company or business location that is closest to or most convenient for the calling party.

Figure 16. Area Number Calling (ANC) Service

Do Not Disturb Service


Finally, a do not disturb service is displayed in Figure 17. This is a service in
which the Smith family has terminating screening service logic at the SCP.
Whenever someone calls them, the service logic determines whether the call
should be routed to the Smiths telephone or play an announcement. In this
particular case, a telemarketer calls the Smith family. The SCP tells the
switching system to route the telemarketer to an announcement.

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Figure 17: Do Not Disturb Service


The customers SCP service logic may also contain a list of numbers that will
be received while do not disturb is active. In that case, if the SCP finds a
match between the calling party number and a number on the list, the call is
routed to the Smith family.

11. OTHER AIN SERVICES


The following list describes the services that companies have developed using
AIN/IN technology. Some services are tariffed, deployed in the network, and
generating revenues. Others are in market or technical trials, getting ready for
deployment. There are other services that are either planned for deployment
or were developed for demonstration purposes.
N11 access service: With this service, a unique code is used to access a
service gateway to information service providers (ISPs), such as newspapers or libraries. The subscriber may either preselect an ISP for automatic routing or request block calls to ISPs.
Basic routing: The basic routing function allows the subscriber to
route calls to a single destination, as defined in the system.

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Single-number service: Routing by single-number service allows calls
to have different call treatments based on the originating geographical
area and the calling party identification.
Routing by day of week: The routing by day of week function allows
the service subscriber to apply variable call routings based on the day
of the week that the call is placed.
Routing by time of day: The routing by time-of-day function allows
service subscribers to apply variable call routings based on the time of
the day that the call is made.
Selective routing: This service is tied to the call-forwarding feature
generally offered as a switch-based feature. With the AIN, when a call
to a selective routing customer is forwarded, the SCP determines
where to route the forwarded call based on the callers number.
Call allocator: The call-allocator service feature allows the service subscriber to specify the percentage of calls to be distributed randomly up
to five alternate call handling treatments.
Alternate destination on busy day: The alternate destination on busy
day (ADOB) service feature allows the service subscriber to specify a
sequence of destinations to which calls will be routed if the first destination is busy.
Command routing: A service subscriber predefines a set of alternate
call treatments to handle traffic in cases of emergency, unanticipated
or anticipated demand peaks, or for any other reason that warrants an
alternate call treatment.
Call gate: This is a versatile out-going call-screening service. Call gate
supports a personal identification number (PIN) and screening based
on time of day and day of week.
Personal access: Personal access is a type of follow-me service. A virtual telephone number is assigned to the personal access service subscriber. When a caller dials this number, the software determines how
to route the call.
Calling party pays: A service offered to cellular customers. It notifies
the calling party that they are trying to reach a cellular number. If they

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choose to complete the call, they will incur the connect charge of the
called party. If they elect not to incur the cost, the call may either be
terminated or routed to called partys voice mail.
Remote access to call forwarding (ultraforward): The ultraforward service allows remote access to call forwarding. Callers may, from any
location in the world, call in remotely and activate and/or change their
call forwarding number.
Portable number service: Portable number service (PNS) features
enhanced call forwarding for large business subscribers. It provides
subscribers with the ability to maintain a personal itinerary, which
includes time-of-day, day-of-week (TOD/DOW) schedules, callsearching schedules, and call-routing information. PNS subscribers also
have the ability to override their schedules with default routing instructions. This service is intended for companies with employees who are
in highly mobile environments requiring immediate availability.
Enhanced 800 Service (freephone): A customers call to an 800 service
subscriber can be routed to different destinations. Instances of routing
include the geographical location of the caller, the time and day the call
is made, and the caller responses to prompts. The subscriber sets alternate routing parameters for the call if the destination is busy or unavailable, thereby redirecting and allowing for completion of the call.
Mass calling service: mass calling service (MCS) is a polling and information service that permits simultaneous calling by a large number of
callers to one or more telephone numbers. MCS provides a variety of
announcement-related services that connect a large number of callers
(who dial an advertised number) to recorded announcement devices.
Two types of offerings are mass announcements, such as time and
weather, and televoting, which allows callers to register their opinions
on a topic of general interest.
Automatic route selection/least-cost routing: With this service, subscribers design a priority route for every telephone number dialed. The
system either directs calls or blocks calls to restricted-privilege users.
Work-at-home: This service allows an individual to be reached at
home by dialing an office number, and permits the employee to dial

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an access code from home, make long-distance calls, and have them
billed and tracked to a business telephone number.
Inmate service routes prisoners calls, tracks the call information, and
offers call control features such as prompts for personal identification
numbers, blocking certain called numbers, and time or day restrictions.
Holding room: Transportation companies passengers use this service
to inform families or business associates of transportation delays or
cancellations.
Call prompter: The call-prompter service feature allows a service subscriber to provide an announcement that requests the caller to enter a
digit or series of digits via a dual-tone multifrequency (DTMF) telephone. These digits provide information that are used to direct routing
or as a security check during call processing.
Call counter: The call-counter service feature increases a counter in the
televoting (TV) counting application when a call is made to a TV number. The counts are managed in the SCP, which can accumulate and
send the results during a specific time period.
500 access service: This routing service allows personal communications service (PCS) providers the ability to route calls to subscribers
who use a virtual 500 number.
PBX extend service: This service provides a simple way for users to
gain access to the Internet network.
Advertising effectiveness service: This service collects information on
incoming calls (for example, ANI, time, and date). This information is
useful to advertisers to determine the demographics of their customers.
Virtual foreign exchange service: Uses the public switched network to
provide the same service as wired foreign exchange service.
Automated customer name and address (ACNA) originating line
blocking: This is ACNA with the ability to block a line from being
accessed by the service.

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AIN for the case teams: AIN for the case teams (ACT) allows technicians to dial from a customer-premises location anywhere in the service region and connect to a service representative supported by an
ACD. Through voice prompts, the technician is guided to the specific
representative within a case team pool within seconds, with no toll
charges to the customer.
Regional intercept: Regional intercept instructs callers of new telephone
numbers and locations of regional customers. This service also forwards calls to the new telephone number of the subscriber. Various levels of the service can be offered, based upon the customers selection.
Work-at-home billing: A person who is working at home dials a 4digit feature access code, which prompts the system to track and
record the billing information for the calls. Calls tracked in this manner are billed directly to the company rather than to the individual.
Inbound call restriction: This service allows a customer to restrict certain calls from coming into the subscribers location. This service is
flexible enough to restrict calls either by area code, NNX, or particular
telephone numbers. Restrictions may even be specified by day of
week or time of day.
Outbound call restriction: This service allows a customer to restrict
certain calls from being completed from the subscribers location. This
service is flexible enough to restrict calls by either area code, NNX, or
particular telephone numbers. Restrictions may even be specific to day
of week or time of day.
Flexible hotline: Allows a customer to pick up a telephone handset and
automatically connect to a merchant without dialing any digits. An
example of this is a rent-a-car phone in an airport, which allows a customer to notify the rent-a-car company to pick them up at the terminal.
This list of services is only a sample of the potential that IN technology offers the industry. Everyone benefits in the following ways:
1. The network provider reuses embedded plant.
2. The service provider provides services to the marketplace faster.
3. The customer customizes their services to meet their particular needs.

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12. SELF-TEST
Multiple Choice
1.

One of the major advances that results from implementing intelligent networks is
that
.
a. Computers no longer have control over the network.
b. Wide-area service outages are much less likely.
c. Services can be adapted to user needs more efficiently.
d. Service providers are now limited to a single equipment provider.

2.

At each trigger detection point (TDP), the AIN component that checks for active
triggers is the
.
a. SSP
b. SCP
c. STP
d. PCP

3.

There are three types of triggers. They are

a. subscribed, group based, and trunk


b. subscribed, trunk, and office based
c. subscribed, group based, and office based
d. trunk, group based, and office based
4.

A service control point (SCP) is

a. a switch that is controlled by the intelligent network


b. a database for various IN services
c. a series of triggers that initiate controller actions
d. an instant in time at which the network selects a service to initiate

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5.

The development of the intelligent network places increased emphasis on a


consistent
.
a. switch model
b. network model
c. intelligence model
d. call model

6.

To say that a network is intelligent means that it is

a. able to make decisions on its own


b. controlled by software on distributed computers
c. able to bridge the gap between user and supplier needs
d. controlled by microchips that are easily interchanged
True or False
7.

Trigger check point (TCP) is a more recent term for trigger detection point.
a. true
b. false

8.

When a switching system checks for overload conditions before communicating


with the SCP, the process is called trigger delay.
a. true
b. false

9.

The distinction between the originating side and the terminating side of a service
switching point (SSP) means that both originating and terminating triggers could
influence the same call.
a. true
b. false

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10. The major functions of the service switching point (SSP) are to provide network
access, perform necessary switching, and allow access to AIN capabilities.
a. true
b. false
11. The call model is a generic representation of the call-processing activities necessary
to establish, maintain, and clear a basic call.
a. true
b. false
12. The so-called building-block approach to creating AIN services is based on the
AINs ability to add groups (or blocks) of users to the network without interruption
of service.
a. true
b. false

13. ACRONYM GUIDE


AIN

advanced intelligent network

AMP

AIN maintenance parameter

API

applications programming interface

ASE

application service elements

BCSM basic call state model


BRI

basic rate interface

BSTP

broadband signaling transfer point

CCM

call control module

CFM

call failure message

CSM

call segment model

DAA

directory assistance automation

DAP

data access point

DCN

data communications network

DP

detection point

EDP

event detection point

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EML

element management layer

ETC

event trapping capability

FE

functional entity

GDI

generic data interface

IF

information flow

IN

intelligent network

INAP

intelligent network application protocol

IN/1

intelligent network 1

IP

intelligent peripheral

IPC

intelligent peripheral controller

IPI

intelligent peripheral interface

ISCP

integrated service control point

LIDB

line information database

LNP

local number portability

MP

mediation point

MSC

message sequence chart

NAP

network access point

NCAS noncall associated signaling


NCP

network control point

NE

network element

NEL

next-event list

NML

network management layer

NNI

network-to-network interface

OBCM originating basic call model


ONA

open network architecture

OOP

object-oriented programming

OPC

originating point code

PCS

personal communications service


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PIC

points in call

PP

physical plane

RDC

routing determination check

RE

resource element

RVT

routing verification test

SCE

service creation environment

SCMS service creation and maintenance system


SCP

service control point

SDP

service data point

SIBB

service-independent building block

SLE

service logic editor

SLEE

service logic execution environment

SLI

service logic interpreter

SLL

service logic language

SLP

service logic program

SM

session management

SMS

service management system

SN

service node

SOP

service order provisioning

SP

service plane

SSP

service switching point

STP

signaling transfer point

TBCM terminating basic call model


TCP

trigger check point

TCP

test call parameter

TDP

trigger detection point

TSC

trigger status capability

WIN

wireless intelligent network

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TUTORIAL OVERVIEW
Internet service is expanding rapidly. The demands it has placed upon the
public network, especially the access network, are great. But technological
advances promise big increases in access speeds, enabling public networks to
play a major role in delivering new and improved telecommunications services and applications to consumers.
In todays environment, twisted-pair access using voice-band data modems is the
norm for residential and small business users. With new technologies, speeds of
500 kbps, 1.5 Mbps, and even 10 Mbps are promised for residential users.
How will access systems evolve to ubiquitously provide such capabilities?
This tutorial describes existing and emerging access technologies used to provide Internet access for residential and business applications.

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TOPICS
1.

WHERE IS ACCESS IN THE NETWORK? ............................................ 163

2.

ACCESS EVOLUTION DRIVERS ..........................................................164

3.

AN OVERVIEW OF ACCESS ALTERNATIVES ....................................165

4.

TWISTED-PAIR SOLUTIONS ................................................................166

5.

TWISTED-PAIRREMOTE ......................................................................169

6.

FIBER AND COAX ..................................................................................170

7.

WIRELESS ................................................................................................172

8.

ALL FIBERPON ....................................................................................174

9.

ALL FIBERSONET ................................................................................175

10. SELF-TEST ................................................................................................177


11. ACRONYM GUIDE ................................................................................180

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1. WHERE IS ACCESS IN THE NETWORK?


Internet access has different meanings to different people. In this section, the
concept of access will be explained for consistency within this tutorial. The
three major entities are the following:

Figure 1. Three Major Entities in the Provision of Internet Service:


1. end users who want to have Internet service as well as other services such
as telephony or CATV
2. telephony, wireless, and cable-service providers who want to provide connectivity between end users and data-service providers
3. data-service providers who want to supply Internet access, content services
(like AOL), or other data services, such as virtual private networking
Note that at times, the last two entities are combined. For example, several
companies such as MCI, AT&T, or Pacific Bell provide both telephony and
Internet access.
Now to the question, What is access? To the telephony, wireless, or cableservice provider, access is the network connection from the end users home
or business to the outside-plant termination point within the service node. In
traditional telephony architecture, this is most commonly thought of as the
twisted-pair cross-connect point and is referred to as the main distribution
frame (MDF). The remainder of the telephony-providers network would be
referred to as the switching and transport network.

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The Basics of Telecommunications


In contradistinction, an ISP views access as the connection from its customer
to its network. Here, access is the connection from the end users home or
business to the gateway-access node belonging to the ISP. Thus, to the ISP, the
telephonys access, switching, and transport network is all part of access.
Within this tutorial, the telephony, wireless, and cable-service providers view
of access is the focus of our discussion of new and emerging technologies.

2. ACCESS EVOLUTION DRIVERS


Access evolution is being driven primarily by strong demands for increasing
bandwidth to support a growing variety of user services. Prior to 1994, traffic
sent over the Internet was largely text-based information with file transfer
and e-mail being among the most popular services. The surge in growth of
the Internet during 1995 was in part due to the graphical nature of the World
Wide Web (WWW). A significant aspect of this shift is that graphical images
generally consist of a large number of bits. To transfer large graphical image
files quickly with satisfactory performance meant that higher-speed access
technologies were needed than those used to deliver relatively small text files.
The WWW also became the base for nurturing other capabilities such as animated graphics, audio, and low-rate video. Each of these capabilities have
been pushing the need for increasingly higher-speed access.

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Figure 2. An example of the data rates needed to support various user services and the access rates that have become available over time. (The chart
represents average user rates, not peak burst rates on shared media.) The
chart shows curves for three segments of the user population: the median,
the upper 20th percentile, and the upper 2nd percentile early adopters.
Users are eager for audio and video services, so the challenge is for access
systems to meet that demand.

3. AN OVERVIEW OF ACCESS ALTERNATIVES


Internet-access technologies fit into four broad categories:
1. twisted-pair
2. fiber/coax
3. wireless
4. all fiber
As shown in Figure 3, several technologies and implementations exist within
each of these broad categories.

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Figure 3. Access Alternatives


Twisted-pair telephone line is the access medium used by the vast majority of
individual residential subscribers today. Over time, a number of technologies
have been introduced to provide faster data speeds over this medium.
Fiber/coax systems were originally introduced for video-broadcast applications. Because these systems are inherently broadband, techniques have been
developed to use this advantage to provide high-speed data transmission,
principally for residential Internet access.
Wireless Internet access has two origins: satellite systems established for
broadcast video that have the ability to distribute Internet data at high
speeds, and cellular/PCS systems designed to serve mobile users.
The predominant access systems for business users are optical-fiber SONET
and SDH systems. In the future, passive optical network systems are expected to become an all-fiber access medium for residential users as well.

4. TWISTED-PAIR SOLUTIONS
There are three major categories of twisted-pair solutions that are being used
for Internet access (see Figure 4):

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1. voice band data (VBD) modems
2. ISDN digital subscriber line (DSL)
3. other DSL approaches (xDSL)

Figure 4. Twisted Pair Access


Voice Band Data and ISDN
VBD modems are well known and understood by residential and small-business users. They operate by using the voice-frequency band of the twistedpair facility to transmit data, using FSK or QAM transmission techniques.
Symmetric rates exist up to 33.6 kbps, with the majority running at 14.4 and
28.8 kbps. Emerging is an asymmetric capability with a nominal server rate of
56 kbps and return-path rate operating up to 33.6 kbps.
ISDN is a digital baseband technology that operates with a 144kbps bidirectional payload rate using 2B1Q encoding scheme. The 144kbps rate is divided into two 64kbps (B) channels and one 16kbps (D) channel. The B channels can be used for two separate voice calls, two 64kbps data calls, a separate voice and data call, or a combined 128kbps data call. The wire limit for
ISDN is 18,000 feet on standard twisted-pair.
xDSL Technologies
A variety of xDSL rates and technologies have been standardized, or are in
the process of standardization, by ANSI and the ADSL Forum. As Figure 5
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The Basics of Telecommunications


shows, the higher rates are for customers that are a short distance away from
the network providers xDSL modem. This modem may be located either in a
central office or at a remote terminal site closer to many end users.

50 Mbps

40 Mbps

30 Mbps

20 Mbps

10 Mbps

Figure 5. xDSL: Downstream Rates


ADSL
Asymmetric DSL (ADSL) is one of several types of xDSL technologies. ADSL
has two main standards: ADSL1 specifies a downstream rate of 1.5 or 2
Mbps and an upstream rate of 16 to 64 kbps; ADSL3 specifies a downstream
rate of up to 6.144 Mbps and a bidirectional channel of up to 640 kbps.
Good twisted-pair lines with no bridged taps can support ADSL1 rates up to
18,000 feet (24-gauge wire) and ADSL3 up to 12,000 feet.
ANSI and the ADSL Forum have endorsed discrete multitone technology
(DMT). However, carrierless amplitude and phase (CAP) technology has the
most market share thus far, with thirty times as many ADSL lines using CAP.
DMT and CAP modems are incompatible, but the issue is not nearly as great
as with voiceband data modems. VBD modems must be compatible end-toend, from end user to end user. But ADSL modems only operate over the end
users twisted-pair from end user to the network provider.

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VDSL
Very-high-speed DSL (VDSL) promises even higher speeds than ADSL,
although over much shorter distances. Standardization is underway in four
different standards bodies: ANSI, the ADSL Forum, the ATM Forum, and the
digital audio-visual council (DAVIC). There are four different technologies
proposed (CAP, DMT, DWMT, and SLC), aiming at a goal of lower power
and less cost than ADSL.
RADSL
As the name implies, rate-adaptive DSL (RADSL) modems adjust the data rate
to match the quality of the twisted-pair connection. Emerging software
should make this an automated process with little human intervention.
HDSL and SDSL
High-data-rate DSL (HDSL) modems transmit 1.5 Mbps in each direction. Two
twisted-pairs of wires are used, with half of the traffic on each pair. A 2Mbps
transmission rate is also available, using three pairs of wires (1/3 of the traffic
on each pair). The wire limit is 12,000 feet (24 ga.) or 9,000 feet (26 ga.)
SDSL refers to single-pair DSL or symmetric DSL. SDSL is similar to
HDSL, but requires only one pair of wires. Transmission speed ranges from n
x 64 kbps to 2.0 Mbps in both directions.
HDSL and SDSL are intended as lower-cost replacements for dedicated T1
and fractionalT1 lines, rather than for residential access.

5. TWISTED-PAIRREMOTE
The three major twisted-pair categories discussed under Twisted-Pair
Solutions can all be remoted as is shown in Figure 6.

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Figure 6. Twisted Pair AccessRemote


Remoting mechanisms, such as digital loop-carrier systems, enable fiber
migration into loop-access plant. They are a cost-effective way of bringing
service to end users who are not located near access nodes. As shown in
Figure 6, services are ultimately provided to end users over twisted-pairs from
remote terminals that connect via fiber facilities to the serving node.
As the figure illustrates, remote systems can be terminated in two ways. One
is a termination directly into a circuit switch; this is called the integrated
access approach. The second approach has a host digital-terminal termination
in the service node; this is called the universal access approach. Both integrated and universal remote-access arrangements are used to provide Internet
access. The choice of approach in any specific case depends on the embedded
network and on the capabilities that must be provided to end users in addition to Internet access.

6. FIBER AND COAX


There are three significant categories of combined cable and fiber systems
used for Internet access. With reference to Figure 7, they are as follows:

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Figure 7. Fiber and Coax


Cable TV Hybrid Fiber/Coax (HFC) System (Top)
Traditional systems have only downstream broadcast capability. These traditional cable-TV (CATV) systems broadcast downstream in the 50 to 550/750
MHz band with 6 MHz channels.
Cable modems are used to allow Internet and data transmission in the downstream direction of the HFC system. Internet data speeds up to the 30-Mbps
range can be realized in a nominal 6 MHz video channel. The upstream signal
is provided by an existing telephone channel using VBD or ISDN.

Bidirectional HFC System (Middle)


These newer systems have transmission capability in both directions. Such
bidirectional CATV systems typically broadcast downstream in the 50 to 750
MHz bandwidth of the coax within the 6 MHz nominal video channels. The
upstream bandwidth is shared among all the homes passed by the coax cable
and is nominally limited to the 5 to 40 MHz frequency band.
Downstream Internet data speeds up to the 30Mbps range in a 6 MHz channel can be realized. Upstream data is contention based and operates at
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claimed speeds of up to 10 Mbps. In practical multiuser environments,
though, actual throughput speeds will be significantly less.
Cable modems can either be overlaid onto the HFC system or be an integrated part of the HFC system.
Switched Digital Broadband (SDB) Systems (Bottom)
SDB is classified as a baseband digital system with nominal 50Mbps pointto-point downstream rates that can be apportioned as desired between digital
video and data. For data, a 1.5Mbps nominal contention-based upstream
data bandwidth is available. Though the system is contention based, there is
always a minimum guaranteed upstream data rate availabletypically, in the
order of 16 kbps.
The three architectures described all have provisions for both analog and digital video broadcast capability.
Both the bidirectional HFC and SDB systems are broadband systems that are
applicable to telephony, video, Internet/data, and PCS wireline access. Note
that the architectures have a number of similar characteristics and components. The bidirectional HFC system provides fiber distribution to the fiber
node. At the fiber node, signals are collected and distributed to multiple-coax
feeds that cover a given residential area. Fiber nodes are designed to serve
from 500 to 2000 homes.
SDB systems push fiber closer to the end user. In typical systems, feeder fiber
can be optically split. Optical network units (ONU) terminate the fiber and provide individual coax (and twisted-pair) drops to subscribers. A typical ONU can
serve from 4 to 60 homes. Thus, SDB provides fiber closer to the customer.
In many ways, HFC, SDB, and PON (discussed shortly) can be viewed as a continuum of technology where fiber moves ever closer to the customer premises.

7. WIRELESS
As Figure 8 illustrates, three means are used to provide Internet access using
wireless technology: satellite broadcast, terrestrial broadcast, and cellular/PCS.

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Cellular/
PCS

Figure 8. Wireless Access


Cellular/PCS
Internet access can be provided via existing cellular systems using voiceband
modems. Because cellular channels are narrowband, access rates are limited to
9.6 kbps for AMPS and TDMA systems and to 14.4 kbps for CDMA systems.
Cellular digital packet data (CDPD) is a technique that enables the data rate
of AMPS to be extended to 19.2 kbps. CDPD achieves the higher rate by
inserting IP packets directly into cellular channels that do not contain voice
traffic (i.e., channels that are temporarily idle).
Techniques are being investigated to provide Internet access and other data
services using personal communications services (PCS). PCS data standards
are being investigated by a joint technical committee of ANSI T1 and the
Telecommunications Industry Association (TIA), ITUT, and others.
Terrestrial Broadcast
The multichannel multipoint distribution service (MMDS), sometimes called
wireless cable, can provide Internet-access downlinks over a distance of
about 50 km from a central-transmitter site. MMDS downlinks combined
with telephony uplinks provide a complete Internet-access arrangement.

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MMDS operates in the 2 GHz frequency band with 33 channels, each capable of
supporting downlink data rates that are currently about 10 Mbps. Technology
improvements are expected to increase data rates to 27 Mbps in the future.
Local multipoint distribution service (LMDS) is similar to MMDS in that it will
use microwave transmission to provide Internet-access downlinks and wireline
telephony to provide uplink access. LMDS will use transmitters operating in
the 28 GHz frequency band, with each transmitter covering a distance of
about 5 km. The relatively close transmitter spacing, coupled with the fact that
LMDS will have about four times the bandwidth of MMDS, should enable
LMDS to serve a much higher density of Internet users than MMDS.
Satellite Broadcast
Several approaches have been proposed for using satellites to provide
Internet-access downlinks. Some proposals are based on using a single fixedposition satellite whereas others would use clusters of satellites. Proposed
data rates vary from low-speed, single-user channels to shared channels with
rates greater than 1 Mbps.
The first widely available system operates in the 12 GHz band and uses a
data rate of 400 kbps. Equipment at the end-user location consists of a dish
antenna, approximately 52 cm in diameter, a microwave receiver, and a digital decoder card that plugs directly into a PC computer bus. Satellite systems
also use telephony circuits for uplink access.

8. ALL FIBERPON
This all-fiber access system called a passive optical network (PON) is intended
for residential applications for Internet and other services access. The architecture shown at the top of Figure 9 has fiber from the service node to the optical
splitter. At the splitter, multiple fibers fan out to terminate on a single-home
ONU. The ONU then splits out to provide individual service to the home. There
are many schemes for technical realization of PONs. One of the more interesting
ones consists of wave division multiplexing (WDM) for up to 16 ONU drops
from the optical splitter. WDM techniques would again be used in the upstream
direction to realize a highly secure point-to-point PON architecture.

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Figure 9. Pure Fiber AccessPON


As it is all fiber, PON has many advantages. All fiber yields a robust outside
plant that has low maintenance costs associated with it. All-fiber point-to-point
architecture allows for secure transmissions and broadband service applications.
The PON architecture represents the target wireline architecture because of
its versatility and evolution-proof capabilities. But initial costs of PON systems are still higher than most all of the other alternatives discussed within
this tutorial.

9. ALL FIBERSONET
All-fiber access systems consisting of SONET or SDH fiber rings are commonly used to provide high-capacity, multiservices (including Internet) access
from and to campus and business locations. Synchronous optical network
(SONET), is a North Americanbased standard for such an architecture, with
interface rates from 1.544 Mbps (DS1) to 10 Gbps (OC192). Similarly, synchronous digital hierarchy (SDH), is the European-based standard for equipment with similar capabilities.
The bottom portion of Figure 10 shows a SONET/SDH connection from a service node to a business. The SONET/SDH ring provides service assurance via

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path-diversity ring architecture. In a campus environment, the ring could be
extended through the various buildings that make up the campus.

Figure 10. Pure Fiber AccessSONET


On the customer premises, the customers intranet, depicted by the LAN, is
connected to the public network via a firewall. The firewall provides data and
security protection for the business. Firewalls provide security by isolating
undesired Internet traffic from that traffic, which is carried on intranet LANs.
Multiplexers (mux) provide transport efficiency by combining separate datastreams onto a single fiber-optic facility. Synchronous transfer mode (STM)
multiplexers widely deployed in telecommunications networks carry datastreams within discrete tributaries. Asynchronous transfer mode (ATM) multiplexers can provide more interface-rate granularity because individual user
datastreams are concatenated into a single high-speed cell stream for transport
within the network. In addition, ATM multiplexers can provide further efficiency by combining variable-rate datastreams using statistical multiplexing.

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10. SELF-TEST
Match Access Technology to Access Type:
1. Hybrid fiber/coax (HFC):
a. Twisted-pair
b. Wireless
c. Coax and fiber
d. All fiber
2. Local multipoint distribution service (LMDS):
a. Twisted-pair
b. Wireless
c. Coax and fiber
d. All fiber
3. Synchronous optical network (SONET):
a. Twisted-pair
b. Wireless
c. Coax and fiber
d. All fiber

4. Integrated services digital network (ISDN):


a. Twisted-pair
b. Wireless
c. Coax and fiber
d. All fiber

True or False
5. Fiber/coax systems are the access media used by the vast majority of individual
residential users.
a. true
b. false

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6. The earliest Internet traffic was primarily text-based traffic.
a. true
b. false
7. Wireless Internet has two origins: satellite systems and cellular.
a. true
b. false
8. The predominant access system for business users is twisted pair.
a. true
b. false
9. Currently, VBD modems run primarily at 144 kbps.
a. true
b. false
10. XDSL modems offer higher rates but only for customers who are relatively close to
a providers xDSL modem.
a. true
b. false
Multiple Choice
11. ISDN operates with

a. 33.6 kbps
b. 16 kbps
c. 14.4 kbps
d. 128 kbps
12. How many main standards does ADSL have?
a. 1
b. 2
c. 3
d. 4

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13. How far can good twisted-pair lines support ADSL1 rates?
a. 6,000 ft.
b. 12,000 ft.
c. 18,000 ft.
d. 24,000 ft.
14. Currently, PCS

a. is capable of providing Internet access


b. is not capable of providing Internet access
15. Internet access via microwave transmission
a. is provided by MMDS and LMDS
b. is not provided by MMDS and LMDS
16. The initial costs associated with the deployment of the passive optical
network
.
a. are high
b. are low
17. PON systems are primarily designed

a. to provide residential services


b. to provide corporate services
18. The European-based standard for all-fiber access systems is

a. synchronous digital hierarchy (SDH)


b. asynchronous digital hierarchy (ADH)
19. Which transfer mode multiplexers combine individual datastreams into a single
high-speed cell stream?
a. synchronous
b. asynchronous
20. Which transfer mode multiplexers carry datastreams within discrete
tributaries?
a. synchronous
b. asynchronous

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11. ACRONYM GUIDE


2B1Q

2 binary 1 quaternary

ADSL

asymmetric digital subscriber line

AMPS

advanced mobile phone service

AN

access node

ANSI

American National Standards Institute

ATM

asynchronous transfer mode

CAP

carrierless amplitude and phase modulation

CDMA

code division multiple access

CDPD

cellular digital packet data

DMT

discrete multitone

DSL

digital subscriber line

DWMT

discrete wavelet multitone

FSK

frequency shift keying

FTTC

fiber to the curb

HDSL

high-bit-rate digital subscriber line

HDT

host digital terminal

HFC

hybrid fiber/coax

ISDN

integrated services digital network

ISP

Internet service provider

LAN

local-area network

LMDS

local multipoint distribution service

MDF

main distribution frame

MMDS

multichannel multipoint distribution service

NID

network interface device

NIU

network interface unit

ONU

optical network unit

PCS

personal communications services

PON

passive optical network

QAM

quadrature amplitude modulation

RADSL

rate adaptive digital subscriber line

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S

splitter

SDB

switched digital broadband

SDH

synchronous digital hierarchy

SDSL

single-pair digital subscriber line or symmetric digital


subscriber line

SLC

simple line code

SMDS

switched multimegabit data service

SONET

synchronous optical network

STB

set-top box

STM

synchronous transfer mode

TDMA

time division multiple access

TIA

telecommunications industry association

VBD

voiceband data

VDSL

very-high-speed digital subscriber line

WWW

World Wide Web

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Internet Telephony
DEFINITION
Internet telephony refers to communications servicesvoice, facsimile,
and/or voice-messaging applicationsthat are transported via the Internet,
rather than the public switched telephone network (PSTN). The basic steps
involved in originating an Internet telephone call are conversion of the analog
voice signal to digital format and compression/translation of the signal into
Internet protocol (IP) packets for transmission over the Internet; the process is
reversed at the receiving end.

TUTORIAL OVERVIEW
This tutorial discusses the ongoing but rapid evolution of Internet telephony,
the market forces fueling that evolution, and the benefits that users can realize, as well as the underlying technologies. It also examines the hurdles that
must be overcome before Internet telephony can be adopted on a widespread
basis.

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TOPICS
1. INTRODUCTION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .185
2. INTRANET TELEPHONY PAVES THE WAY FOR INTERNET
TELEPHONY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .189
3. TECHNICAL BARRIERS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .191
4. STANDARDS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .192
5. FUTURE OF VOICE-OVER-INTERNET-PROTOCOL (VOIP)
TELEPHONY . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .194
6. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .196
7. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .198

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1. INTRODUCTION
The possibility of voice communications traveling over the Internet, rather
than the public switched telephone network (PSTN), first became a reality in
February 1995 when Vocaltec, Inc. introduced its Internet phone software.
Designed to run on a 486/33-MHz (or higher) PC equipped with a sound
card, speakers, microphone, and modem (see Figure 1), the software compresses the voice signal and translates it into IP packets for transmission over
the Internet. This PCtoPC Internet telephony works, however, only if both
parties are using Internet phone software.

Figure 1: PC Configuration for VoIP

In the relatively short period of time since then, Internet telephony has
advanced rapidly. Many software developers now offer PC telephony software but, more importantly, gateway servers are emerging to act as an interface between the Internet and the PSTN (see Figure 2). Equipped with voiceprocessing cards, these gateway servers enable users to communicate via
standard telephones.

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Figure 2: Gateway Server

Figure 3: Topology of PC-to-Phone

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A call goes over the local PSTN network to the nearest gateway server, which
digitizes the analog voice signal, compresses it into IP packets, and moves it
onto the Internet for transport to a gateway at the receiving end (see Figure 4).
With its support for computer-to-telephone calls, telephone-to-computer calls,
and telephone-to-telephone calls, Internet telephony represents a significant
step toward the integration of voice and data networks.

Figure 4: Sequence of VoIP Connection


Originally regarded as a novelty, Internet telephony is attracting more and
more users because it offers tremendous cost savings relative to the PSTN.
Users can bypass long-distance carriers and their per-minute usage rates and
run their voice traffic over the Internet for a flat monthly Internet-access fee.

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Figure 5: Phone-to-Phone Connection

Figure 6: PC-to-Phone Connection

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2. INTRANET TELEPHONY PAVES THE WAY FOR INTERNET


TELEPHONY
Although progressing rapidly, Internet telephony still has some problems with
reliability and sound quality, due primarily to limitations both in Internet
bandwidth and current compression technology. As a result, most corporations looking to reduce their phone bills today confine their Internet-telephony applications to their intranets. With more predictable bandwidth available
than the public Internet, intranets can support full-duplex, real-time voice
communications. Corporations generally limit their Internet voice traffic to
half-duplex asynchronous applications (e.g., voice messaging).
Internet telephony within an intranet enables users to save on long-distance
bills between sites; they can make point-to-point calls via gateway servers
attached to the LAN. No PCbased telephony software or Internet account is
required.
For example, User A in New York wants to make a (point-to-point) phone call
to User B in the companys Geneva office. He picks up the phone and dials an
extension to connect with the gateway server, which is equipped with a telephony board and compression-conversion software; the server configures the
PBX to digitize the upcoming call. User A then dials the number of the
London office, and the gateway server transmits the (digitized, IPpacketized)
call over the IPbased WAN to the gateway at the Geneva end. The Geneva
gateway converts the digital signal back to analog format and delivers it to
the called party.

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Figure 7: Telephone Gateway Connections

Figure 8: Internet Telephony Gateway

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This version of Internet telephony also enables companies to transmit their
(digitized) voice and data traffic together over the intranet in support of
shared applications and whiteboarding.

3. TECHNICAL BARRIERS
The ultimate objective of Internet telephony is, of course, reliable, high-quality voice service, the kind that users expect from the PSTN. At the moment,
however, that level of reliability and sound quality is not available on the
Internet, primarily because of bandwidth limitations that lead to packet loss.
In voice communications, packet loss shows up in the form of gaps or periods
of silence in the conversation, leading to a clipped-speech effect that is
unsatisfactory for most users and unacceptable in business communications.

Figure 9: Internet Telephony


The Internet, a collection of more than 130,000 networks, is gaining in popularity as millions of new users sign on every month. The increasingly heavy
use of the Internets limited bandwidth often results in congestion which, in
turn, can cause delays in packet transmission. Such network delays mean
packets are lost or discarded.

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In addition, because the Internet is a packet-switched or connectionless network, the individual packets of each voice signal travel over separate network
paths for reassembly in the proper sequence at their ultimate destination.
While this makes for a more efficient use of network resources than the circuit-switched PSTN, which routes a call over a single path, it also increases
the chances for packet loss.
Network reliability and sound quality also are functions of the voice-encoding
techniques and associated voice-processing functions of the gateway servers.
To date, most developers of Internet-telephony software, as well as vendors of
gateway servers, have been using a variety of speech-compression protocols.
The use of various speech-coding algorithmswith their different bit rates
and mechanisms for reconstructing voice packets and handling delaysproduces varying levels of intelligibility and fidelity in sound transmitted over the
Internet. The lack of standardized protocols also means that many Internettelephony products do not interoperate with each other or with the PSTN.

4. STANDARDS
Over the next few years, the industry will address the bandwidth limitations
by upgrading the Internet backbone to asynchronous transfer mode (ATM),
the switching fabric designed to handle voice, data, and video traffic. Such
network optimization will go a long way toward eliminating network congestion and the associated packet loss. The Internet industry also is tackling the
problems of network reliability and sound quality on the Internet through the
gradual adoption of standards. Standards-setting efforts are focusing on the
three central elements of Internet telephony: the audio codec format; transport protocols; and directory services.
In May 1996, the International Telecommunications Union (ITU) ratified the
H.323 specification, which defines how voice, data, and video traffic will be
transported over IPbased local area networks; it also incorporates the T.120
data-conferencing standard (see Figure 10). The recommendation is based on
the real-time protocol/real-time control protocol (RTP/RTCP) for managing
audio and video signals.

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Figure 10: H.323 Call Sequence

As such, H.323 addresses the core Internet-telephony applications by defining


how delay-sensitive traffic (i.e., voice and video) gets priority transport to
ensure real-time communications service over the Internet. (The H.324 specification defines the transport of voice, data, and video over regular telephony
networks, while H.320 defines the protocols for transporting voice, data, and
video over ISDN networks.)
H.323 is a set of recommendations, one of which is G.729 for audio codecs,
which the ITU ratified in November 1995. Despite the ITU recommendation,
however, the Voice over IP (VoIP) Forum in March 1997 voted to recommend
the G.723.1 specification over the G.729 standard. The industry consortium,
which is led by Intel and Microsoft, agreed to sacrifice some sound quality for
the sake of greater bandwidth efficiencyG.723.1 requires 6.3 kbps, while
G.729 requires 7.9 kbps. Adoption of the audio codec standard, while an
important step, is expected to improve reliability and sound quality mostly
for intranet traffic and point-to-point IP connections. To achieve PSTNlike
quality, standards are required to guarantee Internet connections.

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The transport protocol RTP, on which the H.323 recommendation is based,
essentially is a new protocol layer for real-time applications; RTPcompliant
equipment will include control mechanisms for synchronizing different traffic
streams. However, RTP does not have any mechanisms for ensuring the ontime delivery of traffic signals or for recovering lost packets. RTP also does not
address the so-called quality of service (QoS) issue related to guaranteed
bandwidth availability for specific applications. Currently, there is a draft signaling-protocol standard aimed at strengthening the Internets ability to handle
real-time traffic reliably (i.e., to dedicate end-to-end transport paths for specific
sessions much like the circuit-switched PSTN does). If adopted, the resource
reservation protocol, or RSVP, will be implemented in routers to establish and
maintain requested transmission paths and quality-of-service levels.
Finally, there is a need for industry standards in the area of Internet-telephony
directory services. Directories are required to ensure interoperability between
the Internet and the PSTN, and most current Internet-telephony applications
involve proprietary implementations. However, the lightweight directory access
protocol (LDAP v3.0) seems to be emerging as the basis for a new standard.

5. FUTURE OF VOICE OVER INTERNET PROTOCOL (VOIP)


TELEPHONY
Several factors will influence future developments in VoIP products and services. Currently, the most promising areas for VoIP are corporate intranets
and commercial extranets. Their IPbased infrastructures enable operators to
control who canand cannotuse the network.
Another influential element in the ongoing Internet-telephony evolution is the
VoIP gateway. As these gateways evolve from PCbased platforms to robust
embedded systems, each will be able to handle hundreds of simultaneous calls.
Consequently, corporations will deploy large numbers of them in an effort to
reduce the expenses associated with high-volume voice, fax, and videoconferencing traffic. The economics of placing all trafficdata, voice, and video
over an IPbased network will pull companies in this direction, simply because
IP will act as a unifying agent, regardless of the underlying architecture (i.e.,
leased lines, frame relay, or ATM) of an organizations network.

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Commercial extranets, based on conservatively engineered IP networks, will
deliver VoIP and Fax-overIP (FoIP) services to the general public. By guaranteeing specific parameters, such as packet delay, packet jitter, and service interoperability, these extranets will ensure reliable network support for such applications.
VoIP products and services transported via the public Internet will be niche
markets that can tolerate the varying performance levels of that transport
medium. Telecommunications carriers most likely will rely on the public
Internet to provide telephone service between/among geographic locations
that today are high-tariff areas. It is unlikely that the public Internets performance characteristics will improve sufficiently within the next two years to
stimulate significant growth in VoIP for that medium.
However, the public Internet will be able to handle voice and video services
quite reliably within the next three to five years, once two critical changes
take place:
an increase by several orders of magnitude in backbone bandwidth and
access speeds, stemming from the deployment of IP/ATM/SONET and
ISDN, cable modems, and xDSL technologies, respectively
the tiering of the public Internet, in which users will be required to pay
for the specific service levels they require
On the other hand, FoIP products and services via the public Internet will
become economically viable more quickly than voice and video, primarily
because the technical roadblocks are less challenging. Within two years, corporations will take their fax traffic off the PSTN and move it quickly to the
public Internet and corporate intranet, first through FoIP gateways and then
via IPcapable fax machines. Standards for IPbased fax transmission will be
in place by the end of this year.
Throughout the remainder of this decade, videoconferencing (H.323) with
data collaboration (T.120) will become the normal method of corporate communications, as network performance and interoperability increase and business organizations appreciate the economics of telecommuting. Soon, the
video camera will be a standard piece of computer hardware for full-featured
multimedia systems, as well as for the less-than-$500 network-computer
appliances now starting to appear in the market. The latter in particular
should stimulate the residential demand and bring VoIP services to the mass
marketincluding the roughly 60 percent of American households that still
do not have a PC.
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6. SELF-TEST
True or False
1. The first Internet-telephony software, Internet Phone, supported PCtoPC and
telephone-to-telephone voice calls via the Internet.
a. true
b. false
2. The current reliability and sound-quality problems of Internet telephony are attributable to limitations in Internet bandwidth and compression technology.
a. true
b. false
3. As a packet-switched or connectionless network, the Internet decreases the chances
of packet loss for a voice call.
a. true
b. false
4. To date, most developers of Internet-telephony software and vendors of gateway
servers have used the same speech-compression protocols.
a. true
b. false
5. The International Telecommunications Union (ITU) has ratified a standard for
voice, data, and video transmission over IPbased local-area networks.
a. true
b. false
Multiple Choice
6. ITUs H.320 standard defines the protocols for transporting voice, data and video
over
.
a. the public switched telephone network (PSTN)
b. ISDN networks
c. the public Internet
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7. The G.723.1 specification for audio codecs, recommended by the VoIP Forum,
requires
.
a. 6.3 kbps
b. 7.9 kbps
c. 8.4 kbps
8. Internet-telephony directories enable

a. users to determine other users Internet addresses


b. users to determine whether an Internet site is capable of
receiving Internet-telephony transmissions
c. Internet/PSTN interoperability
9. In the near term, the market segment expected to be the biggest driver for VoIP telephony is
.
a. small-office home-office (SOHO) customers
b. military and government networks
c. corporate intranets and extranets
10. The public Internet will be able to transport voice calls reliably and with high quality when
.
a. standards are established for Internet directories
b. manufacturers produce higher-quality, lower-cost audiocodec technology
c. various technologies deliver greater backbone-network and
subscriber-access speeds

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7. ACRONYM GUIDE
API

application programming interface

ATM

asynchronous transfer mode

CATV community access television


CPE

customer-premises equipment

xDSL

digital subscriber line (e.g., x = a for asymmetric, x = h for high bit-rate)

DSP

digital signal processor

FoIP

facsimile over Internet protocol

IP

Internet protocol

ISDN

integrated services digital network

ISP

Internet service provider

ITU

International Telecommunications Union

LAN

local-area network

LDAP

lightweight directory access protocol

MHz

Megahertz

PBX

private branch exchange

PSTN

public switched telephone network

QoS

quality of service

RTCP

real-time protocol

RTP

real-time control protocol

SONET synchronous optical network


VoIP

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Intranets and Virtual Private


Networks (VPNs)
DEFINITION
Private networking involves securely transmitting corporate data across multiple sites throughout an entire enterprise. Creating a truly private corporate
network generally requires an intranet. A virtual private network (VPN) is one
means of accomplishing such an implementation using the public Internet.

TUTORIAL OVERVIEW
This tutorial explores the benefits of a private corporate network and reviews
a traditional wide-area network (WAN) architecture implementation. It then
compares the WAN model to present-day private-networking strategies,
specifically examining two types of modern private-network implementations: encryption-based VPNs and private networks based on frame-relay permanent virtual circuits (PVCs). It also reviews important security issues associated with the different technologies used to implement a private network.

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TOPICS
1. INTRODUCTION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .201
2. BENEFITS OF ISPBASED PRIVATE NETWORKS . . . . . . . . . . . . . . . .201
3. TRADITIONAL WAN NETWORK ARCHITECTURE . . . . . . . . . . . . .202
4. ENCRYPTION-BASED VPNS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .203
5. PRIVATE NETWORKING USING FRAME-RELAY PVCS . . . . . . . . . . .204
6. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .206
7. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .209

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1. INTRODUCTION
Today's corporations are challenged by the need to support a wide variety of
communications across a highly distributed number of sites and offices. At
the same time that the number of sites and offices increases, corporations are
pressured to reduce the cost of their overall communications expenses. In
addition to the increased number of office locations, employees expect to
access corporate resources from a more diverse set of locations, including customer sites, home offices, and travel destinations. As more emphasis is placed
on electronic communication, business partners also expect to access corporate-partner data as well. All of these trends drive the need to establish a corporate private-network infrastructure.
With regard to communications expenses, however, corporations are finding
that traditional architecture does not provide the flexibility and solutions
required. Using dedicated leased-line circuits to interconnect main offices and
branch offices often requires significant planning time, and once in place the
circuits cannot support remote or customer sites. The increase in telecommuting and remote computing is, in turn, increasing resources spent on remoteaccess modems, servers, and long-distance telephone charges.
Private networks that utilize the Internet backbone can significantly reduce
the costs of establishing and maintaining a WAN for private-networking purposes. Internet service provider (ISP)based private networks offer a global
footprint with ubiquitous local network access. Using an ISPbased private
network, corporations can connect their offices to the ISP's local points of
presence (PoPs) rather than purchase costly leased-line circuits to interconnect
their office locations. The corporation takes advantage of the ISP's established
backbone, which is usually more geographically diverse than its WAN architecture. The ISP can also offer local dial-up access at a diverse number of locations, which helps reduce long-distance remote-access costs.

2. BENEFITS OF ISPBASED PRIVATE NETWORKS


ISPbased private networks can offer direct cost savings over traditional WAN
architectures as well as other indirect cost savings. The increased flexibility
and scalability of ISPbased private networks can often reduce equipment
costs while minimizing network management and technical-training
resources.

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The most significant benefit of an ISPbased private network is its direct cost
savings over traditional WANs. A traditional WAN is composed of distancesensitive leased-line circuits, which can be subject to interstate and international tariffs and taxes. In comparison, an ISPbased private network only
requires shorter leased-line circuits from each office to the ISP's closest PoP.
ISPs can also offer flexibility in line speeds; corporations can usually purchase
access in fractional tier-1 (T1) increments rather than in an entire T1 circuit
from a telco or local exchange carrier (LEC).
Outsourcing network management to an ISP can also indirectly reduce operating costs and resources. In-house technical resources are no longer needed
to install, configure, and manage network equipment. A corporation will not
need to support dial-up plain old telephone service (POTS) lines or integrated
services digital network (ISDN) and leased-line circuits. The information technology (IT) department can concentrate resources on data and server equipment rather than on low-level network equipment.
Several different ways to implement an ISPbased private network can be
used. One common implementation transmits corporate traffic over the public Internet but uses encryption to protect the data from unauthorized access.
Frame-relay technology also enables the creation of logically isolated circuits
or PVCs that provide a private network in which data does not need to be
encrypted because it travels only along these logically private circuits.

3. TRADITIONAL WAN NETWORK ARCHITECTURE


Traditional WANbased private networks used leased-line circuits from each
site back to a corporate headquarters. These leased-line circuits were priced
according to distance, making them expensive for geographically dispersed
locations. These traditional WANs often used a hub-and-spoke model, requiring traffic going between branch offices to travel through the corporate headquarters. Despite these disadvantages, traditional WANs did offer the highest
level of security and network performance. Corporations were paying for the
full-dedicated bandwidth of the network.

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Traditional WANs required highly specialized technical in-house resources
and required corporations to manage their own networks. Older corporations
often have extensive WAN infrastructure investments and are hesitant to
adopt newer infrastructure implementations despite large potential cost savings. Figure 1 illustrates a typical WAN implementation where all traffic is
cross-connected at the corporate headquarters. Each branch office or partner
company is connected directly to the headquarters location.

Figure 1. Traditional WANs

4. ENCRYPTION-BASED VPNS
Encryption-based VPNs create a VPN using the public Internet infrastructure.
A corporation establishes public Internet connections from each of its office
locations to an ISP's PoP. The corporation can establish the connections with
a single ISP or multiple ISPs.
Encryption-based VPNs are susceptible to any weaknesses that the public
Internet may experience. Typically, these weaknesses are related to data security and network performance. The original design and implementation of the
Internet did not address the security and performance requirements of private
networks.
Encryption-based VPNs are often the easiest type of ISPbased private network to create. Several different encryption vendors supply a large range of
solutions. Figure 2 shows a typical encryption-based VPN implementation.
Each branch office or partner company connects to any ISP; users simply
must have access to the public Internet. An encryption device (typically a

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router or firewall) is placed at each location. The encryption devices receive
encrypted data from the other locations and perform the appropriate decryption.

Figure 2. Encryption-Based VPNs

5. PRIVATE NETWORKING USING FRAME-RELAY PVCS


Another way to implement a private-networking solution while capitalizing
on an ISP's backbone is to create a private-network using frame-relay PVCs.
Frame-relay PVC is a technology available to homogeneous frame-relay networks; the ISP must be able to implement the frame relaynetworking protocol across its entire network.
A PVC is a way to logically create a separate independent circuit within the
same physical circuit. Figure 3 illustrates three separate PVCs within the same
physical interface. Each PVC acts logically as a private circuit, similar to a traditional WANdedicated circuit.

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Figure 3. Subinterface Model


Frame-relay PVCs offer the advantages of high security because sensitive corporate data is not transmitted to the public Internet. Instead it is only transmitted down a customer's own PVC, which remains logically separate from
the public Internet. A frame-relay PVC private-network implementation is
also not as susceptible to network congestion, as each PVC only carries data
for one customer.
Figure 4 illustrates a private network utilizing frame-relay PVCs. No additional
encryption devices are necessary. Each branch office or partner company is
connected to the homogeneous network of a selected ISP.

Figure 4. Frame-Relay PVC Networking Technology

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6. SELF-TEST
1. Private networking refers to which of the following?
a. different types of network firewalls
b. networking where network protocols are not used
c. securely transmitting corporate data
d. accessing Web servers using the HTTP protocol
2. A VPN is which of the following?
a. an implementation of a private network
b. a network built using frame-relay technology
c. a high-speed network protocol
d. a standard way to encrypt files for secure transmission
3. Corporate networks are now challenged because of which of the following?
a. computer equipment requires a greater amount of storage space
b. centrally located computers are consuming greater amounts of
electrical power
c. multiple protocols are taxing existing network resources
d. the need to support a wide variety of communications across
a large geographic area
4. Traditional WAN architecture ___________________.
a. is growing because it ideally meets corporate network needs
b. is a low-cost solution to a wide variety of needs
c. is less flexible and more costly to implement
d. improves data transmission over long distances
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5. Remote access modems require which of the following?
a. increased equipment and management resources
b. minimal equipment and network resources
c. no additional long-distance charges
d. that telecommuters use specific remote-access hardware to access
the internal network
6. ISPbased private networks _________________.
a. require entirely new server and internal-network equipment
b. are a cost-effective alternative to traditional WANs
c. increase management costs as a result of additional network
equipment
d. are more secure than traditional WANs
7. Compared to a single corporation, ISPs tend to have which of the following?
a. fewer remote-access dial-up ports
b. greater diversity in local remote-access points
c. minimal international remote-access capabilities
d. a smaller, more concentrated network backbone
8. Compared to a WAN, an ISPbased private network ___________________.
a. costs more due to the additional cost of a diverse network
backbone
b. offers significant cost savings
c. offers a greater level of security and network performance
d. requires a greater number of dedicated leased-line circuits

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9. Internal network management resources _____________________.
a. must be increased when using an ISPbased private network,
as a result of Internet security issues
b. are still required when using an ISPbased private network to
support network technologies including POTS, ISDN, and
leased-line circuits
c. can be decreased when using an ISPbased private network due
to the outsourcing of network management
d. never change regardless of private-network type

10. Traditional WANs are built using which of the following?


a. multiple dial-up links interconnecting each office location
b. short leased-line circuits connecting each office to its closest
neighbor
c. inexpensive wireless networktransmission technologies
d. expensive, distance-sensitive leased-line circuits

11. International WANs are which of the following?


a. extremely expensive as a result of distance-sensitive leased-line
circuits
b. impossible to build using traditional WAN technologies
c. inexpensive because corporations can share network costs with
other corporations
d. built using satellite technology, which limits network
performance

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12. Encryption-based VPNs ___________________.
a. use the public Internet infrastructure to create a VPN
b. use inexpensive hardware to encrypt data traveling across the
Internet
c. are more secure than traditional WANs due to data encryption
d. are not susceptible to Internet weaknesses and outages

13. Encryption-based VPNs _______________________.


a. require encryption hardware only at the primary headquarters
location
b. cannot accommodate dial-up remote-access users
c. require an encryption equipment at each location of the network
d. aggregate network traffic through a central location

7. ACRONYM GUIDE
ISDN

integrated services digital network

ISP

Internet service provider

IT

information technology

LEC

local exchange carrier

POP

point of presence

POTS

plain old telephone service

PVC

permanent virtual circuit

VPN

virtual private network

WAN

wide area network


209

Operations Support Systems (OSSs)


DEFINITION
The term operations support system (OSS) generally refers to the system (or systems) that performs management, inventory, engineering, planning, and repair
functions for communications service providers and their networks.

TUTORIAL OVERVIEW
Originally, OSSs were mainframe-based, stand-alone systems designed to
support telephone company staff members in their daily jobs. Essentially,
these systems were designed to automate manual processes, making operation of the network more error-free and efficient. Todays "next-generation
service providers" are required to manage a much more complex set of products and services in a dynamic, competitive marketplace. As a result, these
service providers need next-generation OSS solutions that take advantage of
state-of-the-art information technology to address their enterprise-wide needs
and requirements. Next-generation OSSs help service providers maximize
their return on investment (ROI) in one of their key assetsinformation.
OSSs ultimately help enable next-generation service providers to reduce costs,
provide superior customer service, and accelerate their time to market for
new products and services.

This tutorial focuses on the current and near-future states of OSS technology
and its development to support new and emerging services and technologies.
Note that the tutorial focuses only on the service management layer of the
Telecommunications Management Network (TMN) model. Refer to the Web
ProForum TMN tutorial for a complete discussion of this model.

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TOPICS
1.

THE BASICS OF OSSs ..............................................................................213

2.

OSS INTERCONNECTION ....................................................................218

3.

OPERATIONS SUPPORT OF DATA SERVICES ....................................222

4.

BUSINESS IMPACT OF AN OSS SOLUTION........................................225

5.

CONCLUSION..........................................................................................227

6.

SELF-TEST ................................................................................................228

7.

ACRONYM GUIDE ..................................................................................230

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1. THE BASICS OF OSSs


The easiest place to start a discussion of OSSs is with the fundamental systems in the ordering process for any-voice services provider. The process flow
from placing an order for service to activating that service on the network
leads through workflow, ordering, inventory, circuit design/engineering, provisioning, and activation systems.

Figure 1. Process Flow

Workflow Engine
A workflow engine is generally at the heart of an integrated OSS infrastructure. It can be built in any number of configurations utilizing any number of
technologies, but its purpose is generally the same regardless. The workflow
engine manages the flow of information from system to system, essentially
checking off the tasks associated with any process as it goes. Some OSS vendors package workflow engines with their systems whereas other vendors
specialize in workflow. Workflow systems are sometimes telecom specific,
but just as often they are general information technology products that can
function effectively in any environment from telephony to financial services
to manufacturing. The workflow engine's utility, again, is managing and coordinating interactions between integrated systems.
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Ordering
The ordering system is where all the information necessary for providing service is entered into a service provider's systems. These services range from
basic, residential plain old telephone service (POTS) lines to complex services
such as channelized tier 1s (T1s)high-capacity pipes carrying voice and data
traffic, integrated services digital network (ISDN), asynchronous digital subscriber line (ADSL), and more. Modern ordering systems generally utilize a
graphical user interface (GUI), which guides order takers or customer-care
representatives through the ordering process for any number of services.
These systems also incorporate some default data common to each service a
provider offers to ease the keystroke burden on those entering orders.
Ordering systems also perform a certain amount of error checking to notify
users when required data has been omitted or invalid data has been entered
in order to maintain overall process integrity and stop faulty or incomplete
orders from being passed on.
Once an order is entered, the system generates specific tasks that must be
completed to activate service on the network. The ordering system passes
these tasks on to other systems, which in turn update the ordering system as
they complete each task to provide a current status report for each service
order. The workflow engine generally supervises these tasks, ensuring that
each system performs its specified function in the proper sequence and within established time parameters.
Inventory
In the inventory system, a carrier stores all its information regarding the facilities and equipment available on its network. To process an order, the inventory system must be queried to determine whether or not the requested service
can be supplied. Is the proper equipment in place, or must new equipment be
installed? Are the proper facility circuitsthe high-capacity circuits that provide backbone transportalready assigned, or do they need to be configured?
Circuit Design/Engineering and Provisioning
These systems manage and track the equipment and circuits that physically
provide service and that must be assigned for eventual activation. Often
referred to as design and assign, they basically involve specifying which
pieces of equipment and network routes a given service will utilize. For
example, if T1 service is requested, channels, ports, cards, and circuits must
be assigned on any combination of M13 multiplexers, digital cross-connect

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systems, T3 facility circuits, or synchronous optical network (SONET) channels and network routes connecting carrier network locations to the end user.
Network locations are identified by a Telcordia Technologies (formerly
Bellcore) standard, eight- or eleven-digit common language location identifier
(CLLI) codes. For example, a CLLI code of PLANTXXAH01 would indicate a
SONET shelf at the "A" designated end office located in Plano, Texas.
Similarly, exchange carrier circuit identification (ECCKT) codes identify specific circuits.

Figure 2. CLLI Code


A current trend for design-and-assign systems is to incorporate graphical tools
that allow a system user to create services on a network map with point-andclick capability rather than either drawing maps by hand or relying on an
abstract set of equipment identifiers displayed in a table.
Element Management and Activation and Field Service Management
Once the previous tasks are accomplished, service can be activated on the
network. Activation requires several steps. If new equipment or lines must be
installed, or if equipment or lines must be configured manually, a field servicemanagement system must be notified so that technicians can be dispatched. Field service systems must not only notify technicians of the service
being installed but also of the specific equipment involved and where it is
located. For example, services provided to a large office complex must be
associated with a building, floor, network, closet, and perhaps a certain equipment rack within that closet.
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Some activation can be performed automatically. Today's service providers


are working toward enabling flow-through provisioning and activation, combining provisioning and activation systems to allow order and design-andassign systems to issue commands to an activation system. The activation
system then automatically activates service on the proper network elements
(any piece of network hardware, such as a switch, multiplexer, or cross-connect system).
Current network elements are generally designed with an intelligent element
manager built in that can receive and execute commands sent by activation
systems. Element managers also can feed equipment status data back to
upstream systems for network- and trouble-management functions. Element
managers use protocols such as common management information protocol
(CMIP), transaction language 1 (TL1), or simple network management protocol (SNMP) for traditional data equipment to communicate with activation
and other systems. An activation system often acts as a manager of managers,
overseeing and communicating with a number of various element managers
and equipment types.

Figure 3. Manager of Managers


Network and Trouble Management

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OSSs certainly go beyond service activation. Two critical elements of any
OSS infrastructure are network- and trouble-management systems. Network
management systems are responsible for the overall supervision of a network.
They monitor traffic traversing the network and collect statistics regarding
performance. They also are responsible for spotting trouble on a network and
identifying the cause. Network-management systems are the heart of a network operations center (NOC) and are often known for the graphical network displays projected on large screens on the walls. Network-management
systems utilize protocols such as SNMP and CMIP to communicate with network elements.
Network elements are designed to provide varying levels of self-diagnosis.
While older elements might simply send an alarm to supervisory systems
announcing a problem, newer, more intelligent elements are often designed to
provide more precise trouble messages. A problem in a network, such as
damage to a fiber-optic line or switch failure, can result in a chain reaction
where many network elements along a certain path, or along multiple paths,
will produce alarms. Network management systems are generally designed to
correlate these alarms to locate the source of a problem.
Once the system identifies trouble, it passes information on to a trouble management system that logs the problem and issues a trouble ticket to begin the
repair process. Some network elements have enough intelligent routing capability built in to automatically reroute network traffic around problem areas.
Where this is not the case, trouble spots must be identified to allow human
operators to reroute traffic. A trouble-management system in an integrated
OSS environment can send commands to the appropriate systems, such as
field service management, to dispatch technicians who physically repair
equipment.

Figure 4. Trouble Management

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2. OSS INTERCONNECTION
Regulations
A critical portion of the Telecommunications Act of 1996 and its associated
orders deals with OSS interconnection. Regulations require the regional Bell
operating companies (RBOCs) to allow competitors limited access to their
customer databases and various OSS functions, such as preordering, ordering,
and provisioning. (Preordering is the process by which a competitive localexchange carrier [CLEC], with permission from the customer, requests data
regarding that customer from an RBOC.) The Federal Communications
Commission (FCC) has made it clear that RBOCs will not be permitted to
enter the long-distance business until, among other things, they create access
mechanisms that both state regulatory commissions and the FCC deem sufficient to enable competition. RBOCs and incumbent local-exchange carriers
(ILECs) have built or are building interfaces into which a CLEC can connect
its systems. This is an extremely difficult and time-intensive task, one with
which the industry has wrestled for several years. In the meantime, carriers
often rely on manual means, such as phone calls and faxes, to exchange customer data and service orders. Manual processes are highly error-prone and
slowinsufficient for a truly competitive environment.
Interconnection Challenges
The first problem RBOCs face in enabling interconnection is integrating their
own OSSs. A large number of RBOC OSSs are stand-alone mainframe systems that were never intended to be integrated or accessed by anyone but the
RBOCs. Often referred to as legacy systems, they were designed to assist
people in their daily jobs. Most RBOCs are conglomerations of many smaller
local phone providers and are still in the process of consolidating, integrating,
and eliminating their legacy systems. These systems cannot be easily
replaced, however, due to both the cost and time involved in such a largescale project and the fact that these systems are critical to everyday RBOC
business processes. RBOCs are working steadily, though the process is innately slow, to replace their older systems with modern, integrated OSS packages.

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Figure 5. RBOC Legacy Environment


RBOC legacy systems do not have sufficient security mechanisms to partition
customer datain other words, to keep RBOC customer data separate from
CLEC customer data. Additionally, external interfaces must be added on to
these systems to allow integration with surrounding systems. Without such
integration, functions such as flow-through provisioning are impossible to
enable. These systems also must be able to respond to commands coming
from an interconnection gateway in order to fulfill CLEC data requests.
There are many conceptual and technological approaches to legacy systems
integration. These approaches involve technologies such as middleware,
transaction processors (TPs), workflow systems, and object engines.
Middleware is a term commonly applied to any integration technology, and it
is often used interchangeably with TP. These technologies present a common
application programming interface (API) into which a system can be integrated to manage data translation and exchange among disparate systems.
Workflow systems often work hand in hand with TPs, providing multiple,
dynamic APIs and managing data flow and task sequencing while the TP handles data conversion. Object engines use technologies such as the Object
Management Group's (OMG) common object request broker architecture
(CORBA) or Microsoft's distributed component object model (DCOM).
Object engines abstract application interfaces into definable, flexible software
objects that allow applications tocommunicate in a uniform manner through
the engine itself.
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Figure 6. Integration Technology


While RBOCs work to integrate their systems and make them accessible for
CLECs, they also must develop their interconnection interfaces and integrate
their systems and business processes with them. Because there is no industry
standard or consensus about how this should be accomplished, RBOCs often
rely on technologies they already utilize for information exchange with large
customers and interexchange carriers (IXCs). Once again, these technologies
are often older and not necessarily intended for tasks such as CLEC interconnection. They often make the most sense economically for RBOCs, however,
because a large amount of code is already in place. The most common protocol being used for interconnection is electronic data interchange (EDI) but in
various versions. EDI was originally designed to enable the exchange of business documents and is now being used mainly for ordering and preordering.
Gateway Functions
Many vendors have brought to market flexible gateway products intended to
help CLECs develop the interfaces necessary for interconnection with RBOC
OSSs. The TeleManagement Forum, an industry organization devoted to the
implementation of telecom standards such as the Telecommunications
Management Network (TMN), has led an initiative to develop guidelines for a
common interconnection gateway platform (CIGP). The goal of the CIGP is
to apply vendor-neutral, industry-common technologies to OSS interconnection in order to assist CLECs in developing interconnection interfaces. Most
of the vendors that have developed gateway products have been involved in
the CIGP initiative.

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A gateway's primary function, as mentioned above, is to manage the interfaces between CLEC and RBOC OSSs. Gateways handle data integrity and
security between carriers as critical customer and service data is exchanged.
One of the most important aspects of a gateway, however, is to perform error
checking on service orders as they are passed across carrier boundaries. With
manual processes, a CLEC often sends service orders to an ILEC or RBOC
that end up lost in a pile of faxes for several days. When orders are finally
attended to, they are rejected if they are incomplete or somehow erroneous
a common occurrence because orders can be rejected for simple typographical
errors. They are only then returned to the CLEC for reprocessing. This adds
days and even weeks to the ordering process. A gateway can reduce these
errors by reviewing all orders before submission to the ILEC, returning any
erroneous orders for instant review.
Another critical function of a gateway is to facilitate the preordering process.
In this process, the CLEC secures permission from a potential customer to
obtain its data from the ILEC. This data consists of a customer profile, outlining all the service provided to the customer. This data is often transferred in
the form of universal service order codes (USOC). These codes are cryptic,
and there are thousands of them. In a manual process, a CLEC customer representative must flip through a large catalog to determine the services provided and to build sales quotes for similar offerings. Again, this is an extremely
time-intensive process. New gateway software can read these codes and
match them to a CLEC's product catalog database to automatically generate
product offerings and sales quotes, making the CLEC customer-acquisition
process far more efficient.

Figure 7. Interconnection Process

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3. OPERATIONS SUPPORT OF DATA SERVICES


As complex as the OSS infrastructure for a wireline network is today, it will
only become more complex as new network technologies are introduced to
the carrier environment. Packet technologies, such as Internet protocol (IP),
frame relay (FR), and asynchronous transfer mode (ATM), are becoming
increasingly prevalent in the public network. While service providers have
been managing FR and ATM for several years, the demand for more featurerich services has necessitated reworking OSSs to support the complexities of
service-level agreement (SLA) management, usage-based billing, and flexible
quality-ofservice (QoS) parameters.
IP, the technology that drives the Internet, is developing into a carrier-grade
technology that will enable a mix of voice and data services to be more
advanced and widely available than has ever been possible before. Like ATM
and FR, IP services are demanding support to ensure high QoS. Two major
hurdles must be overcome to meet that goal. First, service providers must
adopt QoS that can map to both connection-oriented and connectionless protocols. They also must address the integration of an IP-address management
system.
Data Service Provisioning

Figure 8. Service-Order Work Request

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Assuming a service-management perspective for offering data services, the
service provider defines the bandwidth access circuits for the A and Z locations and the bandwidth for the QoS, service category parameters, or both as
they relate to the particular permanent virtual circuit (PVC). After provisioning the equipment, the provider defines a virtual layout for the field to use for
the actual mapping of the virtual circuit (VC) to the equipment.
Data Service Activation
As end-to-end automation is becoming the norm, the ability to pass the virtual layout to the network-management layer (NML) for activation is necessary.
This involves using an NML manager to activate the appropriate equipment.
Understanding the service provider's network is key to this process because
service providers cannot activate a PVC for an endpoint that is not under their
control.
Advent of Broadband Access
Broadband-access technologies also are having a huge impact on a service
provider's OSS. Digital subscriber lines (xDSLs) and cable modems are currently the leading data-access technologies most likely to be deployed in the
United States.
The xDSL technologies that enable existing local loops (the copper wires that
connect end users to the public network) to carry higher-capacity data
streams than common analog-modem technologies come in several flavors.
They also permit simultaneous voice and data streams to travel over the same
wire pair. Having accurate records of their copper infrastructure is a major
concern for incumbent service providers. Service providers that are Internet
service providers (ISPs) or CLECs also have major concerns about getting
access to unbundled loops and a clear communication path to the incumbent
provider.
A central office (CO) must incorporate two new components to enable xDSL
technologies: a splitter and a digital subscriber line access multiplexer
(DSLAM). The splitter simply distributes the voice traffic to the POTS network and the data traffic to the DSLAM; it is expected that the splitter will
largely become obsolete as the demand for an all-in-one box increases. The
DSLAM communicates with the xDSL modems installed at the end-user location and aggregates multiple xDSL streams into a switch for transport on
high-capacity circuits using various multiplexing schemes. It is managed and
maintained much like other end-office equipment, but most installed OSSs do

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not yet support the technology. xDSL modules must be added to older OSS
systems to enable automatic provisioning and management of xDSL services.
The splitter simply distributes the voice traffic to the POTS network and the
data traffic to the DSLAM.
The new xDSL technology has several core functions that the existing OSS
should support. For example, the DSLAM and splitter, while specific to xDSL
technologies, are very similar to a service provider's existing equipment (i.e.,
routers and switches) in terms of equipment inventory. Supporting customer
premises equipment (CPE), on the other hand, may be a new challenge for
the service provider. However, providers with a managed service offering
may find they also can handle the CPE network aspects.
The more complex scenarios for broadband access involve incorporating VCs
along with voice services. OSSs traditionally have not viewed an xDSL as
capable of providing this type of service. One approach is to handle the cable
pair as a channelized T1 circuit capable of handling both voice and data circuits. The scenarios typically encountered range from only offering xDSL on
the cable pair, with no voice service, to offering a small office with multiple
users an xDSL solution involving voice channels as well as several VCs that
each have differing levels of service (such as an analog phone, a PVC for
Internet access, a PVC to corporate headquarters, and an Internet phone connection).

Figure 9. DSLAM

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One of the problems with xDSL technologies is that they are susceptible to a
number of network pitfalls. For example, local loops that are equipped with
special noise filters or load coils will filter out the frequencies at which xDSLs
operate, rendering them ineffective. Additionally, some services can create
interference on xDSLs. If an xDSL loop rides in the same bundle as a loop
delivering one of these services, the xDSL service can be disturbed. Also,
some older copper wires, installed years ago, are simply insufficient to support the service. Some RBOC regions lack detailed line records that can
inform service providers of potential problems because line records are kept
on spreadsheets or even by hand and are not updated accurately. A strong
network-inventory system is thus critical to the effective deployment of xDSL
services.

4. BUSINESS IMPACT OF AN OSS SOLUTION


Service providers are constantly striving to differentiate their businesses from
those of their competitors through superior customer service and rapid timetomarket for new products and services. A powerful OSS solution can help
service providers meet these goals while controlling their operating costs.
Quality of Service
In its simplest terms, QoS is a measure of the telephone service quality provided to a subscriber. This measurement can be very subjective, and the ability to define it depends upon the technology being used. For example, ATM
and FR technologies were designed with multiple grades of service delivery in
mind, but IP technology was not. IP is, however, the leading technology for
enabling nextgeneration telecommunications services. Unlike the circuitswitching technology that makes up the public voice network, IP networks
are connectionless. Circuit networks utilize dedicated, 64 kb connections to
support service delivery. This provides high-quality service because only the
traffic for a specific session can utilize the dedicated network path, but it also
is bandwidth inefficient. If two people on a voice call are not speaking, bandwidth goes unused. In a world where time is measured in milliseconds, even
a half-second of dead air is an enormous waste of resources. An IP session
can utilize multiple paths to complete its delivery and only uses as much
bandwidth as it needs, allowing traffic to mix on network paths in order to
maximize bandwidth usage. The downside of IP telephony, however, is that
its best-effort delivery model does not guarantee delivery of packets in order,
in a timely manner, or at all. Therefore, acceptable quality levels must be

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maintained in order to successfully deploy real-time applications over IP networks. Solutions include using virtual private networks (VPNs) and IP over
ATM.
Data Warehousing
Measuring QoS compliance requires having access to accurate and timely
data. While traditional OSSs may be adequate for the service provider's dayto-day operations, they lack the ability to provide these vital performance
metrics. New OSSs, on the other hand, can quickly and efficiently access historical data by taking advantage of data warehousing technology. Data warehousing consists of storing information from disparate systems in a central
repository or a single database and then carefully managing this data to
ensure its integrity. Management can draw upon this wealth of information
when access to the latest business intelligence is necessary, not only for QoS
analysis but also to analyze market trends and adjust product strategy accordingly.
Operational Efficiencies
In order to be successful, an OSS solution must mirror the service provider's
business processes. Most OSS solutions today are considered commercial offthe shelf (COTS) applications. While these packages do not offer pure off-theshelf functionality, they are designed to be customized to fit how a company
does business. For example, a sophisticated OSS solution will offer work
management capabilities that enable users to maintain provisioning plans that
manage the flow of work and information within their unique organization.
The Importance of Flexibility
Given today's dynamically changing marketplace, flexibility also is of utmost
importance in an OSS solution. A technology-neutral OSS is built upon an
architecture that supports both current and new technologies, enabling service providers to respond immediately to business changes, whether these
changes stem from marketing decisions, new technologies, or regulatory
requirements.

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5. CONCLUSION
Ideally, an OSS's architecture enables work to flow electronically across the
organization, providing visibility to the processes and resource utilization. It
also should enable the service provider to manage the end-to-end service
delivery process that often involves more than one type of order or transaction across the organization, as well as with other service or network
providers. Most important, this software should be available in a single solution, eliminating the complexity of dealing with a variety of systems. If a service provider cannot achieve this goal due to complex OSS requirements, the
provider should carefully select best-of-breed vendors offering proven, integrated solutions.

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6. SELF-TEST
1. Which of the following is a function of a workflow engine?
a. to provide network repair information for field service technicians
b. to prompt customer-care representatives to sell specific service
packages
c. to facilitate communication and task sequencing among various
OSSs
d. to draw graphical network maps for capacity planning
2. A CLLI code is a Telcordia Technologies (formerly Bellcore) standard code
used for _____________.
a. identifying specific circuit paths
b. identifying network locations
c. identifying a customer's long-distance carrier
d. transmitting fiber-optic signals
3. Which of the following is not a protocol used for communicating with network elements?
a. CMIP
b. SNMP
c. TL1
d. TMN
4. All network elements are equipped with built-in intelligence that will reroute network traffic around trouble spots.
a. true
b. false

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5. Legacy systems are ________.


a. systems created from parts of other systems
b. any RBOC OSS
c. older, stand-alone mainframe systems common to ILECs
d. systems used to interconnect LEC OSSs
6. OSS interconnection is mandated and a critical factor in determining ILEC entry
into long-distance markets.
a. true
b. false
7. Interconnection gateways often perform error-checking functions to help speed the
ordering process.
a. true
b. false
8. Which of the following is not an inhibitor to DSL deployment?
a. DSLAMs cannot be housed with circuit switches.
b. Some lines carry load coils and filters that can negate DSLs.
c. LEC line records are often inaccurate.
d. Services riding adjacent lines can interfere with DSLs.
9. IP guarantees time and delivery sequence for all packets on a network.
a. true
b. false

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10. Data warehousing provides quick and easy access to performance metrics.
a. true
b. false
11. If Mickey is a mouse, Donald is a duck, and Pluto is a dog, what is Goofy?
a. a dog
b. a dawg

7. ACRONYM GUIDE
ADSL

asynchronous digital subscriber line; current technology that


enables digital, high-capacity, and simultaneous voice and data
transmission over copper local loops; called asynchronous
because it utilizes a higher-capacity channel going to the user
than coming from the user analog modem a device designed to
transmit and receive signals over regular telephone lines, most
common method for accessing the Internet today

API

application programming interface

ATM

asynchronous transfer mode; a switching technology that packages voice, data, or video traffic in fixed-length cells and incorporates variable QoS parameters

circuit
switch

CLEC

230

the network equipment that controls all traffic routing in the traditional voice network by establishing point-to-point circuits connecting network locations; a normal voice call is provided a dedicated 64 kbps circuit, which is multiplexed onto a higher-capacity
circuit for switching and transmission throughout a circuitswitched, voice network
competitive local-exchange carrier; a new class of local voice service providers that compete with the former local carrier monopolies; often called new entrants or emerging carriers

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CLLI

common language location identifier codes; a system devised by


Telcordia Technologies (formerly Bellcore) for identifying locations and equipment in a telephone network

CMIP

common management information protocol; an advanced protocol used to send and receive information to and from network
elements

CIGP

common interconnection gateway platform; an initiative led by


the TeleManagement Forum to develop a technology-neutral
standard model for the design of OSS interconnection gateways

CO

central office

CORBA

common object request broker architecture

COTS

commercial off-the-shelf

DCOM

distributed component object model

DSLAM

digital subscriber line access multiplexer; a multiplexing system


used to aggregate digital subscriber line traffic before transmission across a network

ECCKT

exchange carrier circuit identification

EDI

electronic data interchange; a data transfer technology originally


designed for the automated exchange of business documents;
also one of the predominant technologies currently used for automated exchange of service orders between CLECs and ILECs

facility
circuit

a high-capacity carrier circuit from which serving circuits are


assigned

FCC

Federal Communications Commission

FR

frame relay; a switching technology, mainly used for data transmission, that uses variable length frames to package and send
information

GUI

graphical user interface

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ILEC

incumbent local-exchange carrier; the former monopoly local service provider in a given region; all RBOCs are ILECs, but not all
ILECs are RBOCs

IP

Internet protocol; a best-effort routing technology that uses packets to package and send voice, video, or data traffic; the underlying transmission technology that enables the Internet

ISDN

integrated services digital network; a 1970s digital subscriber line


system, still in relatively wide use today, that enables digital,
high-capacity, simultaneous voice and data transmission over
copper local loops

ISP

Internet service provider

IXC

interexchange carrier; commonly known as a long-distance carrier

legacy
systems

common term for the older vintage systems often employed in


carrier back offices

M13
multiplexer

a type of network equipment that multiplexes multiple T1


circuits onto a T3

multiplexing

the general practice of aggregating and combining lowercapacity traffic streams onto higher-capacity carrier circuits

NML

network-management layer

NOC

network operations center

OMG

Object Management Group

OSS

operations support system; system that directly supports the


operations of the telecommunications infrastructure

POTS

plain old telephone service

PVC

permanent virtual circuit

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QoS

quality of service

RBOC

regional Bell operating company; also known as the baby Bells;


the once seven, now five, companies that were divided out of the
old Bell system and until recently held local telephone monopolies in most of the United States (Bell Atlantic, BellSouth,
Southwestern Bell, U S West, and Ameritech)

SLA

service-level agreement

SNMP

simple network-management protocol; the protocol most commonly used for data network element management

SONET

synchronous optical network; a North American standard for


fiber-optic voicetransmission technology, equipment, and network architecture

T1

a digital transmission link with a capacity of 1.544 Mbps


(1,544,000 bits per second)

TL1

transaction language 1; a predominant control protocol, in use for


many years, for issuing commands to voice network elements

TMN

telecommunications management network; a set of standards,


originally presented by the International Telecommunications
Union (ITU) and further developed by the TeleManagement
Forum, that provides architectures, protocols, and interfaces for
building standardized OSS applications and infrastructures

TP

transaction processor

USOC

universal service order code; the codes used to identify specific


services offered on the voice network; critical to the preordering
process; thousands of such codes exist, and they are rarely uniform from region to region

VPN

virtual private network

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Optical Networks
DEFINITION
Optical networks are high-capacity telecommunications networks based on
optical technologies and components that provide routing, grooming, and
restoration at the wavelength level as well as wavelength-based services.

TUTORIAL OVERVIEW
As networks face increasing bandwidth demand and diminishing fiber availability, network providers are moving toward a crucial milestone in network
evolution: the optical network. Optical networks, based on the emergence of
the optical layer in transport networks, provide higher capacity and reduced
costs for new applications such as the Internet, video and multimedia interaction, and advanced digital services.
As with any new technology, many questions arise: How is the optical network different from existing networks? What are the network elements
required for optical networks? What applications do optical networks best
suit? This tutorial will address all of these questions and explain the technologies, architectures, and market trends of emerging optical networks.

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TOPICS
1.

BENEFITS AND HISTORY OF OPTICAL NETWORKS . . . . . . . . . . 237

2.

OPTICAL NETWORK DRIVERS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 239

3.

ENABLING TECHNOLOGIES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241

4.

TECHNOLOGIES ON THE HORIZON . . . . . . . . . . . . . . . . . . . . . . . 245

5.

COMPONENT APPLICATIONS. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 248

6.

MARKETS FOR OPTICAL NETWORKS . . . . . . . . . . . . . . . . . . . . . . 248

7.

DESIGN AND PLANNING . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 250

8.

RESTORATION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252

9.

NETWORK MANAGEMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253

10. NETWORK EVOLUTIONS: PART I . . . . . . . . . . . . . . . . . . . . . . . . . . 254


11. NETWORK EVOLUTIONS: PART II . . . . . . . . . . . . . . . . . . . . . . . . . 257
12. THE FUTURE OF OPTICAL NETWORKS . . . . . . . . . . . . . . . . . . . . . 260
13. SELF-TEST. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260
14. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263

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1. BENEFITS AND HISTORY OF OPTICAL NETWORKS


In the early 1980s, a revolution in telecommunications networks began,
spawned by the use of a relatively unassuming technology: fiber-optic cable.
Since then, the tremendous cost savings and increased network quality has
led to many advances in the technologies required for optical networks, the
benefits of which are only beginning to be realized.
History
Telecommunication networks have evolved during a century-long history of
technological advances and social changes. The networks that once provided
basic telephone service through a friendly local operator are now transmitting
the equivalent of thousands of encyclopedias per second. Throughout this
history, the digital network has evolved in three fundamental stages: asynchronous, synchronous, and optical.
Asynchronous
The first digital networks were asynchronous networks. In asynchronous
networks, each network elements internal clock source timed its transmitted
signal. Because each clock had a certain amount of variation, signals arriving
and transmitting could have a large variation in timing, which often resulted
in bit errors.
More importantly, as optical-fiber deployment increased, no standards existed
to mandate how network elements should format the optical signal. A myriad
of proprietary methods appeared, making it difficult for network providers to
interconnect equipment from different vendors.
Synchronous (SONET)
The need for optical standards led to the creation of the synchronous optical
network (SONET). SONET standardized line rates, coding schemes, bit-rate
hierarchies, and operations and maintenance functionality. SONET also
defined the types of network elements required, network architectures that
vendors could implement, and the functionality that each node must perform.
Network providers could now use different vendors optical equipment with
the confidence of at least basic interoperability.

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Optical Networks
The one aspect of SONET that has allowed it to survive during a time of
tremendous changes in network capacity needs is its scalability. Based on its
open-ended growth plan for higher bit rates, theoretically no upper limit
exists for SONET bit rates. However, as higher bit rates are used, physical
limitations in the laser sources and optical fiber begin to make the practice of
endlessly increasing the bit rate on each signal an impractical solution.
Additionally, connection to the networks through access rings has also had
increased requirements. Customers are demanding more services and options
and are carrying more and different types of data traffic. To provide full endto-end connectivity, a new paradigm was needed to meet needs for high
capacity and varied customer demands. Optical networks provide the
required bandwidth and flexibility to enable end-to-end wavelength services
(see Figure 1).

Figure 1. End-to-End Wavelength Services


Optical networks began with wavelength division multiplexing (WDM),
which arose to provide additional capacity on existing fibers. Like SONET,
defined network elements and architectures provide the basis of the optical
network. However, unlike SONET, rather than using a defined bit rate and
frame structure as its basic building block, the optical network will be based
on wavelengths. The components of the optical network will be defined
according to how the wavelengths are transmitted, groomed, or implemented
in the network.

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Optical Networks
Viewing the network from a layered approach, the optical network requires
the addition of an optical layer. To help define network functionality, networks are divided into several different physical or virtual layers. The first
layer, the services layer, is where the servicessuch as data trafficenter
the telecommunications network. The next layer, SONET, provides restoration, performance monitoring, and provisioning that is transparent to the
first layer.
Emerging with the optical network is a third layer: the optical layer.
Standards bodies are still defining the optical layer, but it will eventually provide the same functionality as the SONET layer while operating entirely in
the optical domain. The optical network also has the additional requirement
of carrying varied types of high bit-rate nonSONET optical signals that
bypass the SONET layer altogether. Just as the SONET layer is transparent to
the services layer, the optical layer will ideally be transparent to the SONET
layer, providing restoration, performance monitoring, and provisioning of
individual wavelengths rather than electronic SONET signals.

2. OPTICAL NETWORK DRIVERS


Many factors are driving the need for optical networks. A few of the most
important reasons for migrating to the optical layer are described in this module.
Fiber Capacity
The first implementation of what has emerged as the optical network began
on routes that were fiber-limited. Providers needed more capacity between
two sites, but higher bit rates or fiber were not available. The only options in
these situations were either to install more fiber, which is an expensive and
labor-intensive chore, or place more time division multiplexed (TDM) signals
on the same fiber. WDM provided many virtual fibers on a single physical
fiber. By transmitting each signal at a different frequency, network providers
could send many signals on one fiber just as though they were each traveling
on their own fiber.
Restoration Capability
As network planners use more network elements to increase fiber capacity, a
fiber cut can have massive implications. In current electrical architectures,
each network element performs its own restoration. For a WDM system with
many channels on a single fiber, a fiber cut would initiate multiple failures,

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causing many independent systems to fail. By performing restoration in the
optical layer rather than the electrical layer, optical networks can perform protection switching faster and more economically. Additionally, the optical layer
can provide restoration in networks that currently do not have a protection
scheme. By implementing optical networks, providers can add restoration
capabilities to embedded asynchronous systems without first upgrading to an
electrical protection scheme.
Reduced Cost
In systems using only WDM, each location that demultiplexes signals will
need an electrical network element for each channel, even if no traffic is dropping at that site. By implementing an optical network, only those wavelengths that add or drop traffic at a site need corresponding electrical nodes.
Other channels can simply pass through optically, which provides tremendous cost savings in equipment and network management. In addition, performing space and wavelength routing of traffic avoids the high cost of electronic cross-connects, thereby simplifying network management.
Wavelength Services
One of the great revenue-producing aspects of optical networks is the ability
to resell bandwidth rather than fiber. By maximizing capacity available on a
fiber, service providers can improve revenue by selling wavelengths, regardless of the data rate required. To customers, this service provides the same
bandwidth as a dedicated fiber (see Figure 2).

End-to-End
Wavelength
Services

Figure 2. End-to-End Wavelength Services

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3. ENABLING TECHNOLOGIES
The cornerstone of an optical network is the advanced optical technologies
that perform the necessary all-optical functions. Optical technologies continue
to advance by ingenious techniques and implementations to improve the performance and capabilities of the optical network (see Figure 3).

Figure 3. Development Milestones


Early Technologies
As fiber-optics came into use, network providers soon found that some
improvements in technology could greatly increase capacity and reduce cost
in existing networks. These early technologies eventually led to the optical
network as it is today.
Broadband WDM
The first incarnation of WDM was broadband WDM. In 1994, by using fused
biconic tapered couplers, two signals could be combined on the same fiber.
Because of limitations in the technology, the signal frequencies had to be
widely separated, and systems typically used 1310 and 1550 nm signals, providing 5 Gbps on one fiber. Although the performance did not compare to
todays technologies, the couplers provided twice the bandwidth out of the
same fiber, which was a large cost savings compared to installing new fiber.

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Optical Amplifiers
The second basic technology, and perhaps the most fundamental to todays
optical networks as well, was the erbium-doped optical amplifier. By doping
a small strand of fiber with a rare earth metal, such as erbium, optical signals
could be amplified without converting the signal back to an electrical state.
The amplifier provided enormous cost savings over electrical regenerators,
especially in long-haul networks.
Current Technologies
Systems deployed today use devices that perform functions similar to earlier
devices but are much more efficient and precise. In particular, flat-gain optical
amplifiers have been the true enablers for optical networks by allowing the
combination of many wavelengths across a single fiber.
Dense Wavelength Division Multiplexing (DWDM)
As optical filters and laser technology improved, the ability to combine more
than two signal wavelengths on a fiber became a reality. Dense wavelength
division multiplexing (DWDM) combines multiple signals on the same fiber,
ranging up to 40 or 80 channels. By implementing DWDM systems and optical amplifiers, networks can provide a variety of bit rates, i.e., OC48 or
OC192, and a multitude of channels over a single fiber (see Figure 4). The
wavelengths used are all in the range that optical amplifiers perform optimally, typically from about 1530 nm to 1565 nm (see Figure 5).

Figure 4. DWDM Systems and Optical Amplifiers

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Figure 5. ITU Channel Spacing


Two basic types of DWDM are implemented today: unidirectional and bidirectional DWDM (see Figure 6). In a unidirectional sndors and deployed
over a wide time frame, the hodgepodge of network protocols and control
messages makes this situation necessary.
In this access gateway application scenario, the access gateway again is used
to connect the telephone network to the Intl amplifiers.

Figure 6: Unidirectional and Bidirectional DWDM

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The performance of optical amplifiers has improved significantlywith current amplifiers providing significantly lower noise and flatter gainwhich is
essential to DWDM systems. The total power of amplifiers has also steadily
increased, with amplifiers approaching +20dBm outputs, which is many
orders of magnitude more powerful than the first amplifiers.
Narrowband Lasers
Without a narrow, stable, and coherent light source, none of the optical components would be of any value in the optical network. Advanced lasers with narrow bandwidths provide the narrow wavelength source that is the individual
channel in optical networks. Typically, long-haul applications use externally
modulated lasers while shorter applications can use integrated laser technologies.
These laser sources emit a highly coherent signal that has an extremely narrow bandwidth. Depending on the system used, the laser may be part of the
DWDM system or embedded in the SONET network element. When the precision laser is embedded in the SONET network element, the system is called
an embedded system. When the precision laser is part of the WDM equipment in a module called a transponder, it is considered an open system,
because any low-cost laser transmitter on the SONET network element can
be used as input (see Figure 7).

Figure 7. Embedded vs. Open Systems DWDM


Fiber Bragg Gratings
Commercially available fiber Bragg gratings have been important components
for enabling WDM and optical networks. A fiber Bragg grating is a small section of fiber that has been modified to create periodic changes in the index of
refraction. Depending on the space between the changes, a certain frequency
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of lightthe Bragg resonance wavelengthis reflected back while all other
wavelengths pass through (see Figure 8). The wavelength-specific properties
of the grating make fiber Bragg gratings useful in implementing optical
add/drop multiplexers. Bragg gratings are also being developed to aid in dispersion compensation and signal filtering as well.

Figure 8. In-Fiber Bragg Grating Technology: Optical A/D Multiplexer


Thin Film Substrates
Another essential technology for optical networks is the thin film substrate.
By coating a thin glass or polymer substrate with a thin interference film of
dielectric material, the substrate can be made to pass through only a specific
wavelength and reflect all others. By integrating several of these components,
many optical network devices are created, including multiplexers, demultiplexers, and add-drop devices.

4. TECHNOLOGIES ON THE HORIZON


Key functions have been identified as requirements for the emerging optical
network (see Figure 9). As component technologies advance, each of the functions required, such as tunable filters, space switches, and wavelength converters, will become more cost-effective and practical.

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Figure 9. Key Functional Blocks for WDM Transport Systems


One of the most promising technologies for optical networks is the semiconductor optical amplifier (SOA). By integrating amplifier functionality into the
semiconductor material, the same basic component can perform many different applications (see Figure 10). SOAs can provide integrated functionality of
internal switching and routing functions that are required for a feature-rich
network. Space switches, wavelength converters, and wavelength selectors all
can be made from SOAs, which will lead to large cost reductions and
improved performance in future optical-network equipment (see Figure 11).

Figure 10. Semiconductor Optical Amplifier Technology

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Figure 11. All-Optical Semiconductor Wavelength Converter


Promising new gain-switching technology makes possible optical space switches, selectable filters, and wavelength converters. Todays transmission systems
employ NRZ at OC48 (2.5 Gbps) and OC192 (10 Gbps) data rates. However,
new transmission technologies are being studied to open the way to OC768
(40 Gbps). These new systems might be based on either electronic time division multiplexing (ETDM) or optical time division multiplexing (OTDM)
4X10Gbps technologies. Advances are being made with integrated laser modulators that provide lower-cost narrowband transmitters (see Figure 12).

Figure 12. Integrated Laser Modulator


Soliton transmission, first deployed in submarine links, might find application
in terrestrial networks to improve transmission performance or in some types
of all-optical signal processing such as 3R regeneration. Research dealing with
polarization-mode dispersion mitigation, phase-shaped binary transmission
(PSBT), and fiber-grating technologies promise significant advances in the near
future with regard to increasing system performance and network capacity.
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All of these technologies aim to reduce the network cost and provide valuable
new services to customers who are constantly demanding more bandwidthintensive and flexible features from their network providers.

5. COMPONENT APPLICATIONS
Regardless of the component technologies that are implemented in the system, the optical network must perform several specific functions in order to
achieve maximum efficiency.
Wavelength Add/Drop Multiplexers
The first element to be integrated into the optical network is the optical multiplexer. The multiplexer combines multiple wavelengths onto a single fiber,
which allows all the signals to be routed along the same fiber. The initial
application for multiplexers has been to increase capacity on existing fiber
routes without adding more fiber, but they will serve as entry points to the
optical layer in many more aspects, including add/drop multiplexers and optical cross-connects.
Wavelength Switches
The ability to switch individual wavelengths is crucial to maximizing the
capacity and efficiency of optical networks. A wavelength switch provides
functionality similar to an electrical switch by routing an incoming wavelength to a variety of physical output ports.
Wavelength Converters
The final element in optical networks is the wavelength converter, which
converts an incoming signals wavelength to a different outgoing wavelength,
entirely in the optical domain. This will allow the network traffic to be
groomed to optimize for traffic patterns or network architecture.

6. MARKETS FOR OPTICAL NETWORKS


The evolution to the optical layer in telecommunications networks will occur
in stages in different markets because the traffic types and capacity demands
for each are different. Overall, the growth is predicted to be enormous (see
Figure 13). This module will review each potential market, including the main
drivers for deploying optical networks and issues that might arise.
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Figure 13. Optical Networks Market ($M)


Long-Haul Networks
Nowhere else is bandwidth devoured so quickly as in the long-haul network.
Often spanning thousands of miles, long-haul networks are different from all
other markets in several important regards: long spans between nodes and
extremely high-bandwidth requirements.
Long-haul networks were the first to have large-scale deployment of optical
amplifiers and wideband WDM systems, mainly because of cost reductions.
Optical amplifiers are a cheaper alternative to a large number of electrical
regenerators in a span. In addition, using WDM, interexchange carriers
increased the fiber capacity by using WDM, which avoids the large expenditures of installing new fiber.
Metro Interoffice (IOF) Networks
Networks in the metro interoffice (IOF) market have different needs for optical technologies. IOF networks are typically more interconnected and geographically localized. Because of the traffic patterns and distances between
offices, optical rings and optical cross-connects will be required much earlier.
IOF networks must not only distribute traffic throughout a region but connect
to the long-haul network. As the optical network evolves, wavelength
add/drop and interconnections will add the flexibility and value that IOF networks require.
Business Access Networks
The last mile of the network to business customers has gone by many names:
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The Basics of Telecommunications


access networks. Regardless of the name, these networks provide businesses with
connections to the telecommunications infrastructure. It is these networks where
the application of optical networks is not so clear. Many more complexities arise in
these networks, including variable bit-rate interfaces, different cost structures, and
different capacity needs. Similar in architecture to IOF networks, business-access
network sites are much closer together, so fiber amplification is not as important.
An important component for optical networks in business-access networks is the
asynchronous transponder, which allows a variety of bit-rate signals to enter the
optical network. Optical networks designed for the business-access environment
will need to incorporate lower-cost systems to be cost-effective and enable true
wavelength services. The challenge will be proving when and where DWDM is
effective in access networks.

7. DESIGN AND PLANNING


One of the largest challenges facing network planners who are implementing optical networks is the task of designing and planning the optical layer. Only when
providers begin to plan the optical network do some of the more intricate and difficult issues arise.
Span Designs
Ideally, the optical network will provide end-to-end services entirely in the optical
domain, without ever converting signals into electrical format. Unfortunately, for at
least a decade, it is probable that technology will not progress to the point at which
it is possible to transmit signals for long distances without electrical regeneration.
Even as optical regenerators become commercially viable, network spans will still
need to be designed to maintain signal quality throughout the entire signal path.
Planners must design optical networks so that signals traveling on the fiber
between one network element site and another, called a span, maintain their
quality. Many factors must be taken into account, including the optical signal-tonoise ratio (OSNR) (see Figure 14), chromatic dispersion (see Figure 15), and a myriad of nonlinear effects introduced by the interaction of the fiber with the optical
signal. The challenge of designing optical networks increases with the introduction
of optical cross-connects and add/drop multiplexers, which could dynamically
change a signal path to travel across a different physical route.

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Figure 14. Optical Signal-to-Noise Ratio

Figure 15. Chromatic Dispersion


Wavelength Routing Plans
The basic element in the optical network is wavelength. As many wavelengths
of signals are transported across the network, it becomes important to manage
and switch each one individually. One of the benefits of optical networks is
that they allow the network architecture to be different for each wavelength.
For example, one wavelength may be established in the network to be part of
a ring configuration, while another wavelength, using the same physical network, can be provisioned as a point-to-point system. The flexibility of provisioning the network one wavelength at a time has led to two definitions of
end-to-end services: wavelength paths and virtual wavelength paths.
Wavelength Path (WP)
The simplest implementation of a wavelength service in the optical network is a
wavelength path (WP). Using a WP, a signal enters and exits the optical layer at the
same wavelength, without ever changing to a different wavelength throughout the
network. Essentially a wavelength is dedicated to connect the two endpoints.
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Virtual Wavelength Path (VWP)
Although a WP is simple to implement, it can impose some limitations on the
bandwidth available in the network and the cost of implementing it. One
method by which to overcome this limitation is to use a virtual wavelength
path (VWP) in which a signal path can travel on different wavelengths
throughout the network. By avoiding a dedicated wavelength for an end-toend connection, the network can reuse and optimize wavelengths to provide
the greatest amount of capacity.

8. RESTORATION
As optical networks evolve, performing restoration at the optical layer can provide one of the greatest potential cost savings. By implementing a restoration
scheme at the optical layer, optical nodes can perform protection for all the
wavelengths on a path, with switching times similar to that of current electrical SONET rings (see Figure 16). Because protection is performed in the optical
layer, the electrical systems do not need the extensive protection architectures
that have been required historically, which provides tremendous cost savings
to network providers. In addition, optical layer restoration allows better wavelength utilization by implementing 1:N protection in the SONET/SDH layer.
Several methods of protection can be implemented in the optical network, all
of which are logically similar to their electrical counterparts.

Figure 16. Optical Reconfiguration Performance


Link
Link restoration is perhaps the simplest to implement in the optical network.
A link restoration routes the optical path across an alternate link between
sites, providing protection in case of a fiber or equipment failure. Although
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providers can dedicate fibers for a protection link, it is usually not cost-effective to do so. While a link restoration scheme can provide full restoration for
a single link failure, when fibers are shared for working or protection, it provides less than full protection for multiple failures.
Path
Using a 1:N path-restoration scheme allocates a disjointed path for an end-toend connection, but the alternate path is not dedicated for each connection.
So again, although this method provides full restoration for single link failures, it can provide less-than-full protection for multiple link failures.
Hybrid
A restoration similar to link protection is hybrid restoration. Hybrid restoration
provides protection for each link but attempts to improve fiber utilization by
eliminating the backhauling of traffic. To accomplish this, the switching near
the failed link takes place on nodes that might not be adjacent to the failure.
Ring
Perhaps the most robust protection architecture for optical networks is the
optical ring. Optical rings operate identically to their electrical ring equivalents, with the same architectures and alternatives available. Although they
require more fiber than other restoration schemes, optical rings provide the
highest level of availability. By partitioning wavelengths into groups, network
planners can switch certain wavelengths in the optical layer while still performing switching for existing systems in the SONET layer. The partitioning
allows a smooth evolution to optical rings.

9. NETWORK MANAGEMENT
One of the most important and difficult issues involved with the optical network is network management, for several reasons: restoration, performance,
and wavelength services. Although network management of optical networks
is a topic too large to cover extensively in an optical network tutorial, some
of the important issues are briefly discussed in this module.
First, the optical network is evolving and being implemented on top of an
existing SONET architecture, which provides its own restoration and protection schemes. Without a highly intelligent network management system, it
becomes extremely difficult to ensure that restoration schemes between the
electrical and optical layer do not conflict. In addition to mediation between
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the optical and SONET layer, the network management system must be able
to prevent possible conflicts or, at the minimum, enable the service provider
to identify conflicts.
In addition to managing the overall network architecture, network management systems must be able to monitor signal performance for each wavelength. With the addition of optical add/drop multiplexers and optical crossconnects, the end-to-end performance of wavelengths becomes more difficult.
Network management systems for the optical network must assist providers
in troubleshooting the network by isolating questionable wavelengths and the
possible location of degradation. As the number of wavelengths on each fiber
approaches forty or more, it is important to have an intelligent method to
monitor all of them.
Finally, and perhaps most important to the service providers, the ability to
manage and provide new services to customers quickly is crucial. As discussed earlier, provisioning end-to-end services can be difficult, especially as
network capacity decreases. An intelligent network management system can
help providers establish and monitor new end-to-end wavelength services to
maximize their bandwidth revenues.

10. NETWORK EVOLUTIONS: PART 1


As the optical network evolves, network planners must understand the
dilemma of how to utilize the optical network most effectively. On the one
hand, access networks require a transparent optical network that is bit-rate
and format independent. This would provide flexibility and allow connection to the network directly with ATM, TCP/IP, SONET, or any other signal
format without additional equipment costs. It would also allow wavelengths to be added and dropped optically and entirely without affecting
the original signal format.
Unfortunately, this transparent model for the access network falls apart completely when applied to metropolitan or long-haul networks. As the distances
increase, carriers must maximize the capacity to reduce costs, and to allow any
signal data rate onto the network would greatly increase costs. Therein lies the
dilemma: Networks need the flexibility to provide a variety of end-user services
without inefficiencies in the long-haul network. The solution is the optical gateway, which will integrate with existing optical network elements.
Discussed below are some of the optical network elements that make end-to-end
wavelength services a reality, and how they will be integrated into the network.
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Dense Wavelength Division Multiplexing
As discussed earlier, many network providers are already deploying DWDM
on a large scale. Fiber-congested, point-to-point segments of long-distance networks were one of the first applications for WDM terminals. Today, 16-channel DWDM terminals are widely deployed to enhance the bandwidth capacity
of the long-haul network backbone. Throughout 1998, the industry was scheduled to shift to 32 and 40-channel systems, and in following years to even
more.
Optical Add/Drop Multiplexers (OADM)
The OADM enhances the WDM terminals by adding several significant features
(see Figure 17). The OADM systems have the capacity of up to 40 optical wavelengths. They efficiently drop and add various wavelengths at intermediate sites
along the network resolving a significant challenge for existing WDM.

Figure 17. Optical ADM Functionality


Most important, OADM technology introduces asynchronous transponders to
allow the optical network element to interface directly to high-revenuegenerating services. It is now possible for asynchronous transfer mode (ATM),
frame relay, native LANs, high-bandwidth IP, and others to connect to the
network directly via a wavelength in the optical layer. Transponder technology also extends the life of older lightwave systems by accepting its bandwidth directly into the optical layer, converting its frequency to an acceptable
standard, and providing protection and restoration. The OADM is also the
foundation of optical bidirectional line switched rings (OBLSRs), which are
described in the next module.
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Optical Gateways
To access the optical network efficiently and maximize bandwidth capacity
and transport-protocol transparency, the optical gateway becomes a critical
network element (see Figure 18). As a variety of bit rates and signal formats,
ranging from asynchronous legacy networks to 10Gbps SONET systems, a
common transport structure must groom and provision traffic entering the
optical layer. The emerging basic format for high-speed transparent transport
is ATM, and optical gateways will allow a mix of standard SONET and ATM
services. By providing a link between the variety of electrical protocols and
allowing flexible deployment of any mix of them, optical gateways provide
networks with the maximum benefits of optical networking.

Figure 18: Optical Gateways


The optical gateway will be the key element to allow smooth transition to
optical networks. As more intelligence is added to the optical layer, costs can
be reduced in the SONET layer. For example, as optical rings are implemented, the optical gateway can interface lower cost 1:N protected SONET system
with the optical ring. By partitioning wavelengths, existing SONET rings can
be kept intact, while new systems are lower cost integrated 1:N tributaries to
the optical layer.

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Figure 19: Optical Ring Architecture

Figure 20: Optical Layer Rings

11. NETWORK EVOLUTIONS: PART II


Optical Bidirectional Line Switched Rings (OBLSR)
Optical-ring architectures utilize reconfigurable OADMs; the ring architecture is a familiar scheme to the telecommunications industry and is now
applied to the optical domain (see Figure 19). The optical ring uses the same
principles as the fiber ring to provide protection against equipment and network failures (See Figure 20).

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Figure 21: Generic Architecture for OADM/OXS Systems


Network elements have intelligent software that senses a module failure or
break in its fiber connection and automatically routes traffic in the opposite direction around the fiber ring. This architecture allows service
providers to guarantee that customers connections will not go out of service, as Figure 21 shows. However, the network elements now support
multiple optical wavelengths as opposed to multiple DS3 circuits. In the
case of a fiber break, the optical network will automatically reroute up to
40 optical signals in less than 50 milliseconds.
Because optical rings are most cost-effective over large networks, the
switching time is critical. One technology planned for implementation is
called network protection equipment (NPE), which significantly reduces the
switching time required in large optical networks. Instead of routing traffic
from network elements adjacent to a fiber cut, the OBLSR using NPE redirects the traffic from the node where it enters the ring. This redirection prevents the traffic from being backhauled across the network, which greatly
improves overall switching time.
Optical Cross-Connect (OXC)
Efficient use of fiber facilities at the optical level obviously becomes critical as
service providers begin to move wavelengths around the world. Routing and
grooming are key areas that must be addressed. This is the function of the
optical cross-connect (OXC), as shown in Figure 22.

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Figure 22. OXC Block Diagram


Digital cross-connect systems are deployed en masse and provide the critical function of grooming traffic (DS0, DS1, and DS3) to fill output ports on the system
efficiently. Today, output ports can be at the DS3, OC3, or OC12 level. For this
reason, it is critical to ensure that those pipes are full of traffic when they exit the
cross-connect system. In the optical domain, where 40 optical channels can be
transported on a single fiber, a network element is needed that can accept various
wavelengths on input ports and route them to appropriate output ports in the network. To accomplish this, the OXC needs three building blocks (See Figure 23).

Figure 23. Optical Cross-Connects


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Fiber switchingthe ability to route all of the wavelengths on an incoming
fiber to a different outgoing fiber
Wavelength switchingthe ability to switch specific wavelengths from an
incoming fiber to multiple outgoing fibers
Wavelength conversionthe ability to take incoming wavelengths and convert them
(on the fly) to another optical frequency on the outgoing port; this is necessary to
achieve strictly nonblocking architectures when using wavelength switching.

12. THE FUTURE OF OPTICAL NETWORKS


Continued advancements in optical technology promise continued change
as the optical network evolves to the ultimate goal of end-to-end wavelength services.
The impact of the new optical layer in the telecommunications network is
astounding. It can be measured in two wayseconomic impact and carriers
ability to offer new services. Optical layer technology will increase network
capacity, allowing network providers to transport more than 40 times the traffic
on the same fiber infrastructure. That will ultimately lead to lower prices, and
competition in the local exchange (as a result of the 1996 Telecommunications
Act) will ensure that bandwidth becomes more affordable.
Consumers will have access to new high-bandwidth services made possible by
the increased capacity afforded by the optical layer. Services that today are
considered prohibitively expensive, such as video conferencing to the desktop
(or home), electronic commerce, and high-speed video imaging, will become
commonplace, because they will be technologically and economically feasible.
In essence, optical layer technology will improve the way we live.

13. SELF-TEST
Multiple Choice
1. Optical networks are based on the emergence of the ______ layer in ______ networks.
a. optical; transport
b. transport; optical
c. optical; integrated
d. optical; high-capacity
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2. In the ______ a revolution in the technology for telecommunications networks began,
spawned by the use of a relatively unassuming technology: ______.
a. early 1990s; fiber-optic cable
b. early 1980s; synchronous cable
c. early 1980s; fiber-optic cable
d. early 1990s; asynchronous cable

3. The one aspect of SONET that has allowed it to survive during a time of tremendous changes in network capacity needs is its ______.
a. versatility
b. scalability
c. fiber capacity
d. functionality

4. One of the great revenue-producing aspects of optical networks is the ability to resell
____ rather than _____.
a. bandwidth; fiber
b. fiber; bandwidth
c. wavelength; bandwidth
d. single fiber; double fiber

5. The first incarnation of WDM was

a. broadband WDM
b. the erbium-doped optical amplifier
c. dense WDM
d. narrowband lasers

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True or False
6. A fiber Bragg grating is a small section of fiber that has been modified to create
periodic changes in the index of refraction.
a. true
b. false

7. One of the most promising technologies for optical networks is the semiconductor
optical amplifier (SOA).
a. true
b. false
8. A wavelength converter provides functionality similar to an electrical switch by routing an incoming wavelength to a variety of physical output ports.
a. true
b. false
9. One of the benefits of a long-haul network is its conservative use of bandwidth.
a. true
b. false

10. The basic element in the optical network is the wavelength.


a. true
b. false

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14. ACRONYM GUIDE


ATM

asynchronous transfer mode

DWDM

dense wavelength division multiplexing

ETDM

electronic time division multiplexing

IOF

interoffice networks

MAN

metropolitan-area network

OADM

optical add/drop multiplexer

OBLSR

optical rings

OSNR

optical signal-to-noise ration

OTDM

optical time division multiplexing

OXC

optical cross-connect

PBST

phase-shaped binary transmission

SOA

semiconductor optical amplifier

SONET

synchronous optical network

TDM

time division multiplexing

VWP`

virtual wavelength path

WAN

wide-area network

WDM

wavelength division multiplexing

WP

wavelength path

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Personal Communications Services


(PCS)
DEFINITION
Personal communications services (PCS) is a new generation of wirelessphone technology that introduces a range of features and services surpassing
those available in analog- and digital-cellular phone systems. PCS provides
the user with an all-in-one wireless phone, paging, messaging, and data service that has a greatly improved battery-standby time.

TUTORIAL OVERVIEW
This tutorial addresses PCS by introducing the IS136 digital control channel
(DCCH) platform and by describing how PCS features operate on the DCCH
air interface.

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TOPICS
1.

OVERVIEW OF PCS TECHNOLOGY . . . . . . . . . . . . . . . . . . . . . . . . .267

2.

THE DCCH ENVIRONMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .269

3.

THE AIR INTERFACE: MULTILAYERED PROTOCOL . . . . . . . . . . . .271

4.

LOGICAL CHANNELS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .273

5.

SLEEP MODE AND STANDBY TIME . . . . . . . . . . . . . . . . . . . . . . . . .276

6.

PCS MESSAGING . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .277

7.

HIERARCHICAL CELL RELATIONSHIPS . . . . . . . . . . . . . . . . . . . . . .280

8.

PUBLIC, PRIVATE, AND RESIDENTIAL SYSTEMS . . . . . . . . . . . . . .283

9.

SYSTEM IDENTITIES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .286

10. SELF-TEST . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .288


11. ACRONYM GUIDE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .290

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1. OVERVIEW OF PCS TECHNOLOGY


The Telecommunications Industry Association (TIA) IS136 specification is
the basis of the time division multiple access (TDMA) PCS air-interface technology. IS136 is designed to operate in both the 800-MHz and the 1900MHz frequency bands, thus providing seamless operation on cellular and PCS
systems.
The Digital Control Channel (DCCH)
The digital control channel forms the core of the IS136 specification and is
the primary enhancement to TDMA digital-wireless technology. It is a new
control-channel mechanism added to the analog control channel (ACC), the
analog voice channel (AVC), and the digital traffic channel (DTC) of the
TDMA air interface. The IS136 DCCH TDMA technology provides the platform for PCS, introducing new functionalities and supporting enhanced features that make PCS a powerful digital system.
Dual-Band Dual-Mode Operation
PCS dual-band phones operating at 800 MHz and 1900 MHz enable users to
receive full PCS features and services for IS136 systems wherever they roam.
The dual-mode capability provides service continuity and interoperability
between analog and digital networks. As a result, a PCS phone can provide
access to all outdoor wireless services, be used in a private in-building system,
and serve as a flat-rate digital cordless phone at home.

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Features and Capabilities
Table 1 shows important PCS features and capabilities.
Table 1. PCS Features and Capabilities

Feature

Capability

Sleep mode

extends phone standby time and


enhances battery life

Short message service


(SMS)

transfers alphanumeric messages to and


from cellular and PCS phones

Voice and data privacy

increases resistance to eavesdropping

Superior voice quality

results in less background noise and


fewer dropped calls

Hierarchical environment provides support for macrocell-microcell


operation
Intelligent rescan

allows tighter control of system selection

Private and residential


system IDs

provides more simplified and controlled


wireless office service (WOS) and
personal base station (PBS) features

Seamless roaming

enables roaming between frequencies


using dual-band phones and provides
support for international roaming

Circuit-switched data
support

provides highly reliable data transmission


for wireless e-mail, faxing, and Internet
access

Authentication

increases phone security and resistance to


cloning

Calling number
identification (CNI)

allows callers to be identified before


answering

Message waiting
indicator (MWI)

notifies users that they have voice mail


messages

Text dispatch service

uses live operators to take caller mes


sages and send text messages to the PCS
phone

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Comparison of Cellular and PCS Spectrums


Figure 1 illustrates the wireless cellular 800-MHz spectrum and the PCS 1900MHz spectrum.

Figure 1. Comparison of Cellular and PCS Spectrums

2. THE DCCH ENVIRONMENT


A radio channel consists of two frequencies within the RF (radio frequency)
spectrum that are separated by a fixed distance. These two frequencies allow
a cell site and wireless phone to transmit and receive signals simultaneously.
Cell sites communicate with wireless phones using two different radio channels: a voice channel and a control channel.
In TDMA systems, each digital radio channel can carry up to three voice calls

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by time-multiplexing voice traffic into time slots. A digital control channel is
introduced into the TDMA system by reprogramming one of those traffic
channels, called digital traffic channels (DTCs), to become the DCCH on a
frequency that contains the existing digital traffic channels.
Figure 2 depicts the DTC slot pair (1, 4) used for a DCCH and shows each cell
divided into sectors (A, B, C). Only one slot pair is required for a DCCH in
each cell sector, regardless of the number of digital radios in the sector.

Figure 2. IS136 DCCH Operation


Operating Principle
Information carried on the DCCH flows in two directions over the air interface: from the system to the phone (downlink) and from the phone to the
system (uplink). In Figure 2, the base station represents the system.
DCCHcapable and PCS phones monitor (camp on) a digital-control channel
in each sector of a wireless system that supports IS136 services. A PCS
phone will scan for this channel, gain synchronization, and begin to decode
the information provided over a broadcast control channel on the DCCH.
The DCCH serves as the phones control channel until the phone finds another cell that is more appropriate.
PCS phones receive pages, send originations, and communicate with the system on the DCCH. After receiving a page or performing a call origination, a
traffic channel is then designated for the call, and the phone will hand off
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from cell to cell as it moves around the system. At call completion, the phone
returns to the DCCH to await further interaction.

3. THE AIR INTERFACE: MULTILAYERED PROTOCOL


The air interface used in PCS is structured in different layers, each with specific purposes. This conceptual split makes it easier to understand the interactions between the base station and the phone across the air interface. There
are four layers:
a physical layer (layer-1), which deals with the radio interface, bursts, slots,
frames, and superframes
a data link layer (layer-2), which handles the data packaging, error correction, and message transport
a message layer (layer-3), which creates and handles messages sent and
received across the air
upper application layers, which represent the teleservice currently being
used, such as voice and messaging transactions, or future services such as
on-air programming
The Air-Interface Model
Figure 3 shows the air-interface model. This structure simplifies the introduction of current and future services using the IS136 digital control channel
platform because the lower layers in the air-interface protocol (the radio interface, data management, messages, and so on) remain unchanged.

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Figure 3. The Air-Interface Model


Operating Principle
Figure 4 shows how one layer-3 message is mapped into several layer-2
frames and how a layer-2 time frame is mapped onto a time slot. The time
slot is further mapped onto a DCCH channel. The figure shows how information is passed from layer to layer down through the stack until a burst is
created, ready for transmission. At the receiving end, information is stripped
off as needed as the message is passed up to the application.
The layer-3 message shown in Figure 4 can be an uplink registration, a downlink PCS message, a page response, or a broadcast message. The layer-3 message is packaged into a layer-2 frame where header and error-correction fields
are added. The packet is then coded and the individual bits interleaved
(mixed and distributed) to counteract errors introduced in the radio environment.

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layer 3 message

Figure 4. Layered 3-2-1 Mapping

4. LOGICAL CHANNELS
Logical channels were developed in the IS136 DCCH technology to organize
the PCS and other digital information flowing across the air interface.
Logical-Channels Configuration
The logical channels are depicted graphically in Figure 5. The figure shows
how the forward DCCH (FDCCH) consists of many logical channels carrying
information from the system to the phone. The reverse DCCH (RDCCH),
carrying information from the phone to the system, consists of one logical
channel.

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Figure 5. Logical Channels Configuration


Operating Principle
Logical channels sort and prioritize signaling information by functional use.
The data is then mapped onto a DCCH, which is a physical channel. Physical
channels are the actual portions of electromagnetic bandwidth consisting of
frequencies and time divisions. Logical-channel data flows on the DCCH in
both directions: from the system to the phone (downlink) and from the
phone to the system (uplink).
Logical Channels Functions
The multiplexed broadcast channel (BCCH) shown in Figure 5 is designed to
carry information about the system configuration and the rules that phones
must follow at system access. Its primary logical channels are the following:

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the fast broadcast channel (FBCCH), carrying information that phones
need immediately, such as the system ID and registration information
the extended broadcast channel (EBCCH), carrying information that is not
as time critical, such as neighbor cell lists
The system uses the multiplexed SMS point-to-point messaging, paging, and
access response channel (SPACH) shown in Figure 5 to communicate with a
specific phone. Its logical channels are the following:
the short message service channel (SMSCH), carrying PCS messaging and
over-the-air activation and programming (OAA/P); PCS information is carried on the logical channels at both 800 MHz and 1900 MHz
the paging channel (PCH), carrying system pages to the phone
the access response channel (ARCH), providing system response to phone
queries and administration information

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Table 2 outlines the logical channels.
Table 2. Description of Logical Channels

Logical Channel

Description

Broadcast channel(BCCH)

This is a downlink multiplexed


channel comprised of the following:
F-BCCHthe fast broadcast channel
E-BCCHthe extended broadcast
channel

SMS point-to-point
messaging, paging, and
access response channel)
(SPACH)

This is a downlink multiplexed channel


comprised of the following:
SMSCHthe SMS messaging channel
PCHthe paging channel
ARCHthe access response channel

Random access channel


(RACH)

This is a single uplink channel with all


time slots used for system access.

Shared channel
feedback (SCF)

The SCF fields in the downlink are used


to provide a collision-prevention
mechanism for the uplink.

5. SLEEP MODE AND STANDBY TIME


PCS uses the digital control channel to provide a sleep mode during which
phones can turn off much of their circuitry until they need to wake up, at predetermined intervals, to receive system messaging. This feature greatly
increases the battery life, thereby increasing the standby time of phones.
Standby time is the time a wireless phone is idle; that is, the phone is on, but
no calls are being placed or received.
Operating Principle
An idle phone camps on the DCCH. The phone checks for incoming calls
every few milliseconds then reenters the sleep mode. This differs from a
phone using an analog control channel (ACC), where an idle phone must
monitor the control channel constantly, wearing down the battery.
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The system messages received by the phone can be pages (for either a voice
call or PCS messaging) or broadcast messages (for example, updates about cell
changes or neighbor lists) carried on the downlink DCCH. The phone must
decode the downlink information only at intervals on its predetermined paging slots or on the broadcast slots if the broadcast information changes. In
this manner, the phone has extended periods of time in which it can power
down some of its circuitry and sleep between paging opportunities.
Current Consumption and Sleep Periods
Figure 6 depicts analog control channel (ACC) versus DCCH battery-current
consumption and indicates the phones sleep-mode periods on the DCCH.
The time spikes in the DCCH segment of the drawing are representative of
the predetermined paging slots.

Figure 6. Current Consumption and Sleep Periods

6. PCS MESSAGING
PCS messaging is a digital short message service (SMS) feature that allows a
wireless phone to receive numeric pages and short text messages. This lets
one device do the work of both pager and phone. Users can receive messages
on their phones display screens from a variety of sources: computers,

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telephones, e-mail, voice mail, and text dispatch (live operators take caller
messages and send text messages to the PCS phone).
PCS uses the DCCH and digital traffic channels to deliver the alphanumeric
messages to and from the wireless phone. The messages are sent and received
via a message center, which is a node on the wireless intelligent network. The
messages contain a variety of attributes controlling their delivery, storage, and
display behavior.
Message Architecture
Each network-originated PCS message consists of two basic elements:
addressing information, which tells the system to which phone the message
is to be delivered
alphanumeric text, comprising the characters that make up the actual text
message attributes, which tell the phone how to handle and display the
message when it is received
Message Types
PCS messaging can deliver numeric callback messages from a phone and
alphanumeric messages sent via modem and computer. Examples of PCS messaging include paging and notification of new voice messages and e-mail messages. Messages of up to 239 characters can be sent over the air interface.
Operating Principle
The PCS messaging feature uses a dedicated paging terminal. When the network receives a PCS message, it locates the target phone and delivers the
message. The phone notifies the user with a message icon, a beep, or both.
The message can then be displayed and read. If users leave a PCS messaging
area, the network stores any messages until they return. The network will
repeatedly try to deliver a message until the phone is able to receive it.

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Message Generation
The following entities can be used for PCS message generation:
networking from existing paging terminals
voice-response unit
live operator text-dispatch service
dial-up modem
e-mail gateway
data information source
voice-mail system
Figure 7 shows a PCS teleservice messaging scheme in which a message is formulated in a PC (personal computer) and sent to the phone of the message
recipient. Phone-screen displays differ depending on model and manufacturer,
but they all show the number of new messages.

Figure 7. PCS Teleservice Messaging Scheme

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Message Delivery
PCS messaging is designed to operate in practical, everyday situations.
Power On
If the phone is powered on, the message is available immediately, just like a
pager.
Phone Engaged
If the phone is engaged in a voice conversation, the network delivers the
message to the phone using the same digital traffic channel being used for
the conversation.
Power Off
If the phone is powered off, or the phone is out of a service area, the network message center stores the message for later delivery. As soon as the
phone is powered on, the messages are delivered. This way, messages are
not missed if a phone is off, out of a service area, or in an area with poor
reception.
Voice Mail
When a caller reaches a users voice mail, the system provides the option to
send a callback-number message to the phone or to send an alphanumeric
message using special message flash software.
Roaming
If the user is roaming in an area not supporting PCS messaging, the message center will store the message and deliver it when the phone reenters a
PCSsupported area.

7. HIERARCHICAL CELL RELATIONSHIPS


Cell sites have traditionally existed as macrocells on towers that cover areas
up to several miles in diameter. Macrocells are typically public cells, serving
all wireless phone users. IS136 DCCH TDMA technology enables the use of
much smaller cells called microcells. Microcells provide customized service
within the coverage of existing macrocells. Microcells typically provide wireless office service (WOS) features to specific phones within a private building
or campus environment.

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Hierarchical Cell Coverage
The combined coverage of both macrocells and microcells is called hierarchical cell coverage, with the microcells creating a second level of coverage under
the existing level. Although macrocells are usually public and microcells are
usually private, they can reverse roles.
For example, a public macrocell can also provide private WOS services to
offices within its coverage area. Conversely, a microcell can provide public
coverage to fill in geographical gaps due to topography or to enhance coverage in high-density areas.
Figure 8 shows a private-system microcell within a public macrocell.

Figure 8. Macrocell/Microcell Configuration

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Hierarchical Cell Structures
In a PCS environment, a geographical area might be covered by a mix of
macrocells and microcells as well as public and private systems. A PCS phone
must therefore assess the most suitable control channel on which to provide
service, even if the signal strength of a neighboring cell is not the highest signal being received by the phone, but is of a sufficient level to provide quality
service. PCS uses hierarchical cell structures (HCS) to accomplish this by identifying neighboring cells as preferred, regular, or nonpreferred.
Preferred Neighbor Cell
A preferred cell has the highest preference. The phone reselects it even if its signal
strength is lower than the serving cell. The main criterion here is that the preferred neighbor cell must have signal strength sufficient to provide quality service.
Regular Neighbor Cell
A regular cell has the second-highest preference. The phone reselects it if the
cells signal strength is greater than the serving cell (plus a hysteresis value)
and if there is no eligible preferred cell available.
Nonpreferred Neighbor Cell
A nonpreferred cell has the lowest preference. The phone reselects it only if
the signal strength of the serving cell becomes insufficient to provide service
and the signal strength of the non-preferred neighbor is greater than the serving cell (plus a hysteresis value).
Operating Principle
Hierarchical cell structures enable the digital control channel (DCCH) to identify and designate neighboring cells as preferred, regular, or nonpreferred. A
PCS phone uses that hierarchical information to reselect a particular neighbor
cell over another, based on the type of relationship defined between the cell it
is using (serving cell) and the adjacent neighbor cell. Each neighbor cells designation dictates which type of algorithm the phone uses when it considers
the cell as a reselection candidate.

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For example, when a low-power microcell is providing capacity in a dense
traffic area that is also served by a high-power macrocell, the hierarchical cell
structure allows the phone to give preference to the weaker microcell.
Without the multitier environment, the phones would have difficulty capturing microcells, and the cellular system would require highly specific parameter settings.
Figure 9 shows reselection based on HCS cell-type designation.
Macrocell 2

Macrocell 1

SS SUFF

Select Macrocell 1

Office
Building
Macrocell
Select Office-Macrocell

SS SUFF

Select Macrocell 2

KEY:
Neighbor Cell Relationship; P = Preferred; R = Regular; NP = Non-Preferred
SS SUFF = Signal Strength Sufficient Parameter

Figure 9. Reselection Based on HCS Cell Type

8. PUBLIC, PRIVATE, AND RESIDENTIAL SYSTEMS


PCS phones can behave differently according to the type of system providing
service to the user. For example, phones providing only basic service might
not reselect or camp on private cells, thereby improving their time to service.
Similarly, phones providing service on a residential system, such as a personal
base station, might perform different scanning routines to find their home
system.

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Operating Principle
PCS uses IS136 identity structures to categorize each cell into three basic
network typespublic, private, or residentialand allows the phone to react
to serving cells based on the broadcast identifiers of those network types. In
other words, the phone can discriminate between and access different network systems and distinguish the types of services available on particular
cells. Because a cell can have a mixture of network types and subtypes, it can
have a mixture of services.
Figure 10 shows some network system configurations.

Figure 10. Network System Configurations

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Network Types
Designations for the major network types and the subtypes are the following:
Public
The public designation refers to cells that provide the same basic cellular
service to all customers.
Private
These cells provide special services to a predefined group of private or wireless office service customers only, and do not support public use of that cell.
The private designation is used for in-building company systems with specific features.
Semi-Private
A subtype, these cells provide basic service to all customers and also provide special services to a predefined group of private customers. An example would be a cell providing service to a wireless office service system, as
well as to public users.
Residential
These cells provide special services to a predefined group of residential customers only and do not support public use of the cell. The personal base
station that allows a cellular phone to behave like a cordless home phone is
classed as a residential system.
Semi-Residential
A subtype, these cells provide basic service to all customers and also provide special services to a predefined group of residential customers. This
type is used in a neighborhood where the public macrocell also provides
residential cellular service.
Autonomous
These are cells that broadcast a DCCH in the same geographical area as
other DCCH systems but are not listed as a neighbor on the neighbor list of
the public system. Examples of autonomous systems include the personal
base station and private systems that are not coordinated with the public
system. Phones must perform special frequency-scanning algorithms to find
autonomous cells.

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9. SYSTEM IDENTITIES
A system-identity structure allows PCS phones to distinguish among public,
private, semiprivate, and personal base stations. This IS136 feature facilitates
the creation of private systems and allows control of phone behavior around a
wireless office service, personal base station, or residential service area. The
IS136 technology includes private-system identifiers for marking specific base
stations as part of a private system, hierarchical cell structures for defining cell
preferences, and new registration features to complement private systems.
Operating Principle
A private system identity (PSID) is assigned to a specific private system by
the system operator to identify it to phones in the coverage area of the system. PSIDs are broadcast so that a phone can determine whether it has special services from a particular cell when reselecting a digital control channel.
PSIDs can be assigned on a sector-by-sector basis,which allows very small
service areas to be defined. Alternatively, many cells, as well as systems,
could broadcast the same PSID to create a geographically large virtual private
system. Phones that recognize PSIDs notify the system and can activate location ID to inform users that they have entered the private system.
A single DCCH can broadcast up to 16 PSIDs, allowing the support of up to
16 different private systems on one DCCH. This feature is useful in a technology park or campus where it would not be economical to support a
DCCH for each small business requiring wireless office service features.
Residential System Identities (RSIDs)
In a manner similar to a PSID, a residential system identity (RSID) identifies
a residential system within the public cellular and PCS coverage. RSIDs can
be used to create residential-service areas or neighborhood residential systems by broadcasting an identifier that is recognized by phones as being at
home and therefore receiving special services (for example, billing). A primary use of RSIDs is in the personal base station (PBS), which allows a cellular or PCS phone to be used like a cordless phone in conjunction with a
residential base station.

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Location ID
All PCS phones display the name of the wireless carrier providing service. If a
phone also has wireless office service coverage, the location ID feature can
display a company name (see Figure 11) or a system banner to inform subscribers that they have entered their private system. This can be particularly
important when there is a billing or service difference that should be indicated to the subscriber. The identifying name or banner is removed from the display when the subscriber leaves the wireless office service coverage. A nonsubscriber entering a WOS service area would continue to have only the
wireless carrier name displayed.
Figure 11 shows some examples of location ID for private systems and the
wireless carrier.

Figure 11. System Identities and Location ID


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10. SELF-TEST
Multiple Choice
1.

How many frequencies are in a radio channel?


a. 1
b. 2
c. 3
d. 4

2.

How many layers are there to the PCS air interface?


a. 1
b. 2
c. 3
d. 4

3.

What is the maximum possible number of characters in a PCS message?


a. 64
b. 256
c. 1021
d. 239

4.

Which of the following is NOT an example of a PCSenabled service?


a. automatic call back
b. message waiting indicator (MWI)
c. caller ID
d. short messaging service (SMS)

5.

Which kind of cells typically provide WOS services?


a. microcells
b. macrocells

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6.

Which of the following network types does not provide service to the general
public?
a. residential
b. public
c. private
d. autonomous

True or False
7.

PCS phones do not scan constantly for incoming calls.


a. true
b. false

8.

DCCH information is confined to the 21-analog control channels.


a. true
b. false

9.

The lower levels in the IS136 platform are changed.


a. true
b. false

10. There is one logical channel carrying information from PCS phones to their system.
a. true
b. false

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11. ACRONYM GUIDE


ACC

analog control channel

ARCH

access response channel

AVC

analog voice channel

BCCH

broadcast channel

CNI

calling number identification

DCCH

digital control channel

DTC

digital traffic channel

E-BCCH extended broadcast channel


F-BCCH fast broadcast channel
FDCCH forward DCCH
HCS

hierarchical cell structure

MWI

message waiting indicator

OAA/P

over-the-air activation and programming

PBS

personal base station

PCH

paging channel

PCS

personal communications services

PSID

private system identity

RACH

random access channel

RDCCH reverse DCCH


RF

radio frequency

RSID

residential system identity

SCF

shared channel feedback

SMSCH short message service channel


SPACH

SMS point-to-point messaging, paging, and access response channel

TDMA

time division multiple access

TIA

Telecommunications Industry Association

TNPP

telocator network paging protocol

WOS

wireless office services

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Signaling System 7 (SS7)


DEFINITION
Signaling system 7 (SS7) is an architecture for performing out-of-band signaling in support of the call-establishment, billing, routing, and informationexchange functions of the public switched telephone network (PSTN). It identifies functions to be performed by a signaling system network and a protocol
to enable their performance.

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TOPICS
1.

WHAT IS SIGNALING? ..........................................................................293

2.

WHAT IS OUT-OF-BAND SIGNALING? ..............................................293

3.

SIGNALING NETWORK ARCHITECTURE ..........................................294

4.

THE NORTH AMERICAN SIGNALING ARCHITECTURE ................295

5.

BASIC SIGNALING ARCHITECTURE ..................................................298

6.

SS7 LINK TYPES ......................................................................................299

7.

BASIC CALL SET UP EXAMPLE ............................................................301

8.

DATABASE QUERY EXAMPLE ..............................................................303

9.

LAYERS OF THE SS7 PROTOCOL ........................................................305

10 WHAT GOES OVER THE SIGNALING LINK........................................308


11. ADDRESSING IN THE SS7 NETWORK ................................................309
12. SIGNAL UNIT STRUCTURE ..................................................................310
13. WHAT ARE THE FUNCTIONS OF THE DIFFERENT
SIGNALING UNITS? ................................................................................311
14. MESSAGE SIGNAL UNIT STRUCTURE ................................................312
15. SELF-TEST ................................................................................................314
16. ACRONYM GUIDE..................................................................................318

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1. WHAT IS SIGNALING?
Signaling refers to the exchange of information between call components
required to provide and maintain service.
As users of the public switched telephone network, we exchange signaling
with network elements all the time. Examples of signaling between a telephone user and the telephone network include dialing digits, providing dial
tone, accessing a voice mailbox, sending a call-waiting tone, dialing *66 (to
retry a busy number), etc.
SS7 is a means by which elements of the telephone network exchange information. Information is conveyed in the form of messages. SS7 messages can
convey information such as the following:
Im forwarding to you a call placed from 212-555-1234 to 718-555-5678. Look
for it on trunk 067.
Someone just dialed 800-555-1212. Where do I route the call?
The called subscriber for the call on trunk 11 is busy. Release the call and play
a busy tone.
The route to XXX is congested. Please dont send any messages to XXX
unless they are of priority 2 or higher.
Im taking trunk 143 out of service for maintenance.
SS7 is characterized by high-speed packet data, and out-of-band signaling.

2. WHAT IS OUT-OF-BAND SIGNALING?


Out-of-band signaling is signaling that does not take place over the same path
as the conversation.
We are used to thinking of signaling as being in-band. We hear dial tone, then
dial digits, and hear ringing over the same channel on the same pair of wires.
When the call completes, we talk over the same path that was used for the signaling. Traditional telephony used to work in this way as well. The signals to
set up a call between one switch and another always took place over the same
trunk that would eventually carry the call. Signaling took the form of a series
of multifrequency (MF) tones, much like touch tone dialing between switches.

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Out-of-band signaling establishes a separate digital channel for the exchange
of signaling information. This channel is called a signaling link. Signaling links
are used to carry all the necessary signaling messages between nodes. Thus,
when a call is placed, the dialed digits, trunk selected, and other pertinent
information are sent between switches using their signaling links, rather than
the trunks which will ultimately carry the conversation. Today, signaling links
carry information at a rate of 56 or 64 kilobits per second (kbps).
It is interesting to note that while SS7 is only used for signaling between network elements, the ISDN D channel extends the concept of out-of-band signaling to the interface between the subscriber and the switch. With ISDN service, signaling that must be conveyed between the user station and the local
switch is carried on a separate digital channel called the D channel. The voice
or data which comprise the call is carried on one or more B channels.
Why Out-of-Band Signaling?
Out-of-band signaling has several advantages that make it more desirable
than traditional in-band signaling:
It allows for the transport of more data at higher speeds (56 kbps can carry
data much faster than MF outpulsing).
It allows for signaling at any time during the call, not only at the beginning.
It enables signaling to network elements to which there is no
direct trunk connection.

3. SIGNALING NETWORK ARCHITECTURE


If signaling is to be carried on a different path from the voice and data traffic
it supports, then what should that path look like?
The simplest design would be to allocate one of the paths between each
interconnected pair of switches as the signaling link. Subject to capacity constraints, all signaling traffic between the two switches could traverse this link.

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This type of signaling is known as associated signaling, and is shown below
in Figure 1.

Figure 1. Associated Signaling


Associated signaling works well as long as a switchs only signaling requirements are between itself and other switches to which it has trunks. If call set
up and management was the only application of SS7, associated signaling
would meet that need simply and efficiently. In fact, much of the out-of-band
signaling deployed in Europe today uses associated mode.
The North American implementers of signaling system 7, however, wanted to
design a signaling network that would enable any node to exchange signaling
with any other SS7capable node. Clearly, associated signaling becomes
much more complicated when it is used to exchange signaling between nodes
that do not have a direct connection.
From this need, the North American signaling system 7 architecture was born.

4. THE NORTH AMERICAN SIGNALING ARCHITECTURE


The North American signaling architecture defines a completely new and separate signaling network. The network is built out of three essential components, interconnected by signaling links. These components are signal switching points (SSPs), signal transfer points (STPs), and signal control points
(SCPs). They are outlined in Table 1 below.

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Component

Function

Signal switching points (SSPs)

SSPs are telephone switches (end offices or


tandems) equipped with SS7-capable software and terminating signaling links. They
generally originate, terminate, or switch calls.

Signal transfer points (STPs)

STPs are the packet switches of the SS7 network. They receive and route incoming signaling messages toward the proper destination. They also perform specialized routing
functions.

Signal control points (SCPs)

SCPs are databases that provide information


necessary for advanced call-processing
capabilities.

Table 1. North American Signaling Architecture Components


Once deployed, the availability of the SS7 network is critical to call processing. Unless SSPs can exchange signaling, they cannot complete any interswitch calls. For this reason, the SS7 network is built using a highly redundant architecture. Each individual element must also meet exacting requirements for availability. Finally, protocol has been defined between interconnected elements to facilitate the routing of signaling traffic around any difficulties that may arise in the signaling network.
To enable signaling-network architectures to be easily communicated and
understood, a standard set of symbols was adopted for depicting SS7 networks. Figure 2 shows the symbols that are used to depict these three key elements of any SS7 network.

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Figure 2. Signaling Network Elements


STPs and SCPs are customarily deployed in pairs. While elements of a pair are
not generally co-located, they work redundantly to perform the same logical
function. When drawing complex network diagrams, these pairs may be
depicted as a single element for simplicity, as shown in Figure 3.

Figure 3. STP and SCP Pairs

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5. BASIC SIGNALING ARCHITECTURE


Figure 4 shows a small example of how the basic elements of an SS7 network
are deployed to form two interconnected networks.

Figure 4: Sample Network


Several points should be noted:
1. STPs W and X perform identical functions. They are redundant. Together,
they are referred to as a mated pair of STPs. Similarly, STPs Y and Z form a
mated pair.
2. Each SSP has two links (or sets of links), one to each STP of a mated pair.
All SS7 signaling to the rest of the world is sent out over these links.
Because the STPs of a mated pair are redundant, messages sent over either
link (to either STP) will be treated equivalently.
3. The STPs of a mated pair are joined by a link (or set of links).
4. Two mated pairs of STPs are interconnected by four links (or sets of links).
These links are referred to as a quad.

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5. SCPs are usually (though not always) deployed in pairs. As with STPs, the
SCPs of a pair are intended to function identically. Pairs of SCPs are also
referred to as mated pairs of SCPs. Note that they are not directly joined by
a pair of links.
Signaling architectures such as this, which provide indirect signaling paths
between network elements, are referred to as providing quasiassociated signaling.

6. SS7 LINK TYPES


SS7 signaling links are characterized according to their use in the signaling
network. Virtually all links are identical in that they are 56kbps or 64kbps
bidirectional data links that support the same lower layers of the protocol;
what is different is their use within a signaling network. The defined link
types are shown in Figure 5 below and defined as follows.

Figure 5. Link Types

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A Links
A links interconnect an STP and either an SSP or an SCP, which are collectively referred to as signaling end points (A is intended to stand for access). A
links are used for the sole purpose of delivering signaling to or from the signaling end points (they could just as well be referred to as signaling beginning
points). Examples of A links are 2-8, 3-7, and 5-12 in Figure 5.
Signaling that an SSP or SCP wishes to send to any other node is sent on either
of its A links to its home STP, which, in turn, processes or routes the messages.
Similarly, messages intended for an SSP or SCP will be routed to one of its
home STPs, which will forward them to the addressed node over its A links.
B Links, D Links, and B/D Links
Links interconnecting two mated pairs of STPs are referred to as either B
links, D links, or B/D links. Regardless of their name, their function is to carry
signaling messages beyond their initial point of entry to the signaling network
toward their intended destination. The B stands for bridge and is intended
to describe the quad of links interconnecting peer pairs of STPs. The D
denotes diagonal and is intended to describe the quad of links interconnecting
mated pairs of STPs at different hierarchical levels. Because there is no clear
hierarchy associated with a connection between networks, interconnecting
links are referred to as either B, D, or B/D links. (7-11 and 7-12 are examples
of B links; 8-9 and 7-10 are examples of D links; 10-13 and 9-14 are examples
of interconnecting links and can be referred to as B, D, or B/D links.)

C Links
C links are links that interconnect mated STPs. As will be seen later, they are used
to enhance the reliability of the signaling network in instances where one or several links are unavailable. C stands for cross; 7-8, 9-10, and 11-12 are C links.

E Links
While an SSP is connected to its home STP pair by a set of A links,
enhanced reliability can be provided by deploying an additional set of links to

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a second STP pair. These links, called E (extended) links, provide backup
connectivity to the SS7 network in the event that the home STPs cannot be
reached via the A links. While all SS7 networks include A, B/D, and C
links, E links may or may not be deployed at the discretion of the network
provider. The decision of whether or not to deploy E links can be made by
comparing the cost of deployment with the improvement in reliability; 1-11
and 1-12 are E links.
F Links
F (fully associated) links are links that directly connect two signaling end
points. F links allow associated signaling only. Because they bypass the security features provided by an STP, F links are not generally deployed between
networks. Their use within an individual network is at the discretion of the
network provider; 1-2 is an F link.

7. BASIC CALL SET-UP EXAMPLE


Before going into much more detail, it might be helpful to look at several
basic calls and the way they use SS7 signaling (see Figure 6).

Figure 6. Call Set-Up Example

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In this example, a subscriber on switch A places a call to a subscriber on
switch B:
1. Switch A analyzes the dialed digits and determines that it needs to send
the call to switch B.
2. Switch A selects an idle trunk between itself and switch B and formulates an
initial address message (IAM), the basic message necessary to initiate a call.
The IAM is addressed to switch B. It identifies the initiating switch (switch
A), the destination switch (switch B), the trunk selected, the calling and called
numbers, as well as other information beyond the scope of this example.
3. Switch A picks one of its A links (say AW) and transmits the message over
the link for routing to switch B.
4. STP W receives a message, inspects its routing label, and determines that it
is to be routed to switch B. It transmits the message on link BW.
5. Switch B receives the message. On analyzing the message, it determines
that it serves the called number and that the called number is idle.
6. Switch B formulates an address complete message (ACM), which indicates
that the IAM has reached its proper destination. The message identifies the
recipient switch (A), the sending switch (B), and the selected trunk.
7. Switch B picks one of its A links (say BX) and transmits the ACM over the
link for routing to switch A. At the same time, it completes the call path in
the backward direction (toward switch A), sends a ringing tone over that
trunk toward switch A, and rings the line of the called subscriber.
8. STP X receives the message, inspects its routing label, and determines that
it is to be routed to switch A. It transmits the message on link AX.
9. On receiving the ACM, switch A connects the calling subscriber line to the
selected trunk in the backward direction (so that the caller can hear the
ringing sent by switch B).
10. When and/or if the called subscriber picks up the phone, switch B formulates an answer message (ANM), identifying the intended recipient switch
(A), the sending switch (B), and the selected trunk.

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11. Switch B selects the same A link it used to transmit the ACM (link BX)
and sends the ANM. By this time, the trunk must also be connected to the
called line in both directions (to allow conversation).
12. STP X recognizes that the ANM is addressed to switch A and forwards it
over link AX.
13. Switch A ensures that the calling subscriber is connected to the outgoing
trunk (in both directions) and that conversation can take place.
14. If the calling subscriber hangs up first (following the conversation), switch
A will generate a release message (REL) addressed to switch B, identifying
the trunk associated with the call. It sends the message on link AW.
15. STP W receives the REL, determines that it is addressed to switch B, and
forwards it using link WB.
16. Switch B receives the REL, disconnects the trunk from the subscriber line,
returns the trunk to idle status, generates a release complete message
(RLC) addressed back to switch A, and transmits it on link BX. The RLC
identifies the trunk used to carry the call.
17. STP X receives the RLC, determines that it is addressed to switch A, and
forwards it over link AX.
18. On receiving the RLC, switch A idles the identified trunk.

8. DATABASE QUERY EXAMPLE


People generally are familiar with the toll-free aspect of 800 (or 888) numbers,
but these numbers have significant additional capabilities made possible by
the SS7 network. 800 numbers are virtual telephone numbers. Although they
are used to point to real telephone numbers, they are not assigned to the subscriber line itself.
When a subscriber dials an 800 number, it is a signal to the switch to suspend
the call and seek further instructions from a database. The database will provide either a real phone number to which the call should be directed or it will
identify another network (e.g., a long-distance carrier) to which the call
should be routed for further processing. While the response from the database
could be the same for every call (as if, for example, you have a personal 800

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number), it can be made to vary based on the calling number, the time of day,
the day of the week, or a number of other factors.
The following example shows how an 800 call is routed (see Figure 7).

Figure 7. Database Query Example


1. A subscriber served by switch A wants to reserve a rental car at a companys nearest location. She dials the companys advertised 800 number.
2. When the subscriber has finished dialing, switch A recognizes that this is
an 800 call and that it requires assistance to handle it properly.
3. Switch A formulates an 800 query message including the calling and called
number and forwards it to either of its STPs (e.g., X) over its A link to that
STP (AX).
4. STP X determines that the received query is an 800 query and selects a
database suitable to respond to the query (e.g., M).
5. STP X forwards the query to SCP M over the appropriate A link (MX).
6. SCP M receives the query, extracts the passed information, and (based on
its stored records) selects either a real telephone number or a network (or
both) to which the call should be routed.

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7. SCP M formulates a response message with the information necessary to
properly process the call, addresses it to switch A, picks an STP and an A
link to use (e.g., MW), and routes the response.
8. STP W receives the response message, recognizes that it is addressed to
switch A, and routes it to A over AW.
9. Switch A receives the response and uses the information to determine
where the call should be routed. It then picks a trunk to that destination,
generates an initial address message (IAM), and proceeds (as it did in the
previous example) to set up the call.

9. LAYERS OF THE SS7 PROTOCOL


As the call-flow examples show, the SS7 network is an interconnected set of
network elements used to exchange messages in support of telecommunications functions. The SS7 protocol is designed to both facilitate these functions
and to maintain the network over which they are provided. Like most modern protocols, the SS7 protocol is layered.
The underlying layers of the SS7 protocol are as follows:
Physical LayerLevel 1
This defines the physical and electrical characteristics of the signaling links of the
SS7 network. Signaling links utilize DS0 channels and carry raw signaling data at
a rate of 56 kbps or 64 kbps (56 kbps is the more common implementation).
Message Transfer PartLevel 2
The Level 2 portion of the message transfer part (MTP Level 2) provides linklayer functionality. It ensures that the two end points of a signaling link can
reliably exchange signaling messages. It incorporates such capabilities as error
checking, flow control, and sequence checking.
Message Transfer PartLevel 3
The Level 3 portion of the message transfer part (MTP Level 3) extends the functionality provided by MTP Level 2 to provide network layer functionality. It
ensures that messages can be delivered between signaling points across the SS7
network regardless of whether they are directly connected. It includes such
capabilities as node addressing, routing, alternate routing, and congestion control. Collectively, MTP Levels 2 and 3 are referred to as the message transfer
part (MTP).
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Signaling Connection Control Part
The signaling connection control part (SCCP) provides two major functions that
are lacking in the MTP. The first of these is the capability to address applications
within a signaling point. The MTP can only receive and deliver messages from a
node as a whole; it does not deal with software applications within a node.
While MTP network management messages and basic call-set-up messages
are addressed to a node as a whole, other messages are used by separate
applications (referred to as subsystems) within a node. Examples of subsystems are 800-call processing, calling-card processing, advanced intelligent network (see the Bell Atlantic Web ProForum tutorial), and custom local areas
signaling services (CLASS) (e.g., repeat dialing and call return). The SCCP
allows these subsystems to be addressed explicitly.
Global Title Translation
The second function provided by the SCCP is the ability to perform incremental routing using a capability called global title translation. Global title
translation frees originating signaling points from the burden of having to
know every potential destination to which they might have to route a message. A switch can originate a query, for example, and address it to an STP
along with a request for global title translation. The receiving STP can then
examine a portion of the message, make a determination as to where the
message should be routed, and then route it.
For example, calling-card queries (used to verify that a call can be properly
billed to a calling card) must be routed to an SCP designated by the company
that issued the calling card. Rather than maintaining a nationwide database of
where such queries should be routed (based on the calling-card number),
switches generate queries addressed to their local STPs, which, using global
title translation, select the correct destination to which the message should be
routed. Note that there is no magic here; STPs must maintain a database that
enables them to determine where a query should be routed. Global title translation effectively centralizes the problem and places it in a node (the STP) that
has been designed to perform this function.

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In performing global title translation, an STP does not need to know the exact
final destination of a message. It can, instead, perform intermediate global
title translation, in which it uses its tables to find another STP farther along
the route to the destination. That STP, in turn, can perform final global title
translation, routing the message to its actual destination.
Intermediate global title translation minimizes the need for STPs to maintain
extensive information about nodes that are far removed from them. Global
title translation is also used at the STP to share load among mated SCPs in
both normal and failure scenarios. In these instances, when messages arrive at
an STP for final global title translation and routing to a database, the STP can
select from among available redundant SCPs. It can select an SCP on either a
priority basis (referred to as primary backup) or to equalize the load across all
available SCPs (referred to as load sharing).
ISDN User Part (ISUP)
The ISDN user part defines the messages and protocol used in the establishment and tear down of voice and data calls over the public switched network,
and the management of the trunk network on which they rely. Despite its
name, ISUP is used for both ISDN and nonISDN calls. In the North
American version of SS7, ISUP messages rely exclusively on MTP to transport
messages between concerned nodes.
Transaction Capabilities Application Part (TCAP)
The transaction capabilities application part defines the messages and protocol used to communicate between applications (deployed as subsystems) in
nodes. It is used for database services such as calling card, 800, and AIN as
well as switch-to-switch services including repeat dialing and call return.
Because TCAP messages must be delivered to individual applications within
the nodes they address, they use the SCCP for transport.
Operations, Administration, and Maintenance Part (OAM&P)
The operations, administration, and maintenance part defines messages and
protocol designed to assist administrators of the SS7 network. To date, the
most fully developed and deployed of these capabilities are procedures for

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validating network routing tables and for diagnosing link troubles. OMA&P
includes messages that use both the MTP and SCCP for routing.

10. WHAT GOES OVER THE SIGNALING LINK


Signaling information is passed over the signaling link in messages, called signal units (SUs).
Three types of signal units are defined in the SS7 protocol:
message signal units (MSUs)
link status signal units (LSSUs)
fill-in signal units (FISUs)
Signal units are transmitted continuously in both directions on any link that is
in service. A signaling point that does not have MSUs or LSSUs to send will
send FISUs over the link. The FISUs perform the function suggested by their
name; they fill in the signaling link until there is a need to send purposeful
signaling. They also facilitate link transmission monitoring and the acknowledgment of other SUs.
All transmission on the signaling link is broken up into 8-bit bytes, referred to
as octets. Signal units on a link are delimited by a unique 8-bit pattern known
as a flag. The flag is defined as the 8-bit pattern 01111110.
Because of the possibility that data within a signal unit would contain this
pattern, bit-manipulation techniques are used to ensure that the pattern does
not occur within the message as it is transmitted over the link. (The signal
unit is reconstructed once it has been taken off the link, and any bit manipulation is reversed.) Thus, any occurrence of the flag on the link indicates the
end of one signal unit and the beginning of another. While in theory two flags
could be placed between SUs (one to mark the end of the current message
and one to mark the start of the next message), in practice a single flag is used
for both purposes.

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11. ADDRESSING IN THE SS7 NETWORK


Every network must have an addressing scheme, and the SS7 network is no
different. Network addresses are required so that a node can exchange signaling nodes to which it does not have a physical signaling link. In SS7, addresses are assigned using a three-level hierarchy. Individual signaling points are
identified as belonging to a cluster of signaling points. Within that cluster,
each signaling point is assigned a member number. Similarly, a cluster is
defined as being part of a network. Any node in the American SS7 network
can be addressed by a three-level number defined by its network, cluster, and
member numbers. Each of these numbers is an 8-bit number and can assume
values from 0 to 255. This three-level address is known as the point code of
the signaling point.
Network numbers are assigned on a nationwide basis by a neutral party.
Regional Bell operating companies (RBOCs), major independent telephone
companies, and interexchange carriers already have network numbers
assigned. As network numbers are a relatively scarce resource, companies
networks are expected to meet certain size requirements to be assigned a
network number. Smaller networks can be assigned one or more cluster
numbers within network numbers 1, 2, 3, and 4. The smallest networks are
assigned point codes within network number 5. The cluster to which they
are assigned is determined by the state in which they are located. The network number 0 is not available for assignment and network number 255 is
reserved for future use.
In short, point code is the term used to describe the three-level address number
created by combining the network, cluster, and member numbers. A point
code uniquely identifies a signaling point within the American SS7 network
and is used whenever it is necessary to address that signaling point.

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12. SIGNAL UNIT STRUCTURE


Signal units of each type follow a format unique to that type. A high-level
view of those formats is shown in Figure 8.

Figure 8. Signaling Unit Formats


All three SU types have a set of common fields that are used by MTP Level 2.
They are as follows:
Flag: Flags delimit SUs. A flag marks the end of one SU and the start of the next.
Checksum: The checksum is an 8-bit sum intended to verify that the SU
has passed across the link error-free. The checksum is calculated from the
transmitted message by the transmitting signaling point and inserted in the
message. On receipt, it is recalculated by the receiving signaling point. If the
calculated result differs from the received checksum, the received SU has
been corrupted. A retransmission is requested.
Length Indicator: The length indicator indicates the number of octets
between itself and the checksum. It serves both as a check on the integrity
of the SU and as a means of discriminating between different types of SUs
at Level 2. As can be inferred from Figure 8, FISUs have a length indicator of
0; LSSUs have a length indicator of 1 or 2 (currently all LSSUs have a length

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indicator of 1), and MSUs have a length indicator greater than 2. According
to the protocol, only 6 of the 8 bits in the length-indicator field are actually
used to store this length; thus, the largest value that can be accommodated
in the length indicator is 63. For MSUs with more than 63 octets following
the length indicator, the value of 63 is used.
BSN/BIB FSN/FIB: These octets hold the backward sequence number (BSN),
the backward indicator bit (BIB), the forward sequence number (FSN), and the
forward indicator bit (FIB). These fields are used to confirm receipt of SUs and
to ensure that they are received in the order in which they were transmitted.
They are also used to provide flow control. MSUs and LSSUs, when transmitted, are assigned a sequence number that is placed in the forward sequence
number field of the outgoing SU. This SU is stored by the transmitting signaling point until it is acknowledged by the receiving signaling point.
As the 7 bits allocated to the forward sequence number can store 128 distinct
values, it follows that a signaling point is restricted to sending 128 unacknowledged SUs before it must await an acknowledgment. By acknowledging an SU, the receiving node frees that SUs sequence number at the transmitting node, making it available for a new outgoing SU. Signaling points
acknowledge receipt of SUs by placing the sequence number of the last correctly received and in-sequence SU in the backward sequence number of
every SU they transmit. In that way, they acknowledge all previously
received SUs as well. The forward and backward indicator bits are used to
indicate sequencing or data-corruption errors and to request retransmission.

13. WHAT ARE THE FUNCTIONS OF THE DIFFERENT


SIGNALING UNITS?
FISUs themselves have no information payload. Their purpose is to occupy
the link at those times when there are no LSSUs or MSUs to send. Because
they undergo error checking, FISUs facilitate the constant monitoring of link
quality in the absence of signaling traffic. FISUs can also be used to acknowledge the receipt of messages using the backward sequence number and backward indicator bit.
LSSUs are used to communicate information about the signaling link between
the nodes on either end of the link. This information is contained in the status field of the SU (see Figure 8). Because the two ends of a link are controlled

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by independent processors, there is a need to provide a means for them to
communicate. LSSUs provide the means for performing this function. LSSUs
are used primarily to signal the initiation of link alignment, the quality of
received signaling traffic, and the status of the processors at either end of the
link. Because they are sent only between the signaling points at either end of
the link, LSSUs do not require any addressing information.
MSUs are the workhorses of the SS7 network. All signaling associated with call
set up and tear down, database query and response, and SS7 network management takes place using MSUs. MSUs are the basic envelope within which all
addressed signaling information is placed. As will be shown below, there are
several different types of MSUs. All MSUs have certain fields in common. Other
fields differ according to the type of message. The type of MSU is indicated in
the service-information octet shown in Figure 8; the addressing and informational content of the MSU is contained in the signaling information field.

14. MESSAGE SIGNAL UNIT STRUCTURE


The functionality of the message signal unit lies in the actual content of the
service information octet and the signaling information field (see Figure 8).
The service information octet is an 8-bit field (as might be inferred from its
name) that contains three types of information as follows:
1. Four bits are used to indicate the type of information contained in the signaling information field. They are referred to as the service indicator. The
values most commonly used in American networks are outlined in Table 2.
Value

Function

0
1

signaling network management


signaling network testing and maintenance
signaling connection control part
(SCCP)
ISDN user part (ISUP)

3
5

Table 2. Common Signaling Indicator Values

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2. Two bits are used to indicate whether the message is intended (and coded)
for use in a national or international network. Messages are generally coded
with a value of 2, national network.
3. The remaining 2 bits are used (in American networks) to identify a message
priority, from 0 to 3, with 3 being the highest priority. Message priorities do
not control the order in which messages are transmitted; they are only used
in cases of signaling network congestion. In that case, they indicate whether
a message has sufficient priority to merit transmission during an instance of
congestion and whether it can be discarded en route to a destination.
The format of the contents of the signaling information field is determined by
the service indicator. (Within user parts, there are further distinctions in message formats, but the service indicator provides the first piece of information
necessary for routing and/or decoding the message.)
The first portion of the signaling information field is identical for all MSUs
currently in use. It is referred to as the routing label. Simply stated, the routing label identifies the message originator, the intended destination of the
message, and a field referred to as the signaling-link selection field, which is
used to distribute message traffic over the set of possible links and routes.
The routing label consists of 7 octets that are outlined below in Table 3 (in
order of transmission):
Octet Group

Function

Number of
Octets Involved

destination point code


(DPC)

contains the address of the


node to which the message is
being sent

3 octets

originating point code


(OPC)

contains the address of


message originator

3 octets

signaling link selection

distributes load
among redundant routes

(SLS)

1 octet

Point codes consist of the three-part identifier (network #, cluster #, member


#), which uniquely identifies a signaling point.
Table 3. Routing Label

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15. SELF-TEST
Questions 14: Fill-in-the-Blank
1. The signaling of SS7 features _____ signaling.
a. in-band
b. out-of-band
2. SS7 _____ multifrequency tones for signaling.
a. uses
b. does not use
3. Associated signaling _____ for signaling to network elements to which there is no
direct trunk connection.
a. allows
b. does not allow
4. Signal switching points (SSPs) _____ of the SS7 network.
a. are the packet switches
b. are not the packet switches
Questions 5-7: Identify Elements of This Figure

Element
Symbol 1

5. Element symbol one is

Element
Symbol 2

Element
Symbol 3

a. SSP
b. STP
c. SCP

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6. Element symbol two is

a. SSP
b. STP
c. SCP
7. Element symbol three is

a. SSP
b. STP
c. SCP
Questions 8-16: True or False
8. STPs and SCPs are not customarily deployed in pairs.
a. true
b. false
9. SCPs are always deployed in pairs.
a. true
b. false
10. A Links connect an STP and either an SSP or an SCP.
a. true
b. false
11. An ACM indicates that an IAM has reached the called subscribers switch and
that the subscriber is not busy.
a. true
b. false
12. An initial address message (IAM) indicates that a call has reached its
proper destination.
a. true
b. false

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13. When a called subscriber picks up the phone, his or her switch sends an answer
message (ANM) to indicate that the trunks should be connected in both directions.
a. true
b. false
14. An REL is a message that indicates that switches at both ends have released the
trunk.
a. true
b. false
15. All 800 numbers are assigned to a subscriber line themselves.
a. true
b. false
16. Dialing an 800 number causes the switch to suspend a call and query a database
for further instructions.
a. true
b. false
Questions 17-20: Multiple Choice
17. Links which connect nonmated STP pairs in different networks may be referred to
as
.
a. B links
b. D links
c. B/D links
d. any of the above
18. MTP management messages and ISUP call set-up messages are addressed to
a. separate applications
b. a signal connection control port
c. B links
d. a node as a whole

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19. The SCCP supports

a. routing to subsystems
b. global title translation
c. load sharing among SCPs
d. all of the above and other functions as well
20. Signaling units are broken up into units of how many bits?
a. 8
b. 16
c. 32
d. 64

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16. ACRONYM GUIDE


ACM

address complete message

A Link

access link

ANM

answer message

BIB

backward indicator bit

B Link

bridge link

BSN

backward sequence number

CLASS

custom local-area signaling services

D Link

diagonal link

DPC

destination point code

E Link

extended link

FIB

forward indicator bit

FISU

fill in signal unit

F Link

fully associated link

FSN

forward sequence number

IAM

initial address message

ISDN

integrated services digital network

ISUP

ISDN user part

LSSU

link status signal unit

MF

multifrequency

MSU

message signal unit

MTP

message transfer part

OAM&P

operations, administration, and maintenance part

OPC

originating point code

PSTN

public switched telephone network

RBOC

regional Bell operating company

RCL

release complete message

REL

release message

RSP

route set prohibited test message

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RSR

restricted test message

SCCP

signaling connection control part

SCP

signal control point

SLS

signaling link selection

SSP

signal switching point

SS7

signaling system 7

STP

signal transfer point

SU

signal unit

TCAP

transaction capabilities application part

TFA

transfer allowed message

TFP

transfer prohibited message

TFR

transfer restricted message

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(SONETs)
DEFINITION
Synchronous optical network (SONET) is a standard for optical telecommunications transport formulated by the Exchange Carriers Standards Association
(ECSA) for the American National Standards Institute (ANSI), which sets
industry standards in the United States for telecommunications and other
industries. The comprehensive SONET standard is expected to provide the
transport infrastructure for worldwide telecommunications for at least the
next two or three decades.

TUTORIAL OVERVIEW
This tutorial provides an introduction to the SONET standard. Standards in
the telecommunications field are always evolving. Information in this SONET
primer is based on the latest information available from the Bellcore and
ITUT standards organizations.
Use this primer as an introduction to the technology of SONET. If more
detailed information is required, consult the latest material from Bellcore and
ITUT, paying particular attention to the latest date.

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TOPICS
1. INTRODUCTION TO SONET ................................................................323
2. WHY SYNCHRONIZE? ............................................................................326
3. FRAME FORMAT STRUCTURE ..............................................................327
4. OVERHEADS..............................................................................................332
5. POINTERS ..................................................................................................346
6. SONET MULTIPLEXING ..........................................................................356
7. SONET NETWORK ELEMENTS ..............................................................358
8. SONET NETWORK CONFIGURATIONS ..............................................363
9. WHAT ARE THE BENEFITS OF SONET?................................................366
10. SDH REFERENCE ....................................................................................370
11. SELF-TEST ................................................................................................373
12. ACRONYM GUIDE ................................................................................375

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1. INTRODUCTION TO SONET
Synchronous optical network (SONET) is a standard for optical telecommunications transport. It was formulated by the Exchange Carriers Standards
Association (ECSA) for the American National Standards Institute (ANSI),
which sets industry standards in the United States for telecommunications
and other industries. The comprehensive SONET/SDH standard is expected
to provide the transport infrastructure for worldwide telecommunications for
at least the next two or three decades.
The increased configuration flexibility and bandwidth availability of SONET
provides significant advantages over the older telecommunications system.
These advantages include the following:
reduction in equipment requirements and an increase in network reliability.
provision of overhead and payload bytes; the overhead bytes permit management of the payload bytes on an individual basis and facilitate centralized fault sectionalization
definition of a synchronous multiplexing format for carrying lower-level
digital signals (such as DS1, DS3) and a synchronous structure that greatly
simplifies the interface to digital switches, digital cross-connect switches,
and add-drop multiplexers
availability of a set of generic standards that enable products from different
vendors to be connected
definition of a flexible architecture capable of accommodating future applications, with a variety of transmission rates; in brief, SONET defines optical
carrier (OC) levels and electrically equivalent synchronous transport signals
(STSs) for the fiber-optic-based transmission hierarchy
Background
Before SONET, the first generations of fiber-optic systems in the public telephone network used proprietary architectures, equipment, line codes, multiplexing formats, and maintenance procedures. The users of this equipment
regional Bell operating companies and interexchange carriers (IXCs) in the
U.S., Canada, Korea, Taiwan, and Hong Kongwanted standards so they
could mix and match equipment from different suppliers. The task of creating

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such a standard was taken up in 1984 by the Exchange Carriers Standards
Association (ECSA) to establish a standard for connecting one fiber system to
another. This standard is called synchronous optical network (SONET).
Synchronization of Digital Signals
To understand correctly the concepts and details of SONET, it is important to
be clear about the meaning of synchronous, asynchronous, and plesiochronous.
In a set of synchronous signals, the digital transitions in the signals occur at
exactly the same rate. There may, however, be a phase difference between the
transitions of the two signals, and this would lie within specified limits. These
phase differences may be a result of propagation time delays or jitter introduced into the transmission network. In a synchronous network, all the clocks
are traceable to one primary reference clock (PRC). The accuracy of the PRC is
better than 1 in 1011 and is derived from a cesium atomic standard.
If two digital signals are plesiochronous, their transitions occur at almost the
same rate, with any variation being constrained within tight limits. For example,
if two networks must interwork, their clocks may be derived from two different
PRCs. Although these clocks are extremely accurate, there is a difference
between one clock and the other. This is known as a plesiochronous difference.
In the case of asynchronous signals, the transitions of the signals do not necessarily occur at the same nominal rate. Asynchronous, in this case, means
that the difference between two clocks is much greater than a plesiochronous
difference. For example, if two clocks are derived from free-running quartz
oscillators, they could be described as asynchronous.

Basic SONET Signal


SONET defines a technology for carrying many signals of different capacities
through a synchronous, flexible, optical hierarchy. This is accomplished by
means of a byte-interleaved multiplexing scheme. Byte-interleaving simplifies
multiplexing and offers end-to-end network management.
The first step in the SONET multiplexing process involves the generation of
the lowest-level or base signal. In SONET, this base signal is referred to as
synchronous transport signal level 1, or simply STS1, which operates at

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51.84 Mbps. Higher-level signals are integer multiples of STS1, creating the
family of STSN signals in Table 1. An STS-N signal is composed of N byteinterleaved STS1 signals. This table also includes the optical counterpart for
each STSN signal, designated OCN (optical carrierlevel N).
Synchronous and nonsynchronous line rates and the relationships between
each are shown in Tables 1 and 2.
Signal

Bit Rate

Capacity

STS1, OC1

51.840 Mbps

28 DS1s or 1 DS3

STS3, OC3

155.520 Mbps

84 DS1s or 3 DS3s

STS12, OC12

622.080 Mbps

336 DS1s or 12 DS3s

STS48, OC48

2488.320 Mbps

1344 DS1s or 48 DS3s

STS192, OC192

9953.280 Mbps

5376 DS1s or 192 DS3s

STS = synchronous transport signal; OC = optical carrier


Table 1. SONET Hierarchy
Signal

Bit Rate

Channels

DS0

64 kbps

1 DS0

DS1

1.544 Mbps

24 DS0s

DS2

6.312 Mbps

96 DS0s

DS3

44.736 Mbps

28 DS1s

Table 2. Nonsynchronous Hierarchy

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2. WHY SYNCHRONIZE?
Synchronous versus Asynchronous
Traditionally, transmission systems have been asynchronous, with each terminal in the network running on its own clock. In digital transmission, clocking is one of the most important considerations. Clocking means using a
series of repetitive pulses to keep the bit rate of data constant and to indicate
where the ones and zeroes are located in a datastream.
As these clocks are totally free-running and not synchronized, large variations
occur in the clock rate and thus the signal bit rate. For example, a DS3 signal
specified at 44.736 Mbps + 20 ppm (parts per million) can produce a variation
of up to 1789 bps between one incoming DS3 and another.
Asynchronous multiplexing uses multiple stages. Signals such as asynchronous DS1s are multiplexed, extra bits are added (bit-stuffing) to account for
the variations of each individual stream, and bits are combined with other
bits (framing bits) to form a DS2 stream. Bit-stuffing is used again to multiplex up to DS3. DS3s are multiplexed up to higher rates in the same manner. At the higher asynchronous rate, they cannot be accessed without
demultiplexing.
In a synchronous system, such as SONET, the average frequency of all clocks
in the system will be the same (synchronous) or nearly the same (plesiochronous). Every clock can be traced back to a highly stable reference supply.
Thus, the STS1 rate remains at a nominal 51.84 Mbps, allowing many synchronous STS1 signals to be stacked together when multiplexed without any
bit-stuffing. Thus, the STS1s are easily accessed at a higher STSN rate.
Low-speed synchronous virtual tributary (VT) signals are also simple to interleave and transport at higher rates. At low speeds, DS1s are transported by
synchronous VT1.5 signals at a constant rate of 1.728 Mbps. Single-step multiplexing up to STS1 requires no bit-stuffing, and VTs are easily accessed.
Pointers accommodate differences in the reference source frequencies and phase
wander, and prevent frequency differences during synchronization failures.

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Synchronization Hierarchy
Digital switches and digital cross-connect systems are commonly employed in
the digital network synchronization hierarchy. The network is organized with
a master-slave relationship, with clocks of the higher-level nodes feeding timing signals to clocks of the lower-level nodes. All nodes can be traced up to a
primary reference source, a Stratum 1 atomic clock with extremely high stability and accuracy. Less-stable clocks are adequate to support the lower nodes.
Synchronizing SONET
The internal clock of a SONET terminal may derive its timing signal from a
building-integrated timing supply (BITS) used by switching systems and other
equipment. Thus, this terminal will serve as a master for other SONET nodes,
providing timing on its outgoing OCN signal. Other SONET nodes will
operate in a slave mode called loop timing with their internal clocks timed by
the incoming OCN signal. Current standards specify that a SONET network
must be able to derive its timing from a Stratum 3 or higher clock.

3. FRAME FORMAT STRUCTURE


SONET uses a basic transmission rate of STS1equivalent to 51.84 Mbps.
Higher-level signals are integer multiples of the base rate. For example, STS3
is three times the rate of STS1 (3 x 51.84 = 155.52 Mbps). An STS12 rate
would be 12 x 51.84 = 622.08 Mbps.
STS1 Building Block
The frame format of the STS1 signal is shown in Figure 1. In general, the
frame can be divided into two main areas: transport overhead and the synchronous payload envelope (SPE).

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Figure 1. STS1 Frame Format

The synchronous payload envelope can also be divided into two parts: the
STS path overhead and the payload. The payload is the revenue-producing
traffic being transported and routed over the SONET network. Once the payload is multiplexed into the synchronous payload envelope, it can be transported and switched through SONET without having to be examined and
possibly demultiplexed at intermediate nodes. Thus, SONET is said to be service-independent or transparent.
Transport overhead is composed of section overhead and line overhead. The
STS1 path overhead is part of the synchronous payload envelope.
The STS1 payload has the capacity to transport loads of up to the following:
28 DS1s
1 DS3
21 2.048 Mbps signals
combinations of each.
STS1 Frame Structure
STS1 is a specific sequence of 810 bytes (6,480 bits), which includes various
overhead bytes and an envelope capacity for transporting payloads. It can be
depicted as a 90-column by 9-row structure. With a frame length of 125 s
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(8,000 frames per second), STS1 has a bit rate of 51.840 Mbps. The order of
transmission of bytes is row-by-row from top to bottom, left to right (most
significant bit first).
As shown in Figure 1, the first three columns of the STS1 frame are for the
transport overhead. The three columns contain 9 bytes. Of these, 9 bytes are
overhead for the section layer (for example, each section overhead), and 18
bytes are overhead for the line layer (for example, line overhead). The
remaining 87 columns constitute the STS1 envelope capacity (payload and
path overhead).
As stated before, the basic signal of SONET is the synchronous transport signallevel 1, or STS1. The STS frame format is composed of 9 rows of 90
columns of 8-bit bytes, or 810 bytes. The byte transmission order is row-byrow, left to right. At a rate of 8,000 frames per second, that works out to a
rate of 51.840 Mbps, as the following equation demonstrates:
9 x 90 bytes/frame x 8 bits/byte x 8000 frames/s = 51,840,000 bps=
51.840 Mbps
This is known as the STS1 signal ratethe electrical rate used primarily for
transport within a specific piece of hardware. The optical equivalent of STS1
is known as OC1, and it is used for transmission across the fiber.
The STS1 frame consists of overhead, plus a synchronous payload envelope
(see Figure 2). The first three columns of each STS1 frame make up the transport overhead, and the last 87 columns make up the SPE. SPEs can have any
alignment within the frame, and this alignment is indicated by the H1 and H2
pointer bytes in the line overhead.

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Figure 2. STS1 Frame Elements


STS1 Envelope Capacity and Synchronous Payload Envelope (SPE)
Figure 3 depicts the STS1 SPE, which occupies the STS1 envelope capacity.
The STS1 SPE consists of 783 bytes, and can be depicted as an 87-column by
9-row structure. Column 1 contains 9 bytes, designated as the STS path overhead (POH). Two columns (columns 30 and 59) are not used for payload, but
are designated as the fixed stuff columns. The 756 bytes in the remaining
84 columns are designated as the STS1 payload capacity.

Figure 3. STS1 SPE Example

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STS1 SPE in Interior of STS1 Frames
The STS1 SPE may begin anywhere in the STS1 envelope capacity (see
Figure 4). Typically, it begins in one STS1 frame and ends in the next. The
STS payload pointer contained in the transport overhead designates the location of the byte where the STS1 SPE begins.
STS POH is associated with each payload and is used to communicate various information from the point where a payload is mapped into the STS1
SPE to where it is delivered.

Figure 4. STS1 SPE Position in the STS1 Frame

STSN Frame Structure


An STSN is a specific sequence of Nx810 bytes. The STSN is formed by
byte-interleaving STS1 modules (see Figure 5). The transport overheads of the
individual STS1 modules are frame aligned before interleaving, but the associated STS SPEs are not required to be aligned because each STS1 has a payload pointer to indicate the location of the SPE (or to indicate concatenation).

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Figure 5. STSN

4. OVERHEADS
SONET provides substantial overhead information, allowing simpler multiplexing and greatly expanded operations, administration, maintenance, and
provisioning (OAM&P) capabilities. The overhead information has several
layers, which are shown in Figure 6. Path-level overhead is carried from endto-end; it is added to DS1 signals when they are mapped into virtual tributaries and for STS1 payloads that travel end-to-end. Line overhead is for the
STSN signal between STSN multiplexers. Section overhead is used for communications between adjacent network elements, such as regenerators.
Enough information is contained in the overhead to allow the network to
operate and allow OAM&P communications between an intelligent network
controller and the individual nodes.

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Figure 6. Overhead Layers


The following sections detail the different SONET overhead information:
section overhead
line overhead
STS path overhead
VT path overhead
This information has been updated to reflect changes in Bellcore GR253,
Issue 2, December 1995.

Section Overhead
Section overhead contains 9 bytes of the transport overhead accessed, generated, and processed by section-terminating equipment. This overhead supports functions such as the following:
performance monitoring (STSN signal)
local orderwire
data communication channels to carry information for OAM&P
framing
This might be two regenerators, line-terminating equipment and a regenerator, or two line-terminating equipments. The section overhead is found in the
first three rows of Columns 1 to 9 (see Figure 7).
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Figure 7. Section OverheadRows 1 to 3 of Transport Overhead


Table 3 shows section overhead byte-by-byte.

334

Byte

Description

A1 and A2

Framing bytesThese two bytes indicate the


beginning of an STS-1 frame.

J0

Section trace (J0)/section growth (Z0)The byte in


each of the N STS1s in an STSN that was formally defined as the STS1 ID (C1) byte has been
refined either as the section trace byte (in the first
STS1 of the STSN), or as a section growth byte
(in the second through Nth STS1s).

B1

Section bit interleaved parity code (BIP8)


byteThis is a parity code (even parity),
used to check for transmission errors over a
regenerator section. Its value is calculated
over all bits of the previous STSN frame
after scrambling, then placed in the B1 byte
of STS1 before scrambling. Therefore, this
byte is defined only for STS1 number 1 of
an STSN signal.

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E-1

Section orderwire byteThis byte is allocated


to be used as a local orderwire channel for
voice communication between regenerators,
hubs, and remote terminal locations.

F1

Section user channel byteThis byte is set aside


for the users purposes. It terminates at all section-terminating equipment within a line; that
is, it can be read and/or written to at all section terminating equipment in that line.

D1, D2, D3

Section data communications channel (DCC)


bytesTogether, these three bytes form a
192kbps message channel providing a message-based channel for OAM&P between
pieces of section-terminating equipment. The
channel is used from a central location for
alarms, control, monitoring, administration,
and other communication needs. It is available for internally generated, externally generated, or manufacturer-specific messages.

Table 3. Section Overhead


Line Overhead
Line overhead contains 18 bytes of overhead accessed, generated, and processed
by line-terminating equipment. This overhead supports functions such as the following:
locating the SPE in the frame
multiplexing or concatenating signals
performance monitoring
automatic protection switching
line maintenance

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Line overhead is found in Rows 4 to 9 of Columns 1 to 9 (see Figure 8).

Figure 8. Line OverheadRows 4 to 9 of Transport Overhead


Table 4 shows Line overhead byte-by-byte.

336

Byte

Description

H1, H2

STS Payload Pointer (H1 and H2)Two bytes are allocated to a pointer that indicates the offset in bytes
between the pointer and the first byte of the STS SPE.
The pointer bytes are used in all STS1s within an
STSN to align the STS1 transport overhead in the
STSN, and to perform frequency justification. These
bytes are also used to indicate concatenation and to
detect STS path alarm indiciation signals (AISP).

H3

Pointer action byte (H3)The pointer action byte is


allocated for SPE frequency justification purposes. The
H3 byte is used in all STS1s within an STSN to
carry the extra SPE byte in the event of a negative
pointer adjustment. The value contained in this byte
when it is not used to carry the SPE byte is undefined.

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B2

Line bit-interleaved parity code (BIP-8) byteThis parity


code byte is used to determine if a transmission error
has occurred over a line. It is even parity and is calculated over all bits of the line overhead and STS1
SPE of the previous STS1 frame before scrambling.
The value is placed in the B2 byte of the line overhead before scrambling. This byte is provided in all
STS1 signals in an STSN signal.

K1 and K2

Automatic protection switching (APS channel) bytes


These two bytes are used for protection signaling
between line-terminating entities for bidirectional
automatic protection switching and for detecting
alarm indication signals (AISL) and remote defect
indication (RDI) signals.

D4 to D12

Line data communications channel (DCC) bytesThese 9


bytes form a 576kbps message channel from a central
location for OAM&P information (alarms, control, maintenance, remote provisioning, monitoring, administration, and other communication needs) between line entities. Available for internally generated, externally generated, and manufacturer-specific messages. A protocol analyzer is required to access the lineDCC information.

S1

Synchronization status (S1)The S1 byte is located in


the first STS1 of an STSN, and bits 5 through 8 of
that byte are allocated to convey the synchronization
status of the network element.

Z1

Growth (Z1)The Z1 byte is located in the second


through Nth STS1s of an STSN (3</=N</=48) and
are allocated for future growth. Note that an OC1
or STS1 electrical signal does not contain a Z1 byte.

M0

STS1 REIL (M0)The M0 byte is only defined for


STS1 in an OC1 or STS1 electrical signal. Bits 5
through 8 are allocated for a line remote error indication function (REIL, formerly referred to as line
FEBE), which conveys the error count detected by an
LTE (using the line BIP8 code) back to its peer LTE.
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M1

STSN REIL (M1)The M1 byte is located in the


third STS1 (in order of appearance in the byte-interleaved STSN electrical or OCN signal) in an
STSN (N>/=3), and is used for an REIL function.

Z2

Growth (Z2)The Z2 byte is located in the first and


second STS1s of an STS3, and the first, second,
and fourth through Nth STS1s of an STSN
(12</=N</=48). These bytes are allocated for future
growth. Note that an OC1 or STS1 electrical signal
does not contain a Z2 byte.

E2

Orderwire byteThis orderwire byte provides a 64kbps


channel between line entities for an express orderwire.
It is a voice channel for use by technicians and will be
ignored as it passes through the regenerators.

Table 4. Line Overhead


STS Path Overhead
STS path overhead (STS POH) contains 9 evenly distributed path overhead bytes
per 125 microseconds starting at the first byte of the STS SPE. STS POH provides for communication between the point of creation of an STS SPE and its
point of disassembly. This overhead supports functions such as the following:
performance monitoring of the STS SPE
signal label (the content of the STS SPE, including status of mapped payloads)
path status
path trace
The path overhead is found in Rows 1 to 9 of the first column of the STS1
SPE (see Figure 9).

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Figure 9. Path OverheadRows 1 to 9


Table 5 describes path overhead byte by byte.
Byte

Description

J1

STS path trace byteThis user-programmable byte


repetitively transmits a 64-byte, or 16-byte E.164 format string. This allows the receiving terminal in a
path to verify its continued connection to the intended transmitting terminal.

B3

STS path bit interleaved parity code (Path BIP8)


byteThis is a parity code (even) used to determine if
a transmission error has occurred over a path. Its value
is calculated over all the bits of the previous synchronous payload envelope (SPE) before scrambling.

C2

STS path signal label byteThis byte is used to indicate the content of the STS SPE, including the status
of the mapped payloads.

Table 5. STS Path Overhead

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G1

Path status byteThis byte is used to convey the


path-terminating status and performance back to the
originating pathterminating equipment. Therefore,
the duplex path in its entirety can be monitored
from either end or from any point along the path.
Bits 1 through 4 are allocated for an STS Path REI
function (REIP, formerly referred to as STS path
FEBE). Bits 5, 6, and 7 of the G1 byte are allocated
for an STS path RDI (RDIP) signal. Bit 8 of the G1
byte is currently undefined.

F2

Path user channel byteThis byte is used for user


communication between path elements.

H4

Virtual Tributary (VT) multiframe indicator byteThis


byte provides a generalized multiframe indicator for
payload containers. At present, it is used only for
tributary unit structured payloads.

Note: The path overhead portion of the SPE remains with the payload until it
is demultiplexed.

VT Path Overhead
VT path overhead (VT POH) contains 4 evenly distributed path overhead
bytes per VT SPE starting at the first byte of the VT SPE. VT POH provides
for communication between the point of creation of an VT SPE and its point
of disassembly.
Four bytes (V5, J2, Z6, and Z7) are allocated for VT POH. The first byte of a
VT SPE (i.e., the byte in the location pointed to by the VT payload pointer) is
the V5 byte, while the J2, Z6, and Z7 bytes occupy the corresponding locations in the subsequent 125 microsecond frames of the VT superframe.
The V5 byte provides the same functions for VT paths that the B3, C2, and
G1 bytes provide for STS paths: namely, error checking, signal label, and path
status. The bit assignments for the V5 byte are illustrated in Figure 10.

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Figure 10. VT Path OverheadV5 Byte


Bits 1 and 2 of the V5 byte are allocated for error performance monitoring.
Bit 3 of the V5 byte is allocated for a VT path REI function (REIV, formerly
referred to as VT path FEBE) to convey the VT path-terminating performance
back to an originating VT PTE.
Bit 4 of the V5 byte is allocated for a VT path remote failure indication
(RFIV) in the byte-synchronous DS1 mapping.
Bits 5 through 7 of the V5 byte are allocated for a VT path signal label to indicate the content of the VT SPE.
Bit 8 of the VT byte is allocated for a VT path remote defect indication
(RDIV) signal.

SONET Alarm Structure


The SONET frame structure has been designed to contain a large amount of
overhead information. The overhead information provides a variety of management and other functions such as the following:
error performance monitoring
pointer adjustment information
path status
path trace
section trace
remote defect, error, and failure indications

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signal labels
new data flag indications
data communications channels (DCC)
automatic protection switching (APS) control
orderwire
synchronization status message
Much of this overhead information is involved with alarm and in-service
monitoring of the particular SONET sections.
SONET alarms are defined as follows:
AnomalyThe smallest discrepancy that can be observed between the actual
and desired characteristics of an item. The occurrence of a single anomaly does
not constitute an interruption in the ability to perform a required function.
DefectThe density of anomalies has reached a level where the ability to perform a required function has been interrupted. Defects are used as input for
performance monitoring, the control of consequent actions, and the determination of fault cause.
FailureThe inability of a function to perform a required action persisted
beyond the maximum time allocated.
Table 6 describes SONET alarm anomalies, defects, and failures.

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Abbrev.

Description

Criteria

LOS

loss of signal

LOS is raised when the synchronous signal (STSN) level drops below the
threshold at which a BER of 1 in 103 is
predicted. It could be due to a cut cable,
excessive attenuation of the signal or
equipment fault.
The LOS state clears when two consecutive framing patterns are received and
no new LOS condition is detected.

OOF

out of frame alignment

OOF state occurs when four or five consecutive SONET frames are received
with invalid (errored) framing patterns
(A1 and A2 bytes). The maximum time
to detect OOF is 625 microseconds.
OOF state clears when two consecutive
SONET frames are received with valid
framing patterns.

LOF

loss of frame alignment

LOF state occurs when the OOF state


exists for a specified time in milliseconds. The LOF state clears when an inframe condition exists continuously for
a specified time in milliseconds.

LOP

loss of pointer

LOP state occurs when N consecutive


invalid pointers are received or N consecutive new data flags (NDF) are
received (other than in a concatenation
indicator), where N=8, 9, or 10.
LOP state is cleared when three equal
valid pointers or three consecutive AIS
indications are received.
LOP can also be identified as the following:
SPLOP (STS path loss of pointer)
VPLOP (VT path loss of pointer)

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Abbrev.

Description

Criteria

AIS

alarm indication signal

The AIS is an allONES characteristic or


adapted information signal. It is generated
to replace the normal traffic signal when
it contains a defect condition to prevent
consequential downstream failures being
declared or alarms being raised.
AIS can also be identified as the following:
AISL (line alarm indication signal)
SPAIS (STS path alarm indiciation signal)
VPAIS (VT path alarm indiciation signal)

REI

remote error indication

An indication returned to a transmitting


node (source) that an errored block has
been detected at the receiving node
(sink). This indication was formerly
known as far end block error (FEBE).
REI can also be identified as the following:
REIL (line remote error indication)
REIP (STS path remote error indication)
REIV (VT path remote error indication)

RDI

remote defect indication

A signal returned to the transmitting terminating equipment upon detecting a loss of


signal, loss of frame, or AIS defect. RDI
was previously known as FERF.
RDI can also be identified as the following:
RDIL (line remote defect indication)
RDIP (STS path remote defect indication)
RDIV (VT path remote defect indication)

RFI

remote failure indication

A failure is a defect that persists beyond


the maximum time allocated to the transmission system protection mechanisms.
When this situation occurs, an RFI is sent
to the far end and will initiate a protection
switch if this function has been enabled.
RFI can also be identified as the following:
RFIL (line remote failure indication)
RFIP (STS path remote failure indication)
RFIV (VT path remote failure indication)

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Abbrev.

Description

Criteria

B1 error

B1 error

Parity errors evaluated by byte B1 (BIP8)


of an STSN are monitored. If any of the
eight parity checks fail, the corresponding
block is assumed to be in error.

B2 error

B2 error

Parity errors evaluated by byte B2 (BIP24


x N) of an STSN are monitored. If any of
the N x 24 parity checks fail, the corresponding block is assumed to be in error.

B3 error

B3 error

Parity errors evaluated by byte B3 are


monitored. If any of the eight parity
checks fail, the corresponding block is
assumed to be in error.

BIP-2 error

BIP2 error

Parity errors contained in bits 1 and 2


(BIP2: bit interleaved parity2) of byte V5
of a VTM (M=11,12,2) are monitored. If
any of the two parity checks fail, the corresponding block is assumed to be in error.

LSS

loss of sequence
synchronization

Bit error measurements using pseudorandom sequences can only be performed if


the reference sequence produced on the synchronization receiving side of the test set-up
is correctly synchronized to the sequence
coming from the object under test. To
achieve compatible measurement results, it
is necessary that the sequence synchronization characteristics are specified.
Sequence synchronization is considered
to be lost and resynchronization shall be
started if the events occur as follows:
Bit error ratio is >/= 0.20 during an integration interval of 1 second; or it can be
unambiguously identified that the test
sequence and the reference sequence are
out of phase

Note: One method for recognizing the out-of-phase condition is the evaluation of the
error pattern resulting from the bit-by-bit comparison. If the error pattern has the same
structure as the pseudo-random test sequence, the out-of-phase condition is reached.

Table 6. Anomalies, Defects, and Failures


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5. POINTERS
SONET uses a concept called pointers to compensate for frequency and phase
variations. Pointers allow the transparent transport of synchronous payload
envelopes (either STS or VT) across plesiochronous boundaries (i.e., between
nodes with separate network clocks having almost the same timing). The use
of pointers avoids the delays and loss of data associated with the use of large
(125-microsecond frame) slip buffers for synchronization.
Pointers provide a simple means of dynamically and flexibly phase-aligning
STS and VT payloads, thereby permitting ease of dropping, inserting, and
cross-connecting these payloads in the network. Transmission signal wander
and jitter can also be readily minimized with pointers.
Figure 11 shows an STS1 pointer (H1 and H2 bytes), which allows the SPE to
be separated from the transport overhead. The pointer is simply an offset value
that points to the byte where the SPE begins. The diagram depicts the typical
case of the SPE overlapping onto two STS1 frames. If there are any frequency
or phase variations between the STS1 frame and its SPE, the pointer value will
be increased or decreased accordingly to maintain synchronization.

Figure 11. PointerSPE Position in the STS1 Frame

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VT Mappings
There are several options for how payloads are actually mapped into the VT.
Locked-mode VTs bypass the pointers with a fixed byte-oriented mapping of
limited flexibility. Floating mode mappings use the pointers to allow the payload to float within the VT payload. There are three different floating mode
mappingsasynchronous, bit-synchronous, and byte-synchronous.

Concatenated Payloads
For future services, the STS1 may not have enough capacity to carry some
services. SONET offers the flexibility of concatenating STS1s to provide the
necessary bandwidth. Consult the Acronym Guide/Glossary for an explanation of concatenation. STS1s can be concatenated up to STS3c. Beyond
STS3, concatenation is done in multiples of STS3c. Virtual tributaries can be
concatenated up to VT6 in increments of VT1.5, VT2, or VT6.

Payload Pointers
When there is a difference in phase or frequency, the pointer value is adjusted. To accomplish this, a process known as byte-stuffing is used. In other
words, the SPE payload pointer indicates where in the container capacity a
VT starts, and the byte-stuffing process allows dynamic alignment of the SPE
in case it slips in time.

Positive Stuffing
When the frame rate of the SPE is too slow in relation to the rate of the
STS1, bits 7, 9, 11, 13, and 15 of the pointer word are inverted in one frame,
thus allowing 5-bit majority voting at the receiver. (These bits are known as
the I-bits or increment bits.) Periodically, when the SPE is about one byte off,
these bits are inverted, indicating that positive stuffing must occur. An additional byte is stuffed in, allowing the alignment of the container to slip back
in time. This is known as positive stuffing, and the stuff byte is made up of
noninformation bits. The actual positive stuff byte immediately follows the
H3 byte (that is, the stuff byte is within the SPE portion). The pointer is incre-

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mented by one in the next frame, and the subsequent pointers contain the
new value. Simply put, if the SPE frame is traveling more slowly than the
STS1 frame, every now and then stuffing an extra byte in the flow gives
the SPE a one-byte delay (see Figure 12).

Figure 12. Payload PointerPositive Justification


Negative Stuffing
Conversely, when the frame rate of the SPE frame is too fast in relation to the
rate of the STS1 frame, bits 8, 10, 12, 14, and 16 of the pointer word are
inverted, thus allowing 5-bit majority voting at the receiver. (These bits are
known as the D-bits, or decrement bits.) Periodically, when the SPE frame is
about one byte off, these bits are inverted, indicating that negative stuffing
must occur. Because the alignment of the container advances in time, the
envelope capacity must be moved forward. Thus, actual data is written in the
H3 byte, the negative stuff opportunity (within the overhead); this is known
as negative stuffing.

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The pointer is decremented by one in the next frame, and the subsequent
pointers contain the new value. Simply put, if the SPE frame is traveling more
quickly than the STS1 frame, every now and then pulling an extra byte from
the flow and stuffing it into the overhead capacity (the H3 byte) gives the SPE
a one-byte advance. In either case, there must be at least three frames in
which the pointer remains constant before another stuffing operation (and
therefore a pointer value change) can occur (see Figure 13).

Figure 13. Payload PointerNegative Justification


Virtual Tributaries
In addition to the STS1 base format, SONET also defines synchronous formats at subSTS1 levels. The STS1 payload may be subdivided into virtual
tributaries, which are synchronous signals used to transport lower-speed
transmissions. The sizes of VTs are displayed in Table 7.

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VT Type

Bit Rate

Size of VT

VT 1.5

1.728 Mbps

9 rows, 3 columns

VT 2

2.304 Mbps

9 rows, 4 columns

VT 3

3.456 Mbps

9 rows, 6 columns

VT 6

6.912 Mbps

9 rows, 12 columns

Table 7. Virtual Tributaries (VT)


To accommodate mixes of different VT types within an STS1 SPE, the VTs
are grouped together. An STS1 SPE that is carrying virtual tributaries is
divided into 7 VT groups, with each VT group using 12 columns of the STS1
SPE; note that the number of columns in each of the different VT types, 3, 4,
6, and 12, are all factors of 12. Each VT group can contain only one size (type)
of virtual tributary, but within an STS1 SPE, there can be a mix of the different VT groups.
For example, an STS1 SPE may contain four VT1.5 groups and three VT6
groups, for a total of seven VT groups. Thus, an SPE can carry a mix of any of
the seven groups. The groups have no overhead or pointers; they are just a
way of organizing the different VTs within an STS1 SPE.
As each of the VT groups is allocated 12 columns of the synchronous payload
envelope, a VT group would contain one of the following combinations:
four VT1.5s (with 3 columns per VT1.5)
three VT2s (with 4 columns per VT2)
two VT3s (with 6 columns per VT3)
one VT6 (with 12 columns per VT6)
The 12 columns in a VT group are not consecutive within the SPE; they are
interleaved column by column with respect to the other VT groups. As well,
column 1 is used for the path overhead; the two columns of fixed stuff are
assigned to columns 30 and 59.
The first VT group, called group 1, is found in every seventh column, starting
with column 2, and skipping columns 30 and 59. That is, the 12 columns for

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VT group 1 are columns 2, 9, 16, 23, 31, 38, 45, 52, 60, 67, 74, and 81.
Just as the VT group columns are not placed in consecutive columns in an
STS1 SPE, the virtual tributary columns within a group are not placed in
consecutive columns within that group. The columns of the individual VTs
within the VT group are interleaved as well (see Figure 14).

Figure 14. SONET TributariesVT Structured STS1 SPE

The VT structure is designed for transport and switching of subSTS1 rate


payloads. There are four sizes of VTs: VT1.5 (1.728 Mbps), VT2 (2.304 Mbps),
VT3 (3.456 Mbps), and VT6 (6.912 Mbps). In the 87 column by 9 row structure of the STS1 SPE, these VTs occupy columns 3, 4, 6, and 12, respectively.
To accommodate a mix of VT sizes efficiently, the VT structured STS1 SPE
is divided into seven VT groups. Each VT group occupies 12 columns of the
87-column STS1 SPE, and may contain 4 VT1.5s, 3 VT2s, 2 VT3s, or 1 VT6.
A VT group can contain only one size of VTs; however, a different VT size is
allowed for each VT group in an STS1 SPE (see Figure 15).

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Figure 15. VT Structure, VT Sizes


STS-1 VT1.5 SPE Columns
One of the benefits of SONET is that it can carry large payloads (above 50
Mbps). However, the existing digital hierarchy can be accommodated as well,
thus protecting investments in current equipment. To achieve this capacity,
the STS synchronous payload envelope (SPE) can be subdivided into smaller
components or structures, known as virtual tributaries (VT), for the purpose
or transporting and switching payloads smaller than the STS1 rate. All services below the DS3 rate are transported in the VT structure. Figure 16 shows
the VT1.5 structured STS1 SPE. Table 8 matches up the VT1.5 locations and
the STS1 SPE column numbers, per the Bellcore GR253CORE standard.

Figure 16. STS1 VT1.5 SPE Columns


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VT Group #, VT #

Column #s

1,1
2,1
3,1
4,1
5,1
6,1
7,1
1,2
2,2
3,2
4,2
5,2
6,2
7,2
1,3
2,3
3,3
4,3
5,3
6,3
7,1
1,4
2,4
3,4
4,4
5,4
6,4
7,4

2,31,60
3,32,61
4,33,62
5,34,63
6,35,64
7,36,65
8,37,66
9,38,67
10,39,68
11,40,69
12,41,70
13,42,71
14,43,72
15,44,73
16,45,74
17,46,75
18,47,76
19,48,77
20,49,78
21,50,79
22,51,80
23,52,81
24,53,82
25,54,83
26,55,84
27,56,85
28,57,86
29,58,87

Column 1 = STS1 POH


30 = fixed stuff
59 = fixed stuff
Table 8. VT1.5 Locations Matched to the STS1 SPE Column Numbers

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DS1 Visibility
Because the multiplexing is synchronous, the low-speed tributaries (input signals) can be multiplexed together but are still visible at higher rates. An individual VT containing a DS1 can be extracted without demultiplexing the
entire STS1. This improved accessibility improves switching and grooming
at VT or STS levels.
In an asynchronous DS3 frame, the DS1s have gone through two levels of
multiplexing (DS1 to DS2; DS2 to DS3) which include the addition of stuffing
and framing bits. The DS1 signals are mixed somewhere in the informationbit fields and cannot be easily identified without completely demultiplexing
the entire frame.
Different synchronizing techniques are used for multiplexing. In existing
asynchronous systems, the timing for each fiber-optic transmission system
terminal is not locked onto a common clock. Therefore, large frequency variations can occur. Bit-stuffing is a technique used to synchronize the various
low-speed signals to a common rate before multiplexing.

VT Superframe and Envelope Capacity


In addition to the division of VTs into VT groups, a 500-microsecond structure
called a VT superframe is defined for each VT. The VT superframe contains
the V1 and V2 bytes (the VT payload pointer), and the VT envelope capacity,
which in turn contains the VT SPE. The VT envelope capacity, and therefore
the size of the VT SPE, is different for each VT size. V1 is the first byte in the
VT superframe, while V2 through V4 appear as the first bytes in the following
frames of the VT superframe, regardless of the VT size (see Figure 17).

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Figure 17. VT Superframe and Envelope Capacity

VT SPE and Payload Capacity


Four consecutive 125-microsecond frames of the VT structured STS1 SPE are
organized into a 500-microsecond superframe, the phase of which is indicated by the H4 (indicator) byte in the STS POH.
The VT payload pointer provides flexible and dynamic alignment of the VT
SPE within the VT envelope capacity, independent of other VT SPEs. Figure 18
illustrates the VT SPEs corresponding to the four VT sizes. Each VT SPE contains four bytes of VT POH (V5, J2, Z6, and Z7), and the remaining bytes
constitute the VT payload capacity, which is different for each VT.

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Figure 18. VT SPE and Payload Capacity

6. SONET MULTIPLEXING
The multiplexing principles of SONET are as follows:
MappingThis process is used when tributaries are adapted into virtual tributaries (VTs) by adding justification bits and path overhead (POH) information.
AligningThis process takes place when a pointer is included in the STS
Path or VT path overhead to allow the first byte of the virtual tributary to
be located.
MultiplexingThis process is used when multiple, lower order path-layer
signals are adapted into a higher-order path signal or when the higher-order
path signals are adapted into the line overhead.
StuffingSONET has the ability to handle various input tributary rates from
asynchronous signals. As the tributary signals are multiplexed and aligned,
some spare capacity has been designed into the SONET frame to provide
enough space for all these various tributary rates. Therefore, at certain points
in the multiplexing hierarchy, this space capacity is filled with fixed stuffing
bits that carry no information but are required to fill up the particular frame.
One of the benefits of SONET is that it can carry large payloads (above 50
Mbps). However, the existing digital hierarchy signals can be accommodated
as well, thus protecting investments in current equipment.
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To achieve this capability, the STS synchronous payload envelope can be subdivided into smaller components or structures, known as virtual tributaries
(VTs), for the purpose of transporting and switching payloads smaller than the
STS1 rate. All services below DS3 rate are transported in the VT structure.
Figure 19 illustrates the basic multiplexing structure of SONET. Any type of
service, ranging from voice to high-speed data and video, can be accepted by
various types of service adapters. A service adapter maps the signal into the
payload envelope of the STS1 or virtual tributary (VT). New services and
signals can be transported by adding new service adapters at the edge of the
SONET network.

Figure 19. SONET Multiplexing Hierarchy


Except for concatenated signals, all inputs are eventually converted to a base
format of a synchronous STS1 signal (51.84 Mbps or higher). Lower-speed
inputs such as DS1s are first bit- or byte-multiplexed into virtual tributaries.
Several synchronous STS1s are then multiplexed together in either a singleor two-stage process to form an electrical STSN signal (N = 1 or more).
STS multiplexing is performed at the byte interleave synchronous multiplexer.
Basically, the bytes are interleaved together in a format such that the lowspeed signals are visible. No additional signal processing occurs except a
direct conversion from electrical to optical to form an OCN signal.

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7. SONET NETWORK ELEMENTS


Terminal Multiplexer
The path terminating element (PTE), an entry-level path-terminating terminal
multiplexer, acts as a concentrator of DS1s as well as other tributary signals.
Its simplest deployment would involve two terminal multiplexers linked by
fiber with or without a regenerator in the link. This implementation represents the simplest SONET link (a section, line, and path all in one link; see
Figure 20).

Figure 20. Terminal Multiplexer


Regenerator
A regenerator is needed when, due to the long distance between multiplexers,
the signal level in the fiber becomes too low.
The regenerator clocks itself off of the received signal and replaces the section
overhead bytes before retransmitting the signal. The line overhead, payload,
and path overhead are not altered (see Figure 21).

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Figure 21. Regenerator


Add/Drop Multiplexer (ADM)
Although network elements (NEs) are compatible at the OCN level, they
may differ in features from vendor to vendor. SONET does not restrict manufacturers to providing a single type of product, nor require them to provide all
types. For example, one vendor might offer an add/drop multiplexer with
access at DS1 only, whereas another might offer simultaneous access at DS1
and DS3 rates (see Figure 22).

Figure 22. Add/Drop Multiplexer


A single-stage multiplexer/demultiplexer can multiplex various inputs into an
OCN signal. At an add/drop site, only those signals that need to be accessed
are dropped or inserted. The remaining traffic continues through the network
element without requiring special pass-through units or other signal processing.

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In rural applications, an ADM can be deployed at a terminal site or any intermediate location for consolidating traffic from widely separated locations.
Several ADMs can also be configured as a survivable ring.
SONET enables drop and repeat (also known as drop and continue)a key
capability in both telephony and cable-TV (CATV) applications. With drop
and repeat, a signal terminates at one node, is duplicated (repeated), and is
then sent to the next and subsequent nodes.
In ring-survivability applications, drop and repeat provides alternate routing
for traffic passing through interconnecting rings in a matched-nodes configuration. If the connection cannot be made through one of the nodes, the signal
is repeated and passed along an alternate route to the destination node.
In multinode distribution applications, one transport channel can efficiently carry
traffic between multiple distribution nodes. When transporting video, for example, each programming channel is delivered (dropped) at the node and repeated
for delivery to the next and subsequent nodes. Not all bandwidth (program
channels) need be terminated at all the nodes. Channels not terminating at a
node can be passed through without physical intervention to other nodes.
The add/drop multiplexer provides interfaces between the different network
signals and SONET signals.
Single-stage multiplexing can multiplex/demultiplex one or more tributary
(DS1) signals into and from an STSN signal. It can be used in terminal sites,
intermediate (add/drop) sites, or hub configurations. At an add/drop site, it
can drop lower-rate signals to be transported on different facilities, or it can
add lower-rate signals into the higher-rate STSN signal. The rest of the traffic simply continues straight through.
Wideband Digital Cross-Connects
A SONET cross-connect accepts various optical carrier rates, accesses the
STS1 signals, and switches at this level. It is ideally used at a SONET hub.
One major difference between a cross-connect and an add/drop multiplexer is
that a cross-connect may be used to interconnect a much larger number of
STS1s. The broadband cross-connect can be used for grooming (consolidating or segregating) of STS1s or for broadband traffic management. For example, it may be used to segregate high-bandwidth from low-bandwidth traffic

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and send them separately to the high-bandwidth (e.g., video) switch and a
low-bandwidth (e.g., voice) switch. It is the synchronous equivalent of a
DS3 digital cross-connect and supports hubbed network architectures.
This type is similar to the broadband cross-connect except that the switching
is done at VT levels (similar to DS1/DS2 levels). It is similar to a DS3/1
cross-connect because it accepts DS1s and DS3s and is equipped with optical interfaces to accept optical-carrier signals. It is suitable for DS1level
grooming applications at hub locations. One major advantage of wideband
digital cross-connects is that less demultiplexing and multiplexing is required
because only the required tributaries are accessed and switched.
The wideband digital cross-connect (WDCS) is a digital cross-connect that
terminates SONET and DS3 signals and has the basic functionality of VT and
DS1level cross-connections. It is the SONET equivalent to the DS3/DS1
digital cross-connect, and accepts optical OCN signals as well as STS1s,
DS1s, and DS3s.
In a wideband digital cross-connect, the switching is done at the VT level
(i.e., it cross-connects the constituent VTs between STSN terminations).
Because SONET is synchronous, the low-speed tributaries are visible and accessible within the STS1 signal. Therefore, the required tributaries can be accessed
and switched without demultiplexing, which is not possible with existing digital
cross-connects. As well, the WDCS cross-connects the constituent DS1s
between DS3 terminations and between DS3 and DS1 terminations.
The features of the WDCS make it useful in several applications. Because it
can automatically cross-connect VTs and DS1s, the WDCS can be used as a
network management system. This capability in turn makes the WDCS ideal
for grooming at a hub location (see Figure 23).

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Figure 23. Wideband Digital Cross-Connect


Broadband Digital Cross-Connects
The broadband digital cross-connect interfaces various SONET signals and
DS3s. It accesses the STS1 signals and switches at this level. It is the synchronous equivalent of the DS3 digital cross-connect, except that the broadband digital cross-connect accepts optical signals and allows overhead to be
maintained for integrated OAM&P (asynchronous systems prevent overhead
from being passed from optical signal to signal).
The broadband digital cross-connect can make two-way cross-connections at
the DS3, STS1, and STSNc levels. It is best used as a SONET hub, where it
can be used for grooming STS1s, for broadband restoration purposes, or for
routing traffic (see Figure 24).

Figure 24. Broadband Digital Cross-Connect

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Digital Loop Carrier
The digital loop carrier (DLC) may be considered a concentrator of low-speed
services before they are brought into the local central office (CO) for distribution. If this concentration were not done, the number of subscribers (or lines)
that a CO could serve would be limited by the number of lines served by the
CO. The DLC itself is actually a system of multiplexers and switches
designed to perform concentration from the remote terminals to the community dial office and, from there, to the CO.
Whereas a SONET multiplexer may be deployed at the customer premises, a
DLC is intended for service in the CO or a controlled environment vault
(CEV) that belongs to the carrier. Bellcore document TRTSY-000303
describes a generic integrated digital loop carrier (IDLC), which consists of
intelligent remote digital terminals (RDTs) and digital switch elements called
integrated digital terminals (IDTs), which are connected by a digital line. The
IDLCs are designed to more efficiently integrate DLC systems with existing
digital switches (see Figure 25).

Figure 25. Integrated Digital Loop Carrier

8. SONET NETWORK CONFIGURATIONS


Point-to-Point
The SONET multiplexer, an entry-level path-terminating terminal multiplexer,
acts as a concentrator of DS1s as well as other tributaries. Its simplest deployment involves two terminal multiplexers linked by fiber with or without a
regenerator in the link. This implementation represents the simplest SONET
configuration.
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In this configuration (see Figure 26), the SONET path and the service path
(DS1 or DS3 links end-to-end) are identical, and this synchronous island can
exist within an asynchronous network world. In the future, point-to-point
service path connections will span across the whole network and will always
originate and terminate in a multiplexer.

Figure 26. Point-to-Point

Point-to-Multipoint
A point-to-multipoint (linear add/drop) architecture includes adding and dropping circuits along the way. The SONET ADM (add/drop multiplexer) is a
unique network element specifically designed for this task. It avoids the current cumbersome network architecture of demultiplexing, cross-connecting,
adding and dropping channels, and then remultiplexing. The ADM is typically
placed along a SONET link to facilitate adding and dropping tributary channels at intermediate points in the network (see Figure 27).

Figure 27. Point-to-Multipoint

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Hub Network
The hub network architecture accommodates unexpected growth and change
more easily than simple point-to-point networks. A hub (Figure 28) concentrates traffic at a central site and allows easy reprovisioning of the circuits.
There are two possible implementations of this type of network:
1. use two or more ADMs and a wideband cross-connect switch that allows
cross-connecting the tributary services at the tributary level
2. use a broadband digital cross-connect switch that allows cross-connecting
at both the SONET level and the tributary level

Figure 28. Hub Network

Ring Architecture
The SONET building block for a ring architecture is the ADM. Multiple ADMs
can be put into a ring configuration for either bidirectional or unidirectional
traffic (see Figure 29). The main advantage of the ring topology is its survivability; if a fiber cable is cut, the multiplexers have the intelligence to send the services affected via an alternate path through the ring without interruption.
The demand for survivable services, diverse routing of fiber facilities, flexibility to rearrange services to alternate serving nodes, as well as automatic
restoration within seconds, have made rings a popular SONET topology.

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Figure 29. Ring Architecture

9. WHAT ARE THE BENEFITS OF SONET?


The transport network using SONET provides much more powerful networking capabilities than existing asynchronous systems. The key benefits provided by SONET will now be discussed.
Pointers, MUX/ DEMUX
As a result of SONET transmission, the networks clocks are referenced to a
highly stable reference point. Therefore, the need to align the data streams or
synchronize clocks is unnecessary. Therefore, a lower rate signal such as DS1
is accessible, and demultiplexing is not needed to access the bit streams. Also,
the signals can be stacked together without bit-stuffing.
For those situations in which reference frequencies may vary, SONET uses
pointers to allow the streams to float within the payload envelope.
Synchronous clocking is the key to pointers. It allows a very flexible allocation and alignment of the payload within the transmission envelope.
Reduced Back-to-Back Multiplexing
Separate M13 multiplexers (DS1 to DS3) and fiber-optic transmission system
terminals are used to multiplex a DS1 signal to a DS2, DS2 to DS3, and then
DS3 to an optical line rate. The next stage is a mechanically integrated
fiber/multiplex terminal.

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In the existing asynchronous format, care must be taken when routing circuits
to avoid multiplexing and demultiplexing too many times, as electronics (and
their associated capital cost) are required every time a DS1 signal is processed.
With SONET, DS1s can be multiplexed directly to the OCN rate. Because of
synchronization, an entire optical signal does not have to be demultiplexed
only the VT or STS signals that need to be accessed.
Optical Interconnect
Because of different optical formats among vendors asynchronous products, it
is not possible to optically connect one vendors fiber terminal to anothers. For
example, one manufacturer may use 417Mbps line rate, another 565Mbps.
A major SONET value is that it allows midspan meet with multivendor compatibility. Todays SONET standards contain definitions for fiber-to-fiber
interfaces at the physical level. They determine the optical line rate, wavelength, power levels, pulse shapes, and coding. Current standards also fully
define the frame structure, overhead, and payload mappings. Enhancements
are being developed to define the messages in the overhead channels to provide increased OAM&P functionality.
SONET allows optical interconnection between network providers regardless
of who makes the equipment. The network provider can purchase one vendors equipment and conveniently interface with other vendors SONET
equipment at either the different carrier locations or customer premises sites.
Users may now obtain the OCN equipment of their choice and meet with
their network provider of choice at that OCN level.
Multipoint Configurations
The difference between point-to-point and multipoint systems was shown previously in Figures 26 and 27. Most existing asynchronous systems are only suitable
for point-to-point, whereas SONET supports a multipoint or hub configuration.
A hub is an intermediate site from which traffic is distributed to three or
more spurs. The hub allows the four nodes or sites to communicate as a single network instead of three separate systems. Hubbing reduces requirements
for back-to-back multiplexing and demultiplexing and helps realize the benefits of traffic grooming.

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Network providers no longer need to own and maintain customer-located
equipment. A multipoint implementation permits OCN interconnects or
midspan meet, allowing network providers and their customers to optimize
their shared use of the SONET infrastructure.
Convergence, ATM, Video, and SONET
Convergence is the trend toward delivery of audio, data, images, and video
through diverse transmission and switching systems that supply high-speed
transportation over any medium to any location. Tektronix is pursuing every
opportunity to lead the market, providing test and measurement equipment
to markets that process or transmit audio, data, image, and video signals over
high-speed networks.
With its modular, service-independent architecture, SONET provides vast capabilities in terms of service flexibility. Many of the new broadband services may
use asynchronous transfer mode (ATM)a fast packet-switching technique
using short, fixed-length packets called cells. Asynchronous transfer mode multiplexes the payload into cells that may be generated and routed as necessary.
Because of the bandwidth capacity it offers, SONET is a logical carrier for ATM.
In principle, ATM is quite similar to other packet-switching techniques; however,
the detail of ATM operation is somewhat different. Each ATM cell is made up of
53 octets or bytes (see Figure 30). Of these, 48 octets make up the user-information field and 5 octets make up the header. The cell header identifies the virtual
path to be used in routing the cell through the network. The virtual path defines
the connections through which the cell is routed to reach its destination.

Figure 30. ATM Cell Consists of Two Parts: A 5-Byte Header and a 48Byte Information Field
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An ATMbased network is bandwidth-transparent, which allows handling of
a dynamically variable mixture of services at different bandwidths. ATM also
easily accommodates traffic of variable speeds. An example of an application
that requires the benefits of variable-rate traffic is that of a video CODEC.
The video signals can be packed within ATM cells for transport.
Grooming
Grooming refers to either consolidating or segregating traffic to make more
efficient use of the facilities. Consolidation means combining traffic from different locations onto one facility.
Segregation is the separation of traffic. With existing systems, the cumbersome technique of back-hauling might be used to reduce the expense of
repeated multiplexing and demultiplexing.
Grooming eliminates inefficient techniques such as back-hauling. It is possible
to groom traffic on asynchronous systems; however, to do so requires expensive back-to-back configurations and manual DSX panels or electronic crossconnects. By contrast, a SONET system can segregate traffic at either an
STS1 or VT level to send it to the appropriate nodes.
Grooming can also provide segregation of services. For example, at an interconnect point, an incoming SONET line may contain different types of traffic,
such as switched voice, data, or video. A SONET network can conveniently
segregate the switched and nonswitched traffic.
Reduced Cabling and Elimination of DSX Panels
Asynchronous systems are dominated by back-to-back terminals because the
asynchronous fiber-optic transmission system architecture is inefficient for
other than point-to-point networks. Excessive multiplexing and demultiplexing are used to transport a signal from one end to another, and many bays of
DSX1 cross-connect and DSX3 panels are required to interconnect the systems. Associated expenses are the panel, bays, cabling, labor installation, and
the inconveniences of increased floor space and congested cable racks.
The corresponding SONET system allows a hub configuration, reducing the
need for back-to-back terminals. Grooming is performed electronically, so
DSX panels are not used except when required to interface with existing
asynchronous equipment.
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Enhanced OAM&P
SONET allows integrated network OAM&P, in accordance with the philosophy of single-ended maintenance. In other words, one connection can reach
all network elements (within a given architecture); separate links are not
required for each network element. Remote provisioning provides centralized
maintenance and reduced travel for maintenance personnelwhich translates
to expense savings.
Enhanced Performance Monitoring
Substantial overhead information is provided in SONET to allow quicker troubleshooting and detection of failures before they degrade to serious levels.

10. SDH REFERENCE


Following development of the SONET standard by ANSI, the CCITT undertook to define a synchronization standard that would address interworking
between the CCITT and ANSI transmission hierarchies. That effort culminated in 1989 with CCITTs publication of the synchronous digital hierarchy
(SDH) standards. Synchronous digital hierarchy is a world standard, and as
such SONET can be considered a subset of SDH.
Transmission standards in the United States, Canada, Korea, Taiwan, and
Hong Kong (ANSI) and the rest of the world (ITUT, formerly CCITT)
evolved from different basic-rate signals in the nonsynchronous hierarchy.
ANSI time-division multiplexing (TDM) combines 24 64kbps channels
(DS0s) into one 1.54Mbps DS1 signal. ITU TDM multiplexes 32 64kbps
channels (E0s) into one 2.048 Mbps E-1 signal.
The issues between ITUT and ANSI standards-makers involved how to efficiently accommodate both the 1.5Mbps and the 2Mbps nonsynchronous hierarchies
in a single synchronization standard. The agreement reached specifies a basic
transmission rate of 52 Mbps for SONET and a basic rate of 155 Mbps for SDH.
Synchronous and nonsynchronous line rates and the relationships between
each are shown in Tables 9 and 10.

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SONET Signal

Bit Rate

SDH Signal

SONET Capacity

SDH Capacity

STS1, OC1

51.840 Mbps

STM0

28 DS1s or 1 DS3

21 E1s

STS3, OC3

155.520 Mbps

STM1

84 DS1s or 3 DS3s

63 E1s or 1 E4

STS12, OC12

622.080 Mbps

STM4

336 DS1s or 12 DS3s

252 E1s or 4 E4s

STS48, OC48

2488.320 Mbps

STM16

1344 DS1s or 48 DS3s 1008 E1s or 16 E4s

STS192, OC192

9953.280 Mbps

STM64

5376 DS1s or 192 DS3s 4032 E1s or 64 E4s

It must be noted that although an SDH STM1 has the same bit rate as the
SONET STS3, the two signals contain different frame structures.
STM = synchronous transport module (ITUT)
STS = synchronous transfer signal (ANSI)
OC

= optical carrier (ANSI)

Table 9. SONET/SDH Hierarchies


ANSI Rate

ITUT Rate

Signal

Bit Rate

Channels

Signal

Digital Bit Rate

Channels

DS0

64 kbps

1 DS0

64kbps

64 kbps

1 64kbps

DS1

1.544 Mbps

24 DS0s

E1

2.048 Mbps

1 E1

DS2

6.312 Mbps

96 DS0s

E2

8.45 Mbps

4 E1s

DS3

44.7 Mbps

28 DS1s

E3

34 Mbps

16 E1s

E4

144 Mbps

64 E1s

not defined

Table 10. Nonsynchronous Hierarchies

Convergence of SONET and SDH Hierarchies


SONET and SDH converge at SONETs 52Mbps base level, defined as
STM0. The base level for SDH is STM1, which is equivalent to SONETs
STS3 (3 x 51.84 Mbps = 155.5 Mbps). Higher SDH rates are STM4 (622
Mbps) and STM16 (2.5 Gbps). STM64 (10 Gbps) has also been defined.

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Multiplexing is accomplished by combiningor interleavingmultiple lowerorder signals (1.5 Mbps, 2 Mbps, etc.) into higher-speed circuits (52 Mbps,
155 Mbps, etc.). By changing the SONET standard from bit-interleaving to
byte-interleaving, it became possible for SDH to accommodate both transmission hierarchies.

Asynchronous and Synchronous Tributaries


SDH does away with a number of the lower multiplexing levels, allowing
nonsynchronous 2Mbps tributaries to be multiplexed to the STM1 level in
a single step. SDH recommendations define methods of subdividing the payload area of an STM1 frame in various ways so that it can carry combinations of synchronous and asynchronous tributaries. Using this method, synchronous transmission systems can accommodate signals generated by equipment operating from various levels of the nonsynchronous hierarchy.

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11. SELF-TEST
Multiple Choice
1. Two digital signals whose transitions occur at almost the same rate are

a. asynchronous
b. synchronous
c. plesiochronous

2. SONET systems are

a. twisted-pair copper-based technology


b. fiber-optic technology
c. wireless technology

3. SONETs base signal (STS1) operates at a bit rate of

a. 64 kbps
b. 1.544 Mbps
c. 51.840 Mbps
d. 155.520 Mbps

4. N in the expression STSN indicates the

a. generation of the STS architecture


b. the integer multiple of the base transmission rate

5. Line overhead contains

a. 18 bytes of information
b. 9 bytes of information
c. 4 bytes of information

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6. Jitter _________ long-term variations in a waveform.
a. is
b. is not

7. The low-speed tributaries that make up a multiplexed SONET signal ______


individually accessible.
a. are
b. are not

8. SONET __________ multivendor compatibility.


a. enables
b. cannot enable

9. Which of following are not basic SONET network elements?


a. switch interface
b. digital loop carrier
c. service control point
d. add/drop multiplexer

10. _________ stuffing is used when the frame rate of the SPE is too slow in relation
to the rate of the STS1.
a. positive
b. negative

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12. ACRONYM GUIDE


Three sets of terms are often used interchangeably to describe SONET
processes. However, it is important to recognize that these terms are not
equivalent; each has a distinct meaning:
1. Add/DropThe process where a part of the information carried in a transmission system is demodulated (dropped) at an intermediate point and different
information is entered (added) for subsequent transmission. The remaining traffic passes straight through the multiplexer without additional processing.
2. Map/DemapA term for multiplexing, implying more visibility inside
the resultant multiplexed bit stream than available with conventional asynchronous techniques.
3. Multiplex/DemultiplexMultiplex allows the transmission two or more
signals over a single channel. Demultiplex is the process of separating previously combined signals and restoring the distinct individual channels of the signals.
Add/Drop Multiplexing (ADM)The process by which a part of the information carried in a transmission system is demodulated (dropped) at an intermediate point and different information is entered (added) for subsequent transmission. The remaining traffic passes straight through the multiplexer without additional processing.
Alarm Indicating Signal (AIS)A code sent downstream indicating that an
upstream failure has occurred. SONET defines four categories of AIS: line
AIS; STS path AIS, VT path AIS; DSn AIS.
Alternate Mark Inversion (AMI)The line-coding format in transmission systems where successive ones (marks) are alternatively inverted (sent with
polarity opposite that of the preceding mark).
American National Standards Institute (ANSI)A membership organization
which develops U.S. industry standards and coordinates U.S. participation in
the International Standards Organization (ISO).
AsynchronousA network where transmission system payloads are not synchronized, and each network terminal runs on its own clock.
Asynchronous Transfer Mode (ATM)A multiplexing/switching technique in
which information is organized into fixed-length cells with each cell consisting of an identification header field and an information field. The transfer

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mode is asynchronous in the sense that the use of the cells depends on the
required or instantaneous bit rate.
AttenuationReduction of signal magnitude or signal loss, usually expressed
in decibels.
Automatic Protection Switching (APS)The ability of a network element to
detect a failed working line and switch the service to a spare (protection) line.
1+1 APS pairs a protection line with each working line. 1:n APS provides one
protection line for every n working lines.
BandwidthInformation-carrying capacity of a communication channel.
Analog bandwidth is the range of signal frequencies that can be transmitted
by a communication channel or network.
BidirectionalOperating in both directions. Bidirectional APS allows protection switching to be initiated by either end of the line.
Binary N-Zero Suppression (BNZS)Line coding system that replaces N number of zeros with a special code to maintain pulse density required for synchronization. N is typically 3, 6, or 8.
BIP-8 (Bit Interleaved Parity-8)A method of error checking in SONET which
allows a full set of performance statistics to be generated. For example, a BIP8 creates 8-bit (one-byte) groups, then does a parity check for each of the 8bit positions in the byte.
BISDN (Broadband Integrated Services Digital Network)A single ISDN network which can handle voice, data, and eventually video services.
BitOne binary digit; a pulse of data.
Bit Error Rate (BER)The number of coding violations detected in a unit of
time, usually one second. Bit error rate (BER) is calculated with this formula:
BER = errored bits received/total bits sent
Bit Error vs. Block ErrorError rate statistics play a key role in measuring the performance of a network. As errors increase, user payload (especially data) must be retransmitted. The end effect is creation of more (nonrevenue) traffic in the network.

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Bit Interleaved Parity (BIP)A parity check that groups all the bits in a block into
units (such as byte), and then performs a parity check for each bit position in a
group.
Bit-StuffingIn asynchronous systems, a technique used to synchronize
asynchronous signals to a common rate before multiplexing.
Bit SynchronousA way of mapping payload into virtual tributaries that synchronizes all inputs into the VTs but does not capture any framing information or allow access to subrate channels carried in each input. For example, bit
synchronous mapping of a channeled DS1 into a VT1.5 does not provide
access to the DS0 channels carried by the DS1.
Bits per second (bps)The number of bits passing a point every second. The
transmission rate for digital information.
Block Error Rate (BLER)One of the underlying concepts of error performance
is the notion of errored blocks; for example, blocks in which one or more bits
are in error. A block is a set of consecutive bits associated with the path or section monitored by means of an error detection code (EDC), such as bit interleaved parity (BIP). Block error rate (BLER) is calculated with this formula:
BLER = errored blocks received/total block sent
BroadbandServices requiring 50600 Mbps transport capacity.
Byte InterleavedBytes from each STS1 are placed in sequence in a multiplexed or concatenated STSN signal. For example, for an STS3, the
sequence of bytes from contributing STS-1s is 1,2,3,1,2,3,...
Byte SynchronousA way of mapping payload into virtual tributaries that
synchronizes all inputs into the VTs, captures framing information, and
allows access to subrate channels carried in each input. For example, byte
synchronous mapping of a channeled DS1 into a VT1.5 provides direct access
to the DS0 channels carried by the DS1.
CCITTThe technical organs of the United Nations specialized agency for
telecommunications, now the International Telecommunications Union
Telecomm. They function through international committees of telephone
administrations and private operating agencies.
CEPT(European Conference of Postal and Telecommunications
Administrations). The CEPT format defines the 2.048Mbps European E-1
signal made up of 32 voice-frequency channels.
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ChannelThe smallest subdivision of a circuit that provides a type of communication service; usually a path with only one direction.
CircuitA communications path or network; usually a pair of channels providing bi-directional communication.
Circuit SwitchingBasic switching process whereby a circuit between two
users is opened on demand and maintained for their exclusive use for the
duration of the transmission.
Coding Violation (CV)A transmission error detected by the difference
between the transmitted and the locally calculated bit-interleaved parity.
ConcatenateThe linking together of various data structures, for example
two bandwidths joined to form a single bandwidth.
Concatenated STSNcA signal in which the STS envelope capacities from
the N STS1s have been combined to carry an STSNc synchronous payload
envelope (SPE). It is used to transport signals that do not fit into an STS1
(52Mbps) payload.
Concatenated VTA virtual tributary (VT x Nc) that is composed of N x VTs combined. Its payload is transported as a single entity rather than separate signals.
Cyclic Redundancy Check (CRC)A technique for using overhead bits to
detect transmission errors.
Data Communications ChannelsOAM&P channels in SONET that enable
communications between intelligent controllers and individual network nodes
as well as internode communications.
DefectA limited interruption in the ability of an item to perform a required
function.
DemultiplexingA process applied to a multiplex signal for recovering signals
combined within it and for restoring the distinct individual channels of the
signals.
Digital Cross-Connect (DCS)An electronic cross-connect that has access to
lower-rate channels in higher-rate multiplexed signals and can electronically
rearrange (cross-connect) those channels.
Digital SignalAn electrical or optical signal that varies in discrete steps. Electrical
signals are coded as voltages; optical signals are coded as pulses of light.

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DSX1May refer to either a cross-connect for DS1 rate signals or the signals
cross-connected at a DSX-1.
DSX3May refer to either a cross-connect for DS3 rate signals or the signals
cross-connected at a DSX-1.
ECSAExchange Carrier Standards Association An organization that specifies telecommunications standards for ANSI.
Envelope CapacityThe number of bytes the payload envelope of a single
frame can carry. The SONET STS payload envelope is the 783 bytes of the
STS-1 frame available to carry a signal. Each virtual tributary has an envelope
capacity defined as the number of bytes in the virtual tributary less the bytes
used by VT overhead.
FailureA termination of the ability of an item to perform a required function. A failure is caused by the persistence of a defect.
FEBE (Far End Block Error)A message sent back upstream to signify that a
receiving network element is detecting errors, usually a coding violation. See
remote error indication (REI).
Far End Receive Failure (FERF)A signal to indicate to the transmit site that a
failure has occurred at the receive site.
Fixed StuffA bit or byte whose function is reserved. Fixed stuff locations,
sometimes called reserved locations, do not carry overhead or payload.
Floating ModeA virtual tributary mode that allows the VT synchronous
payload envelope to begin anywhere in the VT. Pointers identify the starting
location of the VT SPE. VT SPEs in different superframes may begin at different locations.
FramingMethod of distinguishing digital channels that have been multiplexed together.
FrequencyThe number of cycles of periodic activity that occur in a discrete
amount of time.
GroomingConsolidating or segregating traffic for efficiency.
InterleaveThe ability of SONET to mix together and transport different types of
input signals in an efficient manner, thus allowing higher transmission rates.

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IsochronousAll devices in the network derive their timing signal directly or
indirectly from the same primary reference clock.
JitterShort waveform variations caused by vibration, voltage fluctuations,
control system instability, and etc.
LineOne or more SONET sections, including network elements at each end,
capable of accessing, generating, and processing line overhead.
Line Alarm Indication Signal (AISL)AISL is generated by Section
Terminating Equipment (STE) upon the detection of a loss of signal or loss of
frame defect on an equipment failure. AISL maintains operation of the downstream regenerators, and therefore prevents generation of unnecessary alarms.
At the same time, data and orderwire communication is retained between the
regenerators and the downstream Line Terminating Equipment (LTE).
Line Overhead (LOH)18 bytes of overhead accessed, generated, and
processed by line-terminating equipment. This overhead supports functions
such as locating the SPE in the frame, multiplexing or concatenating signals,
performance monitoring, automatic protection switching, and line maintenance.
Line Remote Defect Indication (RDIL)A signal returned to the transmitting
Line Terminating Equipment (LTE) upon detecting a loss of signal, loss of
frame, or AIS-L defect. RDI-L was previously known as Line FERF.
Line Terminating Equipment (LTE)Network elements such as add/drop
multiplexers or digital cross-connect systems that can access, generate, and
process line overhead.
Locked ModeA virtual tributary mode that fixes the starting location of the
VT SPE. Locked mode has less pointer processing than floating mode.
Map/DemapA term for multiplexing, implying more visibility inside the
resultant multiplexed bit stream than available with conventional asynchronous techniques.
MappingThe process of associating each bit transmitted by a service into
the SONET payload structure that carries the service. For example, mapping a
DS1 service into a SONET VT1.5 associates each bit of the DS1 with a location in the VT1.5.
MesochronousA network wherein all nodes are timed to a single clock
source, thus all timing is exactly the same (truly synchronous).

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Multiplex (MUX)To transmit two or more signals over a single channel.
MultiplexerA device for combining several channels to be carried by one line or fiber.
NarrowbandServices requiring up to 1.5 Mbps transport capacity.
Network Element (NE)Any device that is part of a SONET transmission
path and serves one or more of the section, line, and path-terminating functions. In SONET, the five basic network elements are as follows:
add/drop multiplexer
broadband digital cross-connect
wideband digital cross-connect
digital loop carrier
switch interface
Operations, Administration, and Maintenance (OA&M)Also called OAM&P.
Operations, Administration, Maintenance, and Provisioning (OAM&P)
Provides the facilities and personnel required to manage a network.
Optical Carrier Level N (OCN)The optical equivalent of an STSN signal.
Optical Carrier Level 1 (OC1)The optical equivalent of an STS1 signal.
OrderwireA channel used by installers to expedite the provisioning of lines.
Operations System (OS)Sophisticated applications software that oversees
the entire network.
OSI Seven-Layer Model A standard architecture for data communications.
Layers define hardware and software required for multi-vendor information
processing equipment to be mutually compatible. The seven layers from lowest to highest are: physical, link, network, transport, session, presentation,
and application.
OverheadExtra bits in a digital stream used to carry information besides traffic
signals. Orderwire, for example, would be considered overhead information.
Packet SwitchingAn efficient method for breaking down and handling highvolume traffic in a network. A transmission technique that segments and
routes information into discrete units. Packet switching allows for efficient
sharing of network resources, as packets from different sources can all be sent
over the same channel in the same bitstream.

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Parity CheckAn error-checking scheme which examines the number of
transmitted bits in a block which hold the value one. For even parity, an overhead parity bit is set to either one or zero to make the total number of transmitted ONES an even number. For odd parity, the parity bit is set to make the
total number of ONES transmitted an odd number.
PathA logical connection between a point where an STS or VT is multiplexed to the point where it is demultiplexed.
Path Overhead (POH)Overhead accessed, generated, and processed by pathterminating equipment. Path overhead includes 9 bytes of STS path overhead
and, when the frame is VTstructured, 5 bytes of VT path overhead.
Path Terminating Equipment (PTE)Network elements, such as fiber-optic terminating systems, that can access, generate, and process path overhead.
PayloadThe portion of the SONET signal available to carry service signals
such as DS1 and DS3. The contents of an STS SPE or VT SPE.
Payload PointerIndicates the beginning of the synchronous payload envelope.
PhotonThe basic unit of light transmission used to define the lowest (physical) layer in the OSI seven-layer model.
PlesiochronousA network with nodes timed by separate clock sources with
almost the same timing.
PointerA part of the SONET overhead that locates a floating payload structure. STS pointers locate the SPE. VT pointers locate floating mode virtual
tributaries. All SONET frames use STS pointers; only floating mode virtual
tributaries use VT pointers.
PollAn individual control message from a central controller to an individual
station on a multipoint network inviting that station to send.
POP (Point-of-Presence)A point in the network where interexchange carrier
facilities like DS3 or OCN meet with access facilities managed by telephone
companies or other service providers.
RegeneratorDevice that restores a degraded digital signal for continued
transmission; also called a repeater.
Remote Alarm Indication (RAI)A code sent upstream in a DSn network as a
notification that a failure condition has been declared downstream. (RAI signals were previously referred to as yellow signals.)
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Remote Defect Indication (RDI)A signal returned to the transmitting terminating equipment upon detecting a loss of signal, loss of frame, or AIS defect.
RDI was previously known as FERF.
Remote Error Indication (REI)An indication returned to a transmitting node
(source) that an errored block has been detected at the receiving node (sink).
This indication was formerly known as far end block error (FEBE).
Remote Failure Indication (RFI)A failure is a defect that persists beyond the
maximum time allocated to the transmission system protection mechanisms.
When this situation occurs, an RFI is sent to the far end and will initiate a
protection switch if this function has been enabled.
Synchronous Digital Hierarchy (SDH)The ITUTdefined world standard
of transmission whose base transmission level is 52 Mbps (STM0) and is
equivalent to SONETs STS1 or OC1 transmission rate. SDH standards
were published in 1989 to address interworking between the ITUT and
ANSI transmission hierarchies.
SectionThe span between two SONET network elements capable of accessing, generating, and processing only SONET section overhead. This is the
lowest layer of the SONET protocol stack with overhead.
Section Overhead9 bytes of overhead accessed, generated, and processed
by section-terminating equipment. This overhead supports functions such as
framing the signal and performance monitoring.
Section Terminating Equipment (STE)Equipment that terminates the
SONET section layer. STE interprets and modifies or creates the section overhead.
SlipAn overflow (deletion) or underflow (repetition) of one frame of a signal in a receiving buffer.
Synchronous Optical Network (SONET)A standard for optical transport
that defines optical carrier levels and their electrically equivalent synchronous
transport signals. SONET allows for a multivendor environment and positions
the network for transport of new services, synchronous networking, and
enhanced OAM&P.

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SPE (Synchronous Payload Envelope)The major portion of the SONET
frame format used to transport payload and STS path overhead. A SONET
structure that carries the payload (service) in a SONET frame or virtual tributary. The STS SPE may begin anywhere in the frames payload envelope. The
VT SPE may begin anywhere in a floating-mode VT, but begins at a fixed
location in a locked-mode VT.
StratumLevel of clock source used to categorize accuracy.
STSN (Synchronous Transport Signal Level N)The signal obtained by
multiplexing integer multiples (N) of STS-1 signals together.
STS1 (Synchronous Transport Signal Level 1)The basic SONET buildingblock signal transmitted at 51.84 Mbps data rate.
STS Path Overhead (STS POH)Nine evenly distributed Path Overhead
bytes per 125 microseconds starting at the first byte of the STS SPE. STS POH
provides for communication between the point of creation of an STS SPE and
its point of disassembly.
STS Path Remote Defect Indication (RDI-P)A signal returned to the transmitting STS path terminating equipment (PTE) upon detection of certain
defects on the incoming path.
STS Path Terminating Equipment (STS PTE)Equipment that terminates the
SONET STS Path layer. STS PTE interprets and modifies or creates the STS
path overhead. An NE that contains STS PTE will also contain LTE and STE.
SuperframeAny structure made of multiple frames. SONET recognizes
superframes at the DS1 level (D4 and extended superframe) and at the VT
(500 s STS superframes).
SynchronousA network where transmission system payloads are synchronized to a master (network) clock and traced to a reference clock.
Synchronous Transfer Module (STM)An element of the SDH transmission
hierarchy. STM1 is SDHs base-level transmission rate equal to 155 Mbps.
Higher rates of STM4, STM16, and STM48 are also defined.
T1X1 SubcommitteeA committee within ANSI that specifies SONET optical interface rates and formats.
VT (Virtual Tributary)A signal designed for transport and switching of
subSTS1 payloads.

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VT GroupA 9 row x 12 column structure (108 bytes) that carries one or
more VTs of the same size. Seven VT groups can be fitted into one STS1
payload.
VT Path Overhead (VT POH)4 evenly distributed path overhead bytes per
VT SPE, starting at the first byte of the VT SPE. VT POH provides for communication between the point of creation of a VT SPE and its point of
disassembly.
VT Path Remote Defect Indication (RDIV)A signal returned to the transmitting VT PTE upon detection of certain defects on the incoming path.
VT Path Remote Failure Indication (RFIV)A signal, applicable only to a
VT1.5 with the byte-synchronous DS1 mapping, that is returned to the
transmitting VT PTE upon declaring certain failures. The RFIV signal was
previously known as the VT Path Yellow signal.
VT Path Terminating Equipment (VT PTE)Equipment that terminates the
SONET VT Path layer. VT PTE interprets and modifies or creates the VT path
overhead. An NE that contains VT PTE will also contain STS PTE, LTE, and STE.
WanderLong-term variations in a waveform.
WidebandServices requiring 1.5 to 50 Mbps transport capacity.
Yellow SignalSee remote alarm indication (RAI) and VT path remote failure
indication (RFIV).

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Voice and Fax over Internet


Protocol (V/FoIP)
DEFINITION
A voice-overInternet protocol (VoIP) application meets the challenges of
combining legacy voice networks and packet networks by allowing both
voice and signaling information to be transported over the packet network. A
fax-overInternet protocol (FoIP) application enables the interworking of
standard fax machines with packet networks. It accomplishes this by extracting the fax image from an analog signal and carrying it as digital data over the
packet network.

TUTORIAL OVERVIEW
Organizations around the world seek to reduce rising communications costs.
The consolidation of separate voice, fax, and data resources offers an opportunity for significant savings. Accordingly, the challenge of integrating voice,
fax, and data is becoming a rising priority for many network managers.
Organizations are pursuing solutions that will enable them to take advantage
of excess capacity on broadband networks for voice, fax, and data transmission, as well as utilize the Internet and company Intranets as alternatives to
costlier mediums.
This tutorial discusses the principles related to implementing real-time voiceand fax-over-packet networks. An overview of the embedded software architecture is presented, and a system is described for sending voice, fax image,
data, and signaling information over the packet network. Benefits to designers
and manufacturers of this embedded approach are lower cost of goods sold,
faster time to market, and lower development costs. Customers can gain a
considerable advantage in time to market in building their communication
systems.

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This tutorial references a general class of packet networks, as the modular
software objects allow networks such as asynchronous transfer mode (ATM),
frame relay, and Internet/intranet (Internet protocol [IP]) to transport voice
and fax. Currently, the Frame Relay Forum and the International
Telecommunication Union (ITU) have defined protocols for transmission of
fax over a packet network. However, the principles described are equally
applicable to ATM networks.

TOPICS
1. VOIP APPLICATIONS................................................................................389
2. VOIP QOS ISSUES ....................................................................................391
3. VOIP EMBEDDED SOFTWARE ARCHITECTURE ................................394
4. VOICE PACKET MODULE ......................................................................396
5. SIGNALING, PROTOCOL, AND MANAGEMENT MODULES ..........399
6. VOIP SUMMARY ......................................................................................401
7. FOIP APPLICATIONS ................................................................................402
8. PSTN FAX CALL PROCEDURE ................................................................403
9. FOIP QOS ..................................................................................................406
10. FOIP SOFTWARE ARCHITECTURE ......................................................408
11. FOIP SUMMARY......................................................................................410
12. SELF-TEST ................................................................................................411
13. ACRONYM GUIDE ................................................................................414

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1. VOIP APPLICATIONS
A wide variety of applications are enabled by the transmission of VoIP networks. This tutorial will explore three examples of these applications.
The first application, shown in Figure 1, is a network configuration of an organization with many branch offices (e.g., a bank) that wants to reduce costs
and combine traffic to provide voice and data access to the main office. This
is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the
data. Typically, this network configuration will benefit if the voice traffic is
compressed as a result of the low bandwidth available for this access application. Voice over packet provides the interworking function (IWF), which is
the physical implementation of the hardware and software that allows the
transmission of combined voice and data over the packet network. The interfaces the IWF must support in this case are analog interfaces, which directly
connect to telephones or key systems. The IWF must emulate the functions
of both a private branch exchange (PBX) for the telephony terminals at the
branches, as well as the functions of the telephony terminals for the PBX at
the home office. The IWF accomplishes this by implementing signaling software that performs these functions.

Figure 1. Branch Office Application


A second VoIP application, shown in Figure 2, is a trunking application. In this
scenario, an organization wishes to send voice traffic between two locations
over the packet network and replace the tie trunks used to connect the PBXs
at the locations. This application usually requires the IWF to support a highercapacity digital channel than the branch application, such as a T1/E1

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interface of 1.544 or 2.048 Mbps. The IWF emulates the signaling functions of
a PBX, resulting in significant savings to companies' communications costs.

Figure 2. Interoffice Trunking Application


A third application of VoIP software is interworking with cellular networks,
as shown in Figure 3. The voice data in a digital cellular network is already
compressed and packetized for transmission over the air by the cellular
phone. Packet networks can then transmit the compressed cellular voice packet, saving a tremendous amount of bandwidth. The IWF provides the
transcoding function required to convert the cellular voice data to the format
required by the public switched telephone network (PSTN).

Figure 3. Interoffice Trunking Application


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2. VOIP QOS ISSUES


The advantages of reduced cost and bandwidth savings of carrying voiceover-packet networks are associated with some quality-of-service (QoS)
issues unique to packet networks.
Delay
Delay causes two problems: echo and talker overlap. Echo is caused by the
signal reflections of the speaker's voice from the far-end telephone equipment
back into the speaker's ear. Echo becomes a significant problem when the
round-trip delay becomes greater than 50 milliseconds. As echo is perceived
as a significant quality problem, voice-over-packet systems must address the
need for echo control and implement some means of echo cancellation.
Talker overlap (or the problem of one talker stepping on the other talker's
speech) becomes significant if the one-way delay becomes greater than 250
milliseconds. The end-to-end delay budget is therefore the major constraint
and driving requirement for reducing delay through a packet network.
The following are sources of delay in an end-to-end, voice-over-packet call:
Accumulation Delay (Sometimes Called Algorithmic Delay)
This delay is caused by the need to collect a frame of voice samples to be
processed by the voice coder. It is related to the type of voice coder used and
varies from a single sample time (.125 microseconds) to many milliseconds. A
representative list of standard voice coders and their frame times follows:
* G.726 adaptive differential pulse-code modulation (ADPCM) (16, 24, 32, 40
kbps)0.125 microseconds
* G.728 LDcode excited linear prediction (CELP)(16 kbps)2.5 milliseconds
* G.729 CSACELP (8 kbps)10 milliseconds
* G.723.1 Multirate Coder (5.3, 6.3 kbps)30 milliseconds
Processing Delay
This delay is caused by the actual process of encoding and collecting the
encoded samples into a packet for transmission over the packet network. The
encoding delay is a function of both the processor execution time and the
type of algorithm used. Often, multiple voice-coder frames will be collected

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in a single packet to reduce the packet network overhead. For example, three
frames of G.729 code words, equaling 30 milliseconds of speech, may be collected and packed into a single packet.
Network Delay
This delay is caused by the physical medium and protocols used to transmit
the voice data and by the buffers used to remove packet jitter on the receive
side. Network delay is a function of the capacity of the links in the network
and the processing that occurs as the packets transit the network. The jitter
buffers add delay, which is used to remove the packet-delay variation to
which each packet is subjected as it transits the packet network. This delay
can be a significant part of the overall delay, as packet-delay variations can be
as high as 70 to 100 milliseconds in some frame-relay and IP networks.
Jitter
The delay problem is compounded by the need to remove jitter, a variable
interpacket timing caused by the network a packet traverses. Removing jitter
requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence. This causes
additional delay.
The two conflicting goals of minimizing delay and removing jitter have
engendered various schemes to adapt the jitter buffer size to match the timevarying requirements of network jitter removal. This adaptation has the
explicit goal of minimizing the size and delay of the jitter buffer, while at the
same time preventing buffer underflow caused by jitter.
Two approaches to adapting the jitter buffer size are detailed below. The
approach selected will depend on the type of network the packets are traversing.
The first approach is to measure the variation of packet level in the jitter
buffer over a period of time and incrementally adapt the buffer size to match
the calculated jitter. This approach works best with networks that provide a
consistent jitter performance over time, such as ATM networks.
The second approach is to count the number of packets that arrive late and
create a ratio of these packets to the number of packets that are successfully
processed. This ratio is then used to adjust the jitter buffer to target a

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predetermined, allowable late-packet ratio. This approach works best with
the networks with highly variable packet-interarrival intervalssuch as IP
networks.
In addition to the techniques described, the network must be configured and
managed to provide minimal delay and jitter, enabling a consistent QoS.
Lost-Packet Compensation
Lost packets can be an even more severe problem, depending on the type of
packet network that is being used. Because IP networks do not guarantee service, they will usually exhibit a much higher incidence of lost voice packets
than ATM networks. In current IP networks, all voice frames are treated like
data. Under peak loads and congestion, voice frames will be dropped equally
with data frames. The data frames, however, are not time sensitive, and
dropped packets can be appropriately corrected through the process of
retransmission. Lost voice packets, however, cannot be dealt with in this
manner.
Some schemes used by voice-over-packet software to address the problem of
lost frames are as follows:
* interpolate for lost speech packets by replaying the last packet received during the interval when the lost packet was supposed to be played out; this
scheme is a simple method that fills the time between noncontiguous
speech frames; it works well when the incidence of lost frames is infrequent; it does not work well if there are a number of lost packets in a row
or a burst of lost packets
* send redundant information at the expense of bandwidth utilization; this
basic approach replicates and sends the nth packet of voice information
along with the (n+1)th packet; this method has the advantage of being able
to correct for the lost packet exactly; however, this approach uses more
bandwidth and also creates greater delay
* use a hybrid approach with a much lower bandwidth voice coder to provide redundant information carried along in the (n+1)th packet; this reduces
the problem of the extra bandwidth required but fails to solve the problem
of delay

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Echo Compensation
Echo in a telephone network is caused by signal reflections generated by the
hybrid circuit that converts between a four-wire circuit (a separate transmit
and receive pair) and a two-wire circuit (a single transmit and receive pair).
These reflections of the speaker's voice are heard in the speaker's ear. Echo is
present even in a conventional circuit-switched telephone network. However,
it is acceptable because the round-trip delays through the network are smaller
than 50 milliseconds and the echo is masked by the normal side tone every
telephone generates.
Echo becomes a problem in voice-over-packet networks because the roundtrip delay through the network is almost always greater than 50 milliseconds.
Thus, echo-cancellation techniques are always used. ITU standard G.165
defines performance requirements that are currently required for echo cancellers. The ITU is defining much more stringent performance requirements in
the G.IEC specification.
Echo is generated toward the packet network from the telephone network.
The echo canceller compares the voice data received from the packet network
with voice data being transmitted to the packet network. The echo from the
telephone network hybrid is removed by a digital filter on the transmit path
into the packet network.

3. VOIPEMBEDDED SOFTWARE ARCHITECTURE


Two major types of information must be handled to interface telephony
equipment to a packet network: voice and signaling information.
As shown in Figure 4, VoIP software interfaces to both streams of information
from the telephony network and converts them to a single stream of packets
transmitted to the packet network. The software functions are divided into
four general areas.

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Figure 4. VoIP Software Architecture


Voice Packet Software Module
This software, also known as the voice-processing module, typically runs on
a digital-signal processor (DSP), prepares voice samples for transmission over
the packet network. Its components perform echo cancellation, voice compression, voice-activity detection, jitter removal, clock synchronization, and
voice packetization.
Telephony-Signaling Gateway Software Module
This software interacts with the telephony equipment, translating signaling
into state changes used by the packet protocol module to set up connections.
These state changes are on-hook, off-hook, trunk seizure, etc. This software
supports ear, mouth, earth, and magneto (E&M) Type I, II, III, IV, and V; loop
or ground start foreign exchange station (FXS); foreign exchange office (FXO);
and integrated services digital network (ISDN) basic rate interface (BRI) and
primary rate interface (PRI).
Packet Protocol Module
This module processes signaling information and converts it from the telephony-signaling protocols to the specific packet-signaling protocol used to set
up connections over the packet network (e.g., Q.933 and voice-over-frame
relay signaling). It also adds protocol headers to both voice and signaling
packets before transmission into the packet network.

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Network-Management Module
This module provides the voice-management interface to configure and maintain the other modules of the voice-over-packet system. All management
information is defined in American National Standards Institute (ANSI).1 and
complies with signaling network-management protocol (SNMP) V1 syntax. A
proprietary voice packet management information base (MIB) is supported
until standards evolve in the forums.
The software is partitioned to provide a well-defined interface to the DSP
software usable for multiple voice packet protocols and applications. The DSP
processes voice data and passes voice packets to the microprocessor with
generic voice headers.
The microprocessor is responsible for moving voice packets and adapting the
generic voice headers to the specific voice packet protocol that is called for by
the application, such as real-time protocol (RTP), voice over frame relay
(VoFR), and voice telephony over ATM (VToA). The microprocessor also
processes signaling information and converts it from supported telephony-signaling protocols to the packet network signaling protocol [e.g. H.323 IP, frame
relay, or ATM signaling].
This partitioning provides a clean interface between the generic voice-processing functions, such as compression, echo cancellation, and voice-activity
detection, and the application-specific signaling and voice protocol processing.

4. VOICE PACKET MODULE


This section describes the functions performed by the software in the voice
packet module, also known as the voice-processing module, which is primarily responsible for processing the voice data. This function is usually performed in a DSP. The voice-processing module consists of the following software:
* PCM interfaceThis receives pulse code modulation (PCM) samples from
the digital interface and forwards them to appropriate DSP software modules for processing, forwards processed PCM samples received from various
DSP software modules to the digital interface, and performs continuous
phase resampling of output samples to the digital interface to avoid sample
slips.

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* tone generatorThis generates dual-tone multifrequency (DTMF) tones
and call progress tones under command of the host (e.g., telephone, fax,
modem, PBX, or telephone switch) and is configurable for support of U.S.
and international tones.
* echo cancellerThis performs G.165compliant echo cancellation on
sampled, full-duplex voice port signals. It has a programmable range of tail
lengths.
* voice activation detector/idle noise measurementThis monitors the
received signal for voice activity. When no activity is detected for the configured period of time, the software informs the packet voice protocol. This
prevents the encoder output from being transported across the network
when there is silence, resulting in additional bandwidth savings. This software also measures the idle noise characteristics of the telephony interface.
It reports this information to the packet voice protocol to relay this information to the remote end for noise generation when no voice is present.
* tone detectorThis detects the reception of DTMF tones and performs
voice/fax discrimination. Detected tones are reported to the host so that the
appropriate speech or fax functions are activated.
* voice codec softwareThis compresses the voice data for transmission
over the packet data. It is capable of numerous compression ratios through
the modular architecture. A compression ratio of 8:1 is achievable with the
G.729 voice codec (thus, the normal 64kbps PCM signal is transmitted
using only 8 kbps).
* fax softwareThis performs a fax-relay function by demodulating PCM
data, extracting the relevant information, and packing the fax-line scan data
into frames for transmission over the packet network. Significant bandwidth savings can be achieved by this process.
* voice playout unitThis buffers voice packets received from the packet
network and sends them to the voice codec for playout.
The following features are supported:
* a first in, first out (FIFO) buffer that stores voice code words before playout
removes timing jitter from the incoming packet sequence

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* a continuous-phase resampler that removes timing-frequency offset without causing packet slips or loss of data for voice- or voiceband-modem signals
* a timing jitter measurement that allows adaptive control of FIFO delay
The voice-packetization protocols use a sequence-number field in the transmit packet stream to maintain temporal integrity of voice during playout.
Using this approach, the transmitter inserts the contents of a free-running,
modulo-16 packet counter into each transmitted packet, allowing the receiver
to detect lost packets and to reproduce silence intervals during playout properly.
* packet voice protocolThis encapsulates compressed voice and fax data
for end-to-end transmission over a backbone network between two ports.
* control interface softwareThis coordinates the exchange of monitor
and control information between the DSP and host via a mailbox mechanism. Information exchanged includes software downline load, configuration data, and status reporting.
* real-time portability environmentThis provides the operating environment for the software residing on the DSP. It provides synchronization
functions, task management, memory management, and timer management.
Figure 5 diagrams the architecture of the DSP software. The DSP software
processes PCM samples from the telephony interface and converts them to a
digital format suitable for transmission through a packet network.

Figure 5. Voice Packet Module

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5. SIGNALING, PROTOCOL, AND MANAGEMENT MODULES


The VoIP software performs telephony signaling to detect the presence of a
new call and to collect address (dial digit) information, which is used by the
system to route a call to a destination port. It supports a wide variety of telephony-signaling protocols and can be adaptable to many environments. The
software and configuration data for the voice card can be downloaded from a
network-management system to allow customization, easy installation, and
remote upgrades.
The software interacts with the DSP for tone detection and generation, as
well as mode of operation control based on the line supervision, and interacts
with the telephony interface for signaling functions. The software receives
configuration data from the network-management agent and utilizes operating-system services.
Telephony-Signaling Gateway Module
Figure 6 diagrams the architecture of the signaling software, which consists of
the following components:
* telephony interface unit softwareThis periodically monitors the signaling interfaces of the module and provides basic debouncing and rotary
digit collection for the interface.
* signaling protocol unitThis contains the state machines implementing
the various telephony-signaling protocols, such as E&M.
* network control unitThis maps telephony-signaling information into a
format compatible with the packet voice session establishment signaling
protocol.
* address translation unitThis maps the E.164 dial address to an address
that can be used by the packet network (e.g., an IP address or a data link
connection indentifier (DLCI) for a frame-relay network).
* DSP interface driverThis relays control information between the host
microprocessor and DSPs.
* DSP downline loaderThis is responsible for downline load of the DSPs
at start-up, configuration update, or mode changes (e.g., switching from
voice mode to fax mode when fax tones are detected).

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Figure 6. Signaling Modules


Network-Protocol Module
* IP signaling stackThis involves H.323 call control and transport software, including H.225, H.245, RTP/real-time conferencing protocol (RTCP)
transport protocol, transmission control protocol (TCP), IP, and user datagram protocol (UDP).

* ATM signaling protocol stackATM Forum VToA voice-encapsulation


protocol. ATM Forumcompliant, user-network interface (UNI) signaling
protocol stack for establishing, maintaining, and clearing point-to-point and
point-to-multipoint switched virtual circuits (SVCs).

* frame-relay protocol stackThis includes Frame Relay Forum VoFR


voice-encapsulation protocol, permanent virtual circuit (PVC) and SVC support, local management interface (LMI), congestion management, traffic
monitoring, and committed information range (CIR) enforcement.

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Network-Management Module
The network-management software consists of three major services
addressed in the MIB:
1. physical interface to the telephone endpoint
2. voice channel service for the following:
* processing signaling on a voice channel
* converting between PCM samples and compressed voice packets
3. call-control service for parsing call-control information and establishing
calls between telephony endpoints
The VoIP software is configured and maintained through the use of a proprietary voice service MIB.

6. VOIP SUMMARY
A VoIP software architecture has been described for the interworking of legacy telephony systems and packet networks. Some of the key features
enabling this application to function successfully are as follows:
* an approach that minimizes the effects of delay on voice quality
* an adaptive playout to minimize the effect of jitter
* features that address lost-packet compensation, clock synchronization, and
echo cancellation
* a flexible DSP system architecture that manages multiple channels per single DSP
Carrying VoIP networks provides the most bandwidth-efficient method of
integrating these divergent technologies. While the challenges to this integration are substantial, the potential savings make the investment in a quality
implementation compelling.

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7. FOIP APPLICATIONS
Traditionally, there have been two approaches for sending FoIP networks:
real-time methods and store-and-forward methods. The primary difference in
service between these two approaches lies in the delivery and method of
receipt confirmation. The Frame Relay Forum has defined a real-time protocol
for the transmission of fax-overframe relay networks. Likewise, the ITU and
Internet Engineering Task Force (IETF) are working together to continue to
evolve both the real-time FoIP network standard (T.38) as well as the storeand-forward FoIP network standard (T.37). Both T.37 and T.38 were approved
by the ITU in June, 1998. Furthermore, T.38 is the fax transmission protocol
selected for H.323.
There are tremendous opportunities for cost savings by transmitting fax calls
over packet networks. Fax data in its original form is digital. However, it is
modulated and converted to analog for transmission over the PSTN. This analog form uses 64 kbps of bandwidth in both directions.
The FoIP IWF reverses this analog conversion, transmitting digital data over
the packet network and then reconverting the digital data to analog for the
receiving fax machine. This conversion process reduces the overall bandwidth
required to send the fax, because the digital form is much more efficient, and
the fax transmission is half-duplex (i.e., only one direction is used at any
time). The peak rate for a fax transmission is 14.4 kbps in one direction. A
representation of this process is shown in Figure 7.

Figure 7. FoIP Conversion Process


An application for fax over packet, shown in Figure 8, is a network configuration of a company with numerous branch offices that wants to use the packet
network, instead of the long-distance network, to provide fax access to the
main office. The IWF is the physical implementation of the hardware and
software that enables the transmission of fax over the packet network. The

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IWF must support analog interfaces that directly interface to fax machines at
the branches and to a PBX at the central site. The IWF must emulate the functions of a PBX for the fax machines.

Figure 8. FoIP Application

8. PSTN FAX-CALL PROCEDURE


This module will describe the stages of a standard fax call over the PSTN so
that the processing required for a reliable fax transmission over a packet network can be explored. Fax machines in common use today implement the
ITU recommendations T.30 and T.4 protocols. The T.30 protocol describes
the formatting of non-page data, such as messages that are used for capabilities negotiation. The T.4 protocol describes formatting of page image data.
T.30 and T.4 have evolved substantially over time and are now quite complex, given that they attempt to describe the behavior of an evolving set of
fax machines. The timing related to the message interaction and phases of the
call is critical and is one of the major causes of problems in the transmission
of FoIP networks.
The PSTN fax call is divided into five phases, as shown in Figure 9. This
example assumes that the call is accomplished without errors. The procedure
becomes somewhat more complicated if errors occur or if there is a need for
modem retraining. The five phases are as follows:
* call establishment
* control and capabilities exchange

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* page transfer
* end of page and multipage signaling
* call release

Figure 9. PSTN Fax-Call Flow


Call Establishment
The fax call is established either through a manual process, according to
which someone dials a call and puts the machine into fax mode, or by automatic procedures, according to which no human interaction is required. In
both cases, the answering fax machine returns an answer tone, called a CallEd
station IDentification (CED), which is the high-pitched tone that you would
hear when you call a fax machine. If the call is automatically dialed, the calling station will also indicate the fax call with a calling tone (CNG), which is a
short, periodic tone that begins immediately after the number is dialed. These

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tones are generated to allow a human participant to realize that a machine is
present on the other end call. They are also, although not reliably, used to recognize the presence of a fax call.
Control and Capabilities Exchange
The control and capabilities exchange phase of the fax call is used to identify
the capabilities of the fax machine at the other end of the call. It also negotiates the acceptable conditions for the call. The exchange of control messages
throughout the fax call are sent using the low-speed (300 bps) modulation
mode. Every control message is preceded by a one-second preamble, which
allows the communication channel to be conditioned for reliable transmission.
The called fax machine begins the procedure by sending a digital identification signal (DIS) message, which contains the capabilities of the fax machine.
An example of a capability that could be identified in this message is the support of the V.17 (14,000 bps) data signaling rate. At the same time, the called
subscriber information (CSI) and nonstandard facilities (NSF) messages are
optionally sent.
NSF are capabilities that a particular fax manufacturer has built into a fax
machine to distinguish its product from others. These facilities are not
required to be supported for interoperability.
Once the calling fax machine receives the DIS message, it determines the conditions for the call by examining its own capabilities table. The calling
machine responds with the digital command signal (DCS), which defines the
conditions of the call.
At this stage, high-speed modem training begins. The high-speed modem will
be used in the next phase of the fax call to transfer page data. The calling fax
machine sends a training check field (TCF) through the modulation system to
verify the training and ensure that the channel is suitable for transmission at
the accepted data rate. The called fax machine responds with a confirmation
to receive (CFR), which indicates that all capabilities and the modulation
speed have been confirmed and the fax page may be sent.
Page Transfer
The high-speed modem is used to transmit the page data that has been
scanned in and compressed. It uses the ITU T.4 protocol standard to format
the page data for transmission over the channel.
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End-of-Page and Multipage Signaling
After the page has been successfully transmitted, the calling fax machine
sends an end-of-procedures (EOP) message if the fax call is complete and all
of the pages have been transmitted. If only one page has been sent and there
are additional ones to follow, it sends a multipage signal (MPS). The called
machine would respond with message confirmation (MCF) to indicate the
message has been successfully received and that it is ready to receive more
pages.
Call Release
The release phase is the final phase of the call, in which the calling machine
sends a disconnect message (DCN). While the DCN message is a positive
indication that the fax call is over, it is not a reliable indication, as the fax
machine can disconnect prematurely without ever sending the DCN message.

9. FOIP QOS
The advantages of reduced cost and bandwidth savings of carrying FoIP networks are associated with some QoS issues that are unique to packet networks and can affect the reliability of the fax transmission.
Timing
A major issue in the implementation of FoIP networks is the problem of inaccurate timing of messages caused by delay through the network. The delay of
fax packets through a packet network causes the precise timing that is
required for many portions of the fax protocol to be skewed and can result in
the loss of the call. The FoIP protocol in the IWF must compensate for the
loss of a fixed timing of messages over the packet network so that the T.30
protocol operates without error.
There are two sources of delay in an end-to-end, FoIP call: network delay and
processing delay.
1. network delayThis is caused by the physical medium and protocols
that are used to transmit the fax data and by buffers used to remove packet jitter on the receiving end. This delay is a function of the capacity of the
links in the network and the processing that occurs as the packets transit

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the network. The jitter buffers add delay when they remove the packetdelay variation of each packet as it transits the packet network. This delay
can be a significant part of the overall delay, as packet-delay variations can
be as high as 70 to 100 milliseconds in some frame-relay networks and
even higher in IP networks.
2. processing delayThis is caused by the process of demodulating and
collecting the digital fax information into a packet for transmission over
the packet network. The encoding delay is a function of both the processor execution time and the amount of data collected before sending a
packet to the network. Low-speed data, for instance, is usually sent out
with a single byte per packet, as the time to collect a byte of information
at 300 bps is 30 milliseconds.
Jitter
Delay issues are compounded by the need to remove jitter, a variable interpacket timing caused by the network that a packet traverses. An approach to
removing the jitter is to collect packets and hold them long enough so that
the slowest packets to arrive are still in time to be played in the correct
sequence. This approach, however, causes additional delay. In most FoIP protocols, a time stamp is incorporated in the packet to ensure that the information is played out at the proper instant.
Lost-Packet Compensation
Lost packets can be an even more severe problem, depending on the type of
packet network that is being used. In a VoIP application, the loss of packets
can be addressed by replaying last packets and other methods of interpolation. A FoIP application, however, has more severe constraints on the loss of
data, as the fax protocol can fail if information is lost. This problem varies,
depending on the type of fax machine used and whether error-correction
mode is enabled.
Two schemes that are used by FoIP software to address the problems of lost
frames are as follows:
* repeating information in subsequent frames so that the error can be corrected by the receiver's playout mechanism
* using an error-correcting protocol such as TCP to transport the fax data at
the expense of added delay

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10. FOIP SOFTWARE ARCHITECTURE


The facsimile interface unit (FIU) is the software module that resides within a
FoIP IWF. It demodulates voiceband signals from an analog interface and converts them to a digital format suitable for transport over a packet network. It
also remodulates data received from the packet network and transmits it to
the analog interface. In doing so, the FIU performs protocol conversion
between Group-3 facsimile protocols and the digital facsimile protocol
employed over the packet network.
The FIU, shown in Figure 10, consists of the following three units:
fax-modem unit (FM)This processes PCM samples based on the current
modulation mode and supports the following functions:
* V.21 Channel 2 (300 bps) binary signaling modulation and demodulation
* high-level data link control (HDLC) framing (0 bit insertion/removal, cyclic
redundancy check (CRC) generation/checking)
* V.27 ter (2400/4800 bps) high-speed data modulation and demodulation
* V.29 (7200/9600 bps) high-speed data modulation and demodulation
* V.17 (7200/9600/12000/14400 bps) high-speed data modulation and demodulation
* V.33 (12000/14400 bps) high-speed modulation and demodulation
* CED detection and generation
* CNG detection and generation
* V.21 Channel-2 detection

Figure 10. FoIP Module

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fax protocol unit (FP)This compensates for the effects of timing and lost
packets caused by the packet network. The FP prevents the local fax machine
from timing out while waiting for a response from the other end by generating HDLC flags. If, after a time out, the response from the remote fax
machine is not received, it also sends a command repeat (CRP) frame to
resend the frame. This unit monitors the facsimile transaction timing, the
direction of current transmission, and the proper modem configuration. It performs the following functions:
* protocol processing (group-3 facsimile)
* examination/alteration of binary signaling messages to ensure compatibility
of the facsimile transfer with the constraints of the transmission channel
* network channel interface data formatting
* line state transitions
fax network driver unit (FND)This assembles and disassembles fax
packets to be transmitted over the network and is the interface unit between
the FP and network modules. The control information packets consist of
header and time stamp information. In the direction of the PCM to the packet
network, the FND collects the specified number of bytes and transmits the
packet to the network. In the receive direction, the FND provides data with
the proper timing (as generated on the transmit side and reproduced through
the received time stamp information) to the rest of the FIU. The FND formats
the network packets for transmission to the network based on the specific
network protocol. The FND delays the data to remove timing jitter from the
packet arrival times and performs the following functions:
* formatting of control information
* formatting of fax data
* properly timed playout of data
* elastic (slip) buffering
* lost-packet compensation

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The Basics of Telecommunications

11. FOIP SUMMARY


A fax-over-packet software architecture has been described for the interworking of fax machines and packet networks. Some of the key features enabling
these applications to function successfully are as follows:
* an approach that addresses the effect of delay through the network
* a process that minimizes the effect of jitter
* features that address lost-packet compensation
Though the QoS issues associated with carrying FoIP networks are significant, the future of this approach will be driven by the substantial cost savings
and exciting applications made possible with FoIP software technology.

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12. SELF-TEST
1. The consolidation of separate voice and data networks offers an opportunity for
_____________.
a. utilization of extra broadband bandwidth for voice and data transmission
b. reduced delay over a telephone call
c. reduced computer and telephone hardware requirements

2. Voice-over-packet technology may be used to transfer information over both broadband and wireless networks.
a. true
b. false

3. A QoS issue unique to packet networks is _______________.


a. interworking
b. compression
c. jitter

4. Signal reflections generated by the circuit that converts between a four-wire and a
two-wire circuit can result in ___________________.
a. jitter
b. echo
c. delay

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The Basics of Telecommunications


5. Developers seeking to incorporate voice-over-packet technology face which of the following challenges?
a. still-evolving technical standards
b. network phenomena such as delay, jitter, echo, and lost packets
c. integration of incompatible technologies
d. all of the above

6. The fax-over-packet IWF reduces overall bandwidth because the fax transmission is
_______________.
a. half-duplex
b. real time
c. low-speed data
d. only b and c
e. all of the above

7. A major cause of problems in the transmission of fax-over-packet networks is


_____________.
a. properly dividing the four phases of the call
b. incompatible fax machines
c. using T.30 and T.4 protocols
d. the timing related to the message interaction
e. all of the above

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8. Fax-over-packet software addresses the problems of lost frames by
_______________.
a. using an error-correcting protocol, such as TCP, to transport the
fax data at the expense of added delay
b. replaying last packets and other methods of interpolation
c. repeating information in subsequent frames so that the error can
be corrected by the receivers playout mechanism.
d. only a and c
e. all of the above

9. Processing delay is caused by the process of modulating and correcting the digital
fax information into a packet for transmission over the packet network.
a. true
b. false

10. The fax network driver delays the data and performs which of the following?
a. lost-packet compensation
b. formatting of fax data
c. removal of timing jitter from fax arrival times
d. only a and b
e. all of the above

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The Basics of Telecommunications

13. ACRONYM GUIDE


ADPCM

adaptive pulse-code modulation

ANSI

American National Standards Institute

ATM

asynchronous transfer mode

BRI

basic rate interface

CED

CallED station IDentification

CELP

code excited linear prediction

CFR

confirmation to receive

CIR

committed information range

CNG

calling tone

CRC

cyclic redundancy check

CRP

command repeat

CSI

called subscriber information

DCN

disconnect message

DCS

digital command signal

DIS

digital identification signal

DLCI

data-link connection identifier

DSP

digital signal processor

DTMF

dual-tone multifrequency

E&M

ear & mouth, earth, and magneto

EOP

end of procedures

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FIFO

first in, first out

FIU

facsimile interface unit

FM

fax modem unit

FND

fax network driver unit

FoIP

fax over Internet protocol

FP

fax protocol unit

FXO

foreign exchange office

FXS

foreign exchange station

HDLC

high-level data link control

IETF

Internet Engineering Task Force

IP

Internet protocol

ISDN

integrated services digital network

ITU

International Telecommunications Union

IWF

interworking funtion

LMI

local management interface

MCF

message confirmation

MIB

management information base

MPS

multipage signal

NSF

nonstandard facilities

PBX

private branch exchange

PCM

pulse code modulation

415

The Basics of Telecommunications


POTS

plain old telephone service

PRI

primary rate interface

PSTN

public switched telephone network

PVC

permanent virtual circuit

QoS

quality of service

RTCP

real-time conferencing protocol

RTP

real-time protocol

SVC

switched virtual circuit

TCF

training check field

TCP

transmission control protocol

UDP

user datagram protocol

UNI

user network interface

V/FoIP

voice and fax over Internet protocol

VoFR

voice over frame relay

VoIP

voice over Internet protocol

VToA

voice telephony over ATM

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Self-Test Correct Answers


Asymmetric Digital Subscriber Line (ADSL)
1. d

6. b

2. c

7. c

3. c

8. a

4. a

9. a

5. c

10. b

Asynchronous Transfer Mode (ATM) Fundamentals


1. b

6. b

2. a

7. a

3. a

8. b

4. a

9. a

5. b

10. a

Cable Modems
1. a

6. b

2. b

7. a

3. c

8. a

4. b

9. b

5. b

10. b

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The Basics of Telecommunications

Cellular Communications
1. c

6. a

2. b

7. b

3. a

8. b

4. c

9. b

5. d

10. a

Fiber-Optic Technology
1. c

6. c

2. b

7. b

3. c

8. d

4. b

9. a

5. a

10. b

Fundamentals of Telecommunications
1. b

6. c

2. a

7. e

3. b

8. a

4. b

9. c

5. a

10. a

Intelligent Networks (INs)


1. c

7. a

2. a

8. b

3. c

9. a

4. b

10. a

5. d

11. a

6. a

12. b

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Self-Test Correct Answers

Internet Access
1. c

11. d

2. b

12. b

3. d

13. c

4. a

14. b

5. b

15. b

6. a

16. a

7. a

17. a

8. b

18. a

9. b

19. b

10. a

20. a

Internet Telephony
1. b

6. b

2. a

7. a

3. b

8. c

4. b

9. c

5. a

10. c

Intranets and Virtual Private Networks (VPNs)


1. c

8. b

2. a

9. c

3. d

10. d

4. c

11. a

5. a

12. a

6. b

13. c

7. b
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The Basics of Telecommunications

Operations Support Systems (OSSs)


1. c

7. a

2. b

8. a

3. d

9. b

4. b

10. a

5. c

11. b

6. a

Optical Networks
1. a

6. a

2. c

7. a

3. b

8. b

4. a

9. b

5. a

10. a

Personal Communications Services (PCS)


1. b

6. c

2. d

7. a

3. d

8. b

4. a

9. b

5. a

10. a

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Self-Test Correct Answers

Signaling System 7 (SS7)


1. b

11. a

2. b

12. b

3. b

13. a

4. b

14. b

5. a

15. b

6. b

16. a

7. c

17. d

8. b

18. d

9. b

19. d

10. a

20. a

Synchronous Optical Networks (SONETs)


1. c

6. b

2. b

7. a

3. c

8. a

4. b

9. c

5. a

10. a

Voice and Fax over Internet Protocol (V/FoIP)


1. a

6. a

2. a

7. d

3. c

8. d

4. b

9. b

5. d

10. e

421

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