                                .oO Phrack 49 Oo.

                          Volume Seven, Issue Forty-Nine
                                     
                                  File 05 of 16


                         Introduction to Telephony and PBX
				   by Cavalier[TNO]

                                Table of Contents


       1. . . . . . . . . . . . . . . . . . . . The Central Office
       2. . . . . . . . . . . . . . .Private Branch Exchange (PBX)
       3. . . . . . . . . Properties of Analog and Digital Signals
       4. . . . . . . . . . . . . . . . .Analog-Digital Conversion
       5. . . . . . . . . . . . . . . . . . . Digital Transmission
       6. . . . . . . . . . . . . . . . . . . . . . . Multiplexing
       7. . . . . . . . . . . . . . . . . . . . Transmission Media
       8. . . . . . . . . . . . . . . . . . . . . . . . .Signaling


	.--------------------.
1	| The Central Office |
	`--------------------'

Telephones alone do nothing special.  Their connection to the rest of
world makes them one of mankind's greatest achievements.

In the early days of telephone communications, users had to establish
their own connections to other telephones.  They literally had to string
their own telephone lines.

Although the customer inconvenience of building their own connections
limited the availability of phone service, an even greater problem soon
arose.  As the telephone became more popular, more people wanted to be
connected.   At the time, each phone had to be directly wired to each
other.  In a very short time there was a disorganized maze of wires
running from the homes and businesses.

A simple mathematical formula demonstrates the growth in the number of
connections required in a directly wired network:

                              I = N(N-1)/2
       (I = number of interconnections; N = number of subscribers)

                            I = 100(100-1)/2

If just 100 subscribers attempted to connect to each other, 4950
separate wire connections would be needed!  Obviously, a better method
was needed.


Switching


A Central Office (CO) switch is a device that interconnects user
circuits in a local area, such as a town.  The CO is a building where
all subscriber phone lines are brought together and provided with a
means of interconnection.  If someone wants to call a neighbor, the call
is routed through the CO and switched to the neighbor.

What if someone wanted to call a friend in the next town?  If their
friend was connected to a different CO, there was no way to communicate.

The solution was to interconnect COs.  Then, CO-A routed calls to CO-B
to complete the connection.

Today every CO in the world is connected to every other CO in a vast
communication highway known as the Public Switched Network (PSN).  The
PSN goes by a variety of different names:

                 Dial-up network
                 Switched network
                 Exchange network

The CO provides all users (subscribers) with a connection to each other.
A critical note, however, is that no CO has the resources to switch all
their users simultaneously.  It would be too expensive and it is
unnecessary to attempt to do so because for the vast majority of the
time, only a small percentage of subscribers are on the phone at the
same time.

If, on a rare occasion, all the circuits are busy, the next call will be
blocked. A call is blocked if there are no circuits available to switch
it because all the circuits are in use.

The term `probability of blocking` is a statistical logarithm which
determines the chance that a call cannot be switched.  For modern day
commercial COs, the probability of blocking is very low.


History of COs


Operating switching

In the first COs, a subscriber who wanted to place a call cranked a
magneto-generator to request service from the local phone company.  An
operator at the CO monitored subscriber connections by observing lamps
on a switchboard console.  When a subscriber's lamp lit, indicating the
request for service, the operator would answer: "Number please...".

The operator connected one call to another by plugging one end of a cord
into the jack of the caller  and the other end of the cord into the jack
of the called party, establishing a manual, physical connection.

The switchboard had to have a jack for every incoming and outgoing line
that needed service.  The number of lines an operator could monitor was
limited by her arm's reach.   Billing was accomplished by the operators
writing up a ticket for each call designating its starting and ending
times.

When telephone subscribers were few in number, this method worked fine.
As the popularity of the phone increased, more phones placed more calls
and it became increasingly unmanageable and expensive to manually switch
and bill each call.

Strowger Step-by-Step Switch

A mechanical switch was invented in the 1890's by a Kansas City
mortician named Almon B. Strowger.  He became very suspicious because
callers looking for a mortician were continually referred to his
competition instead to him. When he learned that the local operator was
the wife of his rival, his suspicions were confirmed.  He set about to
invent a switching system that would not be dependent upon human
intervention.

His creation, called the Strowger or Step-by-Step switch, was the first
automated electromechanical switching system.  It placed switching
control in the hands of the subscriber instead of the operator by adding
a dialing mechanism to the phone.

The Strowger switch completed a call by progressing digit by digit
through two axes of a switching matrix in the CO.  A call was stepped
vertically to one of ten levels and rotated horizontally to one of ten
terminals.

It was called step-by-step because calls progress one step at a time as
the customer dialed each digit of the number.  When the final digit was
dialed, the switch seized an available circuit and connected the call.

The result of the step-by step switch was to eliminate the need for
manual operator connection and grant privacy and call control to the
subscriber.

The step-by-step switch was a wonderful invention for its day.   Today
it is obsolete.  Compared to modern day switches, it is slow, noisy
and too expensive to maintain.  It is also both bulky and inefficient.

The Crossbar Switch

The crossbar switch was invented and developed in the late 1920s.  One
of its main technological advanced was the introduction of a hard wired
memory to store dialed digits until the dialing was complete.

Unlike the step-by-step method, calls are not processed under the
direct control of incoming dial pulses.  In the step-by-step method,
each phone call controlled its own pathway through the switching matrix
at the speed the digits were dialed by the user.  The crossbar switch
introduced a better method.

Devices called registers stored the digits in memory as they were dialed
by the callers.  Not until all the digits were dialed would the call
begin to be switched.  Once all the digits were received and stored in
the register, the register handed the digits to a processor to be
examined and used to route the call.

When a pathway had been established and the call was connected, the
register and processor would release and become available to handle
another call. Collectively, this process was called `common control`.

Common control resulted in faster call completion and increased capacity
of the switch.  With the old step-by-step, the time it would take a user
to physically dial the digits would occupy valuable switch time because
dialing the digits was the most time consuming part of switching a call.
This 8 to 12 seconds of dialing time prevented other users from
accessing the switching matrix and generally slowed things down.

The genius of the crossbar common control was to store the dialed digits
as they came in and then after the user finished dialing, send the
digits off for processing.  The act of dialing no longer kept other
calls waiting for switch resources.

Common control created the separation of the control functions (setting
up and directing the call) from the switching functions (physically
creating the connections).

Crossbar Switching Matrix

Calls were connected by sharing a dedicated wire path through the
switching matrix.  Crossbar switches used the intersection of two points
to make a connection.  They selected from a horizontal and vertical
matrix of wires, one row connected to one column.  The system still
stepped the call through the network, but only after all the digits were
dialed.  This method created a more efficient allocation of switch
resources.

There are four important components of a crossbar switch.

   .  The marker is the brain of a crossbar switch.  It identifies a
      line requesting service and allocates a register.

   .  The register provides dial tone and receives and stores the dialed
      digits.

   .  The matrix is a set of horizontal and vertical bars.  The point at
      which the crosspoints meet establishes the connection.

   .  A trunk interface unit, also called a sender, processes calls from
      a PBX.

Although crossbar is faster and less bulky than step-by-step, it is
still electromechanical and requires a lot of maintenance.  It requires
huge amounts of space, generates a lot of heat, and makes a great deal of
noise.

Electronic Switching System  (ESS)

The advent of electronic switching (also called stored program
switching) was made possible by the transistor.  Introduced in 1965, the
Electronic Switching System (ESS) greatly sped up switch processing
capacity and speed and has done nothing less than revolutionize the
industry.

Modern ESS switches perform five main functions to establish and
maintain service in a public network.

   1. Establish a connection between two or more points
   2. Provide maintenance and testing services
   3. Record and sort customer billing charges
   4. Offer customer features, such as call waiting
   5. Allow access to operators for special services

An ESS uses computer-based logic to control the same two primary
operations we introduced with the crossbar -- common control and the
switching matrix.

(In an ESS, the terms stored program control, common control, and
electronic switching are all synonymous.)

ESS Common Control

The function of the common control is similar to its function in the
crossbar. The difference is that common control is accomplished
electronically instead of electromechanically.  Like the crossbar, one
group of control devices controls the functions of all lines.  However,
instead of the hard wired logic of the crossbar, the control device
consists of a computer with memory, storage, and programming capability.

In the ESS, the computer governs the common control.  It monitors all
the lines and trunks coming into the CO, searching for changes in the
electrical state of the circuit, such as a phone going off-hook.  When a
subscriber goes off- hook and dials a number, the common control
equipment detects the request for service and responds by returning the
dial tone.  It then receives, stores, and interprets the dialed digits.

Again, similar to the workings of the crossbar, once the digits have
been processed, the computer establishes a path through the switching
matrix to complete the call.  After the connection for the call has been
established, the common control equipment releases and becomes available
to complete other calls.

ESS Switching Matrix

Recall that in the crossbar, calls were connected by sharing a dedicated
wire path through the matrix, establishing a connection between an input
and an output.  The matrix in an ESS is logically similar to the
crossbar grid except the pathway is electronic instead of
electromechanical.  Called a TDM bus, it is solid state circuitry and is
printed into small computer controlled circuit boards.  The computer
controls the connections and path status map to determine which path
should be established to connect the calling and called parties.

Remember

      Crossbar switching matrix  =  maze of physical wire cross connections

      ESS switching matrix  =  electronic multiplexed TDM (time division
      multiplexing) bus

ESS Advancements

The unprecedented advancement of the ESS was the speed and processing
power advantage it had over the crossbar because it switched calls
digitally instead of electromechanically.  The processing capacity that
would have required a city block of crossbar technology could be
accomplished by one floor of ESS equipment.  Much less effort was
required to maintain the ESS because it was smaller and had fewer moving
parts.

Telephone companies would have moved to the new technology for these
advantages alone.  But, there was much more to be offered.  There was
the power of the computer.

There are major advantages to a computer stored program.  It allows the
system to perform functions earlier switches were incapable of.  For
example, the switch can collect statistical information to determine its
effectiveness.  It can perform self-diagnostics of circuit and system
irregularities and report malfunctions.  If trouble occurs, technicians
can address it via a keyboard and terminal.  The same terminal, often
called a system managers terminal, allows personnel to perform system
changes and to load new software, eliminating the need for manually
rewiring connections.

The computer uses two types of memory:

   .  Read Only Memory (ROM) is used to store basic operating
      instructions and cannot be altered by the end user.  The contents
      of this memory can only be changed by the manufacturer.

   .  Random Access Memory (RAM) stores configuration and database
      information.  The contents of its memory can be changed by a
      system administrator.

Other important functions of the computer include

   .  Performing telephone billing functions
   .  Generating traffic analysis reports
   .  Generating all tones and announcements regarding the status of
      circuits and calls

Computer control operates under the direction of software called its
generic program.  Periodically updating or adding to the generic program
allows the ESS to be much more flexible and manageable than previous
switch generations because it is the software, not the hardware, that
normally has to be upgraded.

Electronic switching heralded the introduction of new customer features
and services.  Credit card calls, last number redial, station transfer,
conference calling, and automatic number identification (ANI) are just 
a few examples of unprecedented customer offerings.

The ESS is an almost fail-safe machine.  Its design objective is one
hour's outage in 20 years.  In today's competitive environment for
higher quality communication equipment, ESS machines provide a level of
service and reliability unachievable in the past.



	.-----------------------------------.
2	| The Private Branch Exchange (PBX) |
	`-----------------------------------'

The two primary goals of every PBX are to

         . facilitate communication in a business
         . be cost effective


Organizations that have more than a few phones usually have an internal
switching mechanism that connects the internal phones to each other and
to the outside world.

A PBX is like a miniature Central Office switching system designed for a
private institution.  A PBX performs many of the same functions as a CO
does. In fact, some larger institutions use genuine COs as their private
PBX.

Although a PBX and a CO are closely related, there are differences
between them

   .  A PBX is intended for private operation within a company.  A CO is
      intended for public service.

   .  A PBX usually has a console station that greets outside callers
      and connects them to internal extensions.

   .  Most PBXs do not maintain the high level of service protection
      that must be maintained in a CO.  Assurance features such as
      processor redundancy (in the event of processor failure) and
      battery backup power, which are standard in a CO, may not be a
      part of a PBX.

   .  COs require a seven digit local telephone number, while PBXs can
      be more flexible and create dialing plans to best serve their
      users (3, 4 5, or 6 digit extensions).

   .  A PBX can restrict individual stations or groups of stations from
      certain features and services, such as access to outside lines.  A
      CO usually has no interest in restricting because these features
      and services are billed to the customer.  COs normally provide
      unlimited access to every member on the network.

A PBX is composed of three major elements.

      1.  Common equipment (a processor and a switching matrix)
      2.  CO trunks
      3.  Station lines


Common Equipment

The operation of a PBX parallels the operation of a Central Office ESS.
Its common control is

   .  A computer operated Central Processing Unit (CPU) running software
      that intelligently determines what must be done and how best to do
      it.

   .  A digital multiplexed switching matrix printed on circuit boards
      that establishes an interconnection between the calling and called
      parties.

The CPU stores operating instructions and a database of information from
which it can make decisions.  It constantly monitors all lines for
supervisory and control signals.  A switching matrix sets up the
connections between stations or between stations and outgoing trunks.

Housed in equipment cabinets, PBX common equipment is often compact
enough to occupy just a closet or small room.  Given the extremely high
rental rates many companies have, a major benefit of a PBX is its small
size.

CO Trunks and Station Lines

A trunk is a communication pathway between switches.  A trunk may
provide a pathway between a PBX and the CO or between two PBXs and two
COs.  A trunk may be privately owned or be a leased set of lines that
run through the Public Switched Network.

A line is a communication pathway between a switch and terminal
equipment, such as between a PBX and an internal telephone or between a
CO and a home telephone.

The function of the PBX is to interconnect or switch outgoing trunks
with internal lines.


Two Varieties of Lines

Station lines are either analog or digital, depending on the station
equipment it is connecting.  If the phone on one desk is digital, it
should be connected to a digital line.  If the phone on the desk is
analog, it should be connected to an analog line.


Varieties of Trunks

There exists a wide variety of trunks that can be connected to a PBX for
off-premises communication.  Each variety has different functions and
capabilities.  It is important to be able to distinguish them.

Tie Trunks

Organizations supporting a network of geographically dispersed PBXs
often use tie trunks to interconnect them.  A tie trunk is a permanent
circuit between two PBXs in a private network.  Tie trunks are usually
leased from the common carrier; however, a private microwave arrangement
can be established. Usually, leased tie trunks are not charged on a per 
call basis but rather on the length of the trunk.  If a tie trunk is
used more than one or two hours a day, distance sensitive pricing is
more economical.

A T1 trunk is a digital CO leased trunk that is capable of being
multiplexed into 24 voice or data channels at a total rate of 1.544
Mbps.  T1 trunks are used as PBX-to-PBX tie trunks, PBX-to-CO trunks as
well as PBX trunks to bypass the local CO and connect directly to a long
distance carrier.  It is a standard for digital transmission in North
America and Japan.

T1 uses two pairs of normal, twisted wire--the same as would be found in
a subscriber's residence.  Pulse Code Modulation is the preferred method
of analog to digital conversion.

A T2 trunk is capable of 96 multiplexed channels at a total rate of
6.312 Mbps.

A T3 trunk is capable of 672 multiplexed channels at a total rate of
44.736 Mbps.

A T4 trunk is capable of 4,032 multiplexed channels at a total of
274.176 Mbps.


Direct Inward Dialing (DID) Trunks

Incoming calls to a PBX often first flow through an attendant position.
DID trunks allow users to receive calls directly from the outside
without intervention from the attendant.  DID offers three main
advantages.

      1. It allows direct access to stations from outside the PBX.
      2. It allows users to receive calls even when the attendant
      switchboard is closed.
      3. It takes a portion of the load off the attendants.

Trunk Pools

Trunks do not terminate at a user's telephone station.  Instead trunks
are bundled into groups of similarly configured trunks called trunk
pools.  When a user wants to access a trunk, he can dial a trunk access
code--for example, he can dial 9 to obtain a trunk in the pool.  Trunk
pools make system administration less complicated because it is easier
to administer a small number of groups than a large number of individual
trunks.


Ports

Ports are the physical and electrical interface between the PBX and a
trunk or station line.


PBX Telephones

Telephone stations in a PBX are not directly connected to the CO but to
the PBX instead.  When a station goes off-hook, the PBX recognizes it
and sends to the station its own dial tone.  The PBX requires some
access digit, usually "9" to obtain an idle CO trunk from a pool to
connect the station with the public network.   This connection between
the telephone and the PBX allows stations to take advantage of a myriad
of PBX features.

The attendant console is a special PBX telephone designed to serve
several functions.  Traditionally, most PBXs have used attendants as the
central answering point for incoming calls.  Calls placed to the PBX
first connected to the attendant, who answered the company name.  The
attendant  then established a connection to the desired party.  The
attendant also provided assistance to PBX users, including directory
assistance and reports of problems.

In recent years a number of cost-saving improvements have been made to
the attendant console.  A feature commonly called automated attendant
can establish connections without a human interface, substantially
decreasing PBX operating costs.

Blocking versus Non-blocking

Blocking is a critical aspect of the functioning of a PBX.  A
non-blocking switch is one that provides as many input/output interface
ports as there are lines in the network.   In other words, the switching
matrix provides enough paths for all line and trunk ports to be
connected simultaneously.

PBX systems are usually blocking.  It requires an exponential increase
in resources and expense to ensure non-blocking.   Based on call traffic
studies and the nature of calls, it is generally acceptable to engineer
a low level of blocking in exchange for a major savings of common
equipment resources.

Grades of service are quantitative measurements of blocking.  They are
written in the form:

                 P.xx

where xx is a two digit number that indicates how many calls out of a
hundred will be blocked.  The smaller the number, the better the grade
of service.

P.01 means one call out of a hundred will be blocked.  It is a better
grade of service than P.05 that block five calls out of a hundred.
Naturally the P.05 service costs less than the better grade of service
provided by P.01.

Even if a PBX's switching matrix is non-blocking, an internal caller may
still not be able to reach an outside trunk if all the trunks are busy.
CO trunks cost money, and very few PBXs dedicate one trunk to every
internal line. Instead, traffic studies are performed to determine the
percentage of time a station will be connected to an outside trunk
during peak hours.

If, for example, it is determined that the average station uses a trunk
only 20% of the time during peak hours, then the switch may be
configured to have a 5:1 line-to-trunk ratio, meaning for every five
lines (or extensions) there is one trunk.  Most PBXs are configured on
this principle as a major cost saving method.


PBX Features

COs and PBXs share many of the same attributes and functionality.
However, COs are built to perform different tasks than a PBX, resulting
in feature differences between them.  The following is an overview of
common PBX features not found in a CO.

Automatic Route Selection (ARS)

A primary concern of any telecommunications manager is to keep costs
down. One of these costs is long distance service.  ARS is a feature
that controls long distance costs.

Most PBXs have more than just public CO trunks connected to them.  They
may have a combination of tie trunks to other PBXs (T1/E1 trunks and
many others). Each type of trunk has a separate billing scheme,
relatively more or less expensive for a given number of variables.

It is extremely difficult to attempt to educate company employees on
which trunks to select for which calls at what time of day.  It defeats
the productivity-raising, user-transparency goal of any PBX if employees
must pour over tariffing charts every time they want to use the phone.

Instead, ARS programs the PBX central processor to select the least
expensive trunk on a call by call basis.  When a user places a call, the
computer determines the most cost effective route, dials the digits and
completes the call.


Feature Access

PBXs support a wide variety of user features.  For example, call
forward, hold, and call pickup are all user features.  There are two
methods of activating a feature.  A code, such as "*62" can be assigned
to the call forward feature. To activate call forward the user presses
"*62" and continues dialing.

Dial codes are not the preferred method of feature access.   The problem
is that users tend to forget the codes and either waste time looking
them up or do not take advantage of time saving features, thereby
defeating the purpose of buying them.

Dedicated button feature access is a better solution.   Programmable
feature buttons, located on most PBX telephones, are pressed to activate
the desired feature.  If a user wants to activate call forward, he
presses a button labeled "call forward" and continues dialing.

The only drawback of telephones with programmable feature buttons is
that they are more expensive than standard phones.


Voice Mail

For a voice conversation to occur, there is one prerequisite so obvious
it is usually overlooked.  The called party must be available to answer
the call.  In today's busy world, people are often not accessible which
can create a major problem resulting in messages not being received and
business not being conducted.

Statistics confirm the need for an alternate method.

      75% of call attempts fail to make contact with the desired party.

      50% of business calls involve one-way information--one party
      wishing to deliver information to another party without any
      response necessary.

      50% of incoming calls are less important than the activity they
      interrupt.

Voice mail (also known as store and forward technology) is a valuable
feature that is designed around today's busy, mobile office.  It is like
a centralized answering machine for all telephone stations in a PBX.
When a telephone is busy or unattended, the systems routes the caller to
a voice announcement that explains that the called party is unavailable
and invites the caller to leave a message.  The message is stored until
the station user enters a security dial access code and retrieves the
message.


Automated Attendant

Automated attendant is a feature sometimes included with voice mail.  It
allows outside callers to bypass a human attendant by routing their own
calls through the PBX.  Callers are greeted with a recorded announcement
that prompts them to dial the extension number of the desired position,
or stay on the line to be connected to an attendant.

Reducing cost is the primary goal of automated attendant.  The decreased
attendant work load more d) an pays for the cost of the software and
equipment.

When automated attendant was first introduced, it met with substantial
resistance from the general public.  People did not want to talk to a
machine. But, as its cost effectiveness drove many companies to employ
it, the public has slowly adjusted to the new technology.

Restriction

Nearly every PBX enforces some combination of inside and outside calling
restrictions on certain phones.  Depending upon the sophistication of
the PBX, a system administrator can have nearly unlimited flexibility in
assigning restrictions.   For example, a tire manufacturing plant could
restrict all lobby phones at corporate headquarters to internal and
local calls only.  The phones at the storage warehouse could be
restricted for only internal calling.  But, all executive phones could
be left unrestricted.

Long distance toll charges can be a crippling expense.  Toll fraud is a
major corporate problem.  Restriction combats unauthorized use of
company telephone resources and is a prime function of any PBX.


Tandems

As stated earlier, it is necessary to have a switching mechanism to
interconnect calls.  If a number of phones all wish to be able to talk
to each other, an enormous amount of cabling would be wasted tying each
of them together.  Thus, the switch was born.

The same principle applies for interconnecting PBXs.  Large firms that
have PBXs scattered all over the country want each PBX to have the
ability to access every other one.  But the expense of directly
connecting each could drive a company out of business.  The solution is
to create a centrally located tandem switching station to interconnect
the phones from one PBX with the phones from any other.  This solution
creates a Private Switched Network.

Directing digits are often used to inform the tandem switch where to
route the call.  Each PBX is assigned a unique number.  Let's say a PBX
in Paris is numbered "4."  To call the Paris PBX from a PBX in Chicago,
a user would dial "4- XXXX."


Uniform Dialing Plan

A network of PBXs can be configured poorly so that calling an extension
at another PBX could involve dialing a long, confusing series of numbers
and create a lot of user frustration.  A Uniform Dialing Plan enables a
caller to dial another internal extension at any PBX on the network with
a minimum of digits, perhaps four or five.  The system determines where
to route the call, translates the digits and chooses the best facility,
all without the knowledge of the user.  As far as the user knows, the
call could have been placed to a station at the next desk.


Call Accounting System (CAS) and Station Message Detail Recording (SMDR)

CAS works in conjunction with SMDR to identify and monitor telephone
usage in the system.  SMDR records call information such as the calling
number, the time of the call, and its duration.  The raw data is usually
listed chronologically and can be printed on reports.

SMDR by itself is not particularly useful because the sheer volume and
lack of sorting capability of the reports make them difficult to work
with.  A Call Accounting Systems is a database program that addresses
these shortcomings by producing clear, concise management reports
detailing phone usage.

The primary function of CAS reports is to help control and discourage
unnecessary or unauthorized use and to bill back calling charges to
users. Many law firms use a call accounting system to bill individual
clients for every call they make on behalf of each client.


Attendant Features

A number of features are available to improve the efficiency of
attendant consoles.

Here are a few of them.

      Direct Station Selection (DSS) allows attendants to call any
      station telephone by pressing a button labeled with its extension.

      Automatic Timed Reminder alerts the attendant that a station has
      not picked up its call.  The attendant may choose to reconnect to
      the call and attempt to reroute it.

      Centralized Attendant Service groups all network attendants into
      the same physical location to avoid redundancies of service and
      locations.


Power Failure Schemes

If a city or a town experiences a commercial power failure, telephones
connected directly to the CO will not be affected because the CO gets
power from its own internal battery source.  A PBX, however, is
susceptible to general power failures because it usually gets its power
from the municipal electric company.

There are several different ways a PBX can be configured to overcome a
power failure.

      A PBX can be directly connected to a DC battery which serves as
      its source of power.  The battery is continually recharged by an
      AC line to the electric company.  In the event of a power failure,
      the PBX will continue functioning until the battery runs out.

      A PBX can have an Uninterruptable Power Supply (UPS) to protect
      against temporary surges or losses of power.

      A PBX can use a Power Failure Transfer (PFT) which, in the event
      of a power failure, immediately connects preassigned analog phones
      to CO trunks, thereby using power from the CO instead of from the
      PBX.


Outgoing Trunk Queuing

In the event all outgoing trunks are busy, this feature allows a user to
dial a Trunk Queuing code and hang up.  As soon as a trunk becomes free,
the system reserves it for the user, rings the station and connects the
outside call automatically.


System Management

PBXs can be so large and complex that without a carefully designed
method of system management chaos can result.  The best, most advanced
systems mimic CO management features--computer access terminals which
clearly and logically program and control most system features.  The
system manager has a wide variety of responsibilities which may include,
but is not limited to

      Programming telephone moves, additions, and changes on the system

      Performing traffic analysis to maximize system configuration
      resources and optimize network performance

      Responding to system-generated alarms

      Programming telephone, system, attendant, and network features.


ISDN


ISDN is not a product.  Rather, it is a series of standards created by
the international body, ITU (previously known as CCITT), to support the
implementation of digital transmission of voice, data, and image through
standard interfaces.  Its goal is to combine all communications services
offered over separate networks into a single, standard network.  Any
subscriber could gain access to this vast network by simply plugging
into the wall.  (At this time not all PBXs are compatible with the ISDN
standard.)


Alternatives to a PBX

There are two main alternatives to purchasing a PBX.  They are
purchasing a Key system or renting Centrex service from the local
telephone company.


Key System

Key systems are designed for very small customers, who typically use
under 15 lines.  There is no switching mechanism as in a PBX.  Instead
every line terminates on every phone. Hence, everyone with a phone can
pick up every incoming call.

Key systems are characterized by a fat cable at the back of each phone.
The cables are fat because each phone is directly connected to each
incoming line and each line has to be wired separately to each phone.

Fat cables have become a drawback to Key systems as building wire
conduits have begun to fill with wire.  It has become increasingly
difficult to add and move stations because technicians must physically
rewire the bulky cables instead of simply programming a change in the
software.

Key telephones are equipped with line assignment buttons that light on
incoming calls and flash on held calls.  These buttons enable a user to
access each line associated with each button.  Unlike a PBX, there is no
need to interface with an attendant console to obtain an outside line.


Differences between Key and PBX Systems

      Key systems have no switching matrix.  In a Key system, incoming
      calls terminate directly on a station user's phone.  In a PBX,
      incoming calls usually first go to the attendant who switches the
      call to the appropriate station.

      PBX accesses CO trunk pools by dialing an access code such as "9."
      Key systems CO trunks are not pooled.  They are accessed directly.

Key systems make use of a limited number of features, many of them
common to the PBX.  These include

           Last number redial
           Speed dialing
           Message waiting lamp
           Paging
           Toll restriction

Today's PBXs can simulate Key system operation.  For example, telephones
can have a line directly terminating on a button for direct access.


Centrex

The other alternative to purchasing a PBX is leasing a Centrex service.

Centrex is a group of PBX-like service offerings furnished by the local
telephone company.  It offers many of the same features and functions
associated with a PBX, but without the expense of owning and maintaining
equipment and supporting in-house administrative personnel.

Because network control remains the responsibility  of the CO, companies
that choose Centrex service over purchasing and maintaining a private
PBX can ignore the sophisticated world of high tech telecommunications
and leave it up to the telephone company representatives.

To provide Centrex service, a pair of wires is extended from the CO to
each user's phone.  Centrex provides an "extension" at each station
complete with its own telephone number.  No switching equipment is
located at the customer premises.  Instead, Centrex equipment is
physically located at the CO.

There are a number of reasons a company would choose a Centrex system
over owning their own PBX.  Currently Centrex has six million customers
in the United States market.

Advantages of a Centrex System over a PBX:

      Nearly uninterruptable service due to large redundancies in the CO

      Easily upgraded to advanced features.

      No floor space requirement for equipment.

      No capital investment

      24-hour maintenance coverage by CO technicians

      Inherent Direct Inward Dialing (DID).  All lines terminate at
      extensions, instead of first flowing through a switchboard.

      Call accounting and user billing as inherent part of the service.

      Reduced administrative payroll.


Disadvantages of a Centrex System:

      Cost.  Centrex is tariffed by the local telephone company and can
      be very expensive.  Companies are charged for each line connected
      to the Centrex, as well for the particular service plan chosen.
      Additionally, Centrex service may be subject to monthly increases.

      Feature availability.  Centrex feature options are generally not
      state of the art, lagging behind PBX technology.  Not all COs are
      of the same generation and level of sophistication--a company
      associated with an older CO may be subject to inferior service and
      limited or outdated feature options.

      Control of the network is the responsibility of the CO.  While
      this release from responsibility is often cited as a positive
      feature of Centrex, there are drawback to relinquishing control.
      CO bureaucracy can be such that a station move, addition or change
      can sometimes take days to achieve.   Furthermore, each request is
      charged a fee.  Also, some companies are more particular about
      certain features of their network (security for example) and
      require direct control for themselves.



	.------------------------------------------.
3	| Properties of Analog and Digital Signals |
	`------------------------------------------'

A man in Canada picks up a telephone and dials a number.  Within
seconds, he begins talking to his business partner in Madrid.  How can
this be?

Telephony is a constantly evolving technology with scientific rules and
standards.  You will learn to make sense of what would otherwise seem
impossible.

Voice travels at 250 meters per second and has a range limited to the
strength of the speaker's lungs.  In contrast, electricity travels at
speeds approaching the speed of light (310,000 Km per second) and can be
recharged to travel lengths spanning the globe.  Obviously, electricity
is a more effective method of transmission.

To capitalize on the transmission properties of electricity, voice is
first converted into electrical impulses and then transmitted.  These
electrical impulses represent the varying characteristics that
distinguish all of our voices.  The impulses are transmitted at high
speeds and then decoded at the receiving end into a recognizable
duplication of the original voice.

For a hundred years, scientists have been challenged by how best to
represent voice by electrical impulses.  An enormous amount of effort
has been devoted to solving this puzzle.  The two forms of electrical
signals used to represent voice are analog and digital.

Both analog and digital signals are composed of waveforms.  However,
their waveforms have very distinctive properties which distinguish them.
To understand the science of telephony, it is necessary to understand
how analog and digital signals function, and what the differences
between them are.

If you do not possess a fundamental understanding of basic waveforms,
you will not understand many of the more advanced concepts of
telecommunications.


Analog Signal Properties

Air is the medium that carries sound.  When we speak to one another, our
vocal chords create a disturbance of the air.  This disturbance causes
air molecules to become expanded and compress thus creating waves.  This
type of wave is called analog, because it creates a waveform similar to
the sound it represents.

Analog waves are found in nature.  They are continually flowing and have
a limitless number of values.  The sine wave is a good example of an
analog signal.


Three properties of analog signals are particularly important in
transmission:

                   amplitude      frequency      phase

Amplitude

Amplitude refers to the maximum height of an analog signal.  Amplitude
is measured in decibels when the signal is measured in the form of
audible sound. Amplitude is measured in volts when the signal is in the
form of electrical energy.


                       Amplitude of an Analog Wave


Volts represent the instantaneous amount of power an analog signal
contains.

Amplitude, wave height, and loudness of an analog signal represent the
same property of the signal.  Decibels and volts are simply two
different units of measurement which are used to quantify this property.

Frequency

Frequency is the number of sound waves or cycles that occur in a given
length of time.  A cycle is represented by a 360 degree sine wave.
Frequency is measured in cycles per second, commonly called hertz (Hz).

Frequency corresponds to the pitch (highness or lowness) of a sound. The
higher the frequency, the higher the pitch.  The high pitch tone of a
flute will have a higher frequency than the low pitch tone of a bass.

Phase refers to the relative position of a wave at a point in time.  It
is useful to compare the phase of two waves that have the same frequency
by determining whether the waves have the same shape or position at the
same time.  Waves that are in-step are said to be in phase, and waves
that are not synchronized are called out-of-phase.

Modulation


The reason these three properties are significant is that each can be
changed (modulated) to facilitate transmission.

The term modulation means imposing information on an electrical signal.

The process of modulation begins with a wave of constant amplitude,
frequency, and phase called carrier wave.  Information signals
representing voice, data, or video modulate a property (amplitude,
frequency, or phase) of the carrier wave to create a representation of
itself on the wave.

Amplitude Modulation is a method of adding information to an analog
signal by varying its amplitude while keeping its frequency constant. AM
radio is achieved by amplitude modulation.

Frequency Modulation adds information to an analog signal by varying its
frequency while keeping its amplitude constant.  FM radio is achieved by
frequency modulation.

Phase Modulation adds information to an analog signal by varying its
phase.

The modulated wave carrying the information is then transmitted to a
distant station where it is decoded and the information is extracted
from the signal. 


Properties of Digital Signals


Unlike analog signals, digital signals do not occur in nature.  Digital
signals are an invention of mankind.  They were created as a method of
coding information.  An early example of digital signals is the Morse
Code.

Digital signals have discrete, non-continuous values.  Digital signals
have only two states:


       Type of Signal                  State
       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
       Light switch           On                    Off


       Voltage                Voltage Level 1       Voltage Level 2
                              (-2 volts)            (+2 volts)

       Morse                  Short beat            Long beat



Computers and humans cannot communicate directly with each other.  We do
not understand what tiny bits and voltage changes mean.  Computers do
not understand the letters of the alphabet or words.

For computers and humans to communicate with each other, a variety of
binary (digital) languages, called character codes, have been created.
Each character of a character code represents a unique letter of the
alphabet:  a digit, punctuation mark, or printing character.

The most popular character code is call ASCII (America Standard Code for
Information Interchange).  It uses a seven bit coding scheme-- each
character consists of a unique combination of seven 1s and 0s.  For
example, the capital letter T is represented by the ASCII 1010100; the
number 3 by the ACSII 0110011. The maximum number of different
characters which can be coded in ASCII is 128).


                      English          ASCII

                         T             1010100

                         3             0110011


Another character code is called Extended ASCII.  Extended ASCII builds
upon the existing ASCII character code.  Extended ASCII codes characters
into eight bits providing 256 character representations).  The extra 127
characters represent foreign language letters and other useful symbols.


Signal Loss - Attenuation

Analog and digital signals are transmitted to provide communication over
long distances.  Unfortunately, the strength of any transmitted signal
weakens over distance.  This phenomenon is called attenuation.  Both
analog and digital signals are subject to attenuation, but the
attenuation is overcome in very different ways.


Analog Attenuation

Every kilometer or so, an analog signal must be amplified to overcome
natural attenuation.  Devices called amplifiers boost all the signals
they receive, strengthening the signals to their original power.  The
problem is that over distance, noise is created and it is boosted along
with the desired signal.

The result of using amplifiers is that both the noise (unwanted
electrical energy) and the signal carrying the information are
amplified.  Because the noise is amplified every kilometer, it can build
up enough energy to make a conversation incomprehensible.  If the noise
becomes too great, communication may become impossible.

Two different types of noise affect signal quality.

      White noise is the result of unwanted electrical signals over
      lines.  When it becomes loud enough, it sounds like the roar of
      the ocean at a distance.

      Impulse noise is caused by intermittent disturbances such as
      telephone company switch activity or lightning.  It sounds like
      pops and crack over the line.

As analog signals pass through successive amplifiers, the noise is
amplified along with the signal and therefore causes the signal to
degenerate.


Digital Attenuation

Although digital signals are also affected by attenuation, they are
capable of a much more effective method to overcome signal loss.  A
device called a regenerative repeater determines whether the incoming
digital signal is a 1 or a 0.  The regenerative repeater then recreates
the signal and transmits it at a higher signal strength.  This method is
more effective than repeating an analog signal because digital signals
can only be one of two possible states.  Remember that an analog signal
is comprised of an infinite number of states.)

The advantage of a digital regenerator is that noise is not reproduced.
At each regenerative repeater, all noise is filtered out-- a major
advantage over analog amplification.


Advantages of Digital over Analog Signals


1.    Digital regenerative repeaters are superior to analog amplifiers.

      A buildup of noise causes a distortion of the waveform.  If the
      distortion is large enough, a signal will not arrive in the same
      form as it was transmitted.  The result is errors in transmission.

      In digital transmission, noise is filtered out leaving a clean,
      clear signal.  A comparison of average error rates shows

                 Analog:    1 error every 100,000 signals

                 Digital:   1 error every 10,000,000 signals

2.    The explosion of modern digital electronic equipment on the market
      has greatly reduced its price, making digital communications
      increasingly more cost effective.  The price of computer chips,
      the brains of electronic equipment, has dropped dramatically in
      recent years further reducing the price of digital equipment.

      This trend will almost certainly continue adding more pressure to
      use digital methods.


3.    An ever increasing bulk of communication is between digital
      equipment (computer-to-computer)

      For most of telephony history, long distance communication meant
      voice telephone conversations.  Because voice is analog in nature,
      it was logical to use analog facilities for transmission.  Now the
      picture is changing.  More and more communication is between
      computers, digital faxes, and other digital transmission devices.

      Naturally, it is preferable to send digital data over digital
      transmission equipment when both sending and receiving devices are
      digital since there is no need to convert the digital signals to
      analog to prepare them for analog transmission.

Historically, telephone networks were intended to carry analog voice
traffic. Therefore, equipment was designed to create, transmit, and
process analog signals.  As technology in computers (microprocessors)
and digital transmission has advanced, nearly all equipment installed in
new facilities are digital.


	.---------------------------.
4	| Analog-Digital Conversion |
	`---------------------------'


Because it offers better transmission quality, almost every long
distance telephone communication now uses digital transmission on the
majority of their lines.  But since voice in its natural form is analog,
it is necessary to convert these.  In order to transmit analog waves
over digital facilities to capitalize on its numerous advantages, analog
waves are converted to digital waves.


Pulse Code Modulation (PCM)

The conversion process is called Pulse Code Modulation (PCM) and is
performed by a device called a codec (coder/decoder).  PCM is a method
of converting analog signals into digital 1s and 0s, suitable for
digital transmission.  At the receiving end of the transmission, the
coded 1s and 0s are reconverted into analog signals which can be
understood by the listener.


Three Step Process of PCM


Step 1 - Sampling


Sampling allows for the recording of the voltage levels at discrete
points in prescribed time intervals along an analog wave.  Each voltage
level is called a sample.  Nyquist's Theorem states:

      If an analog signal is sampled at twice the rate of the highest
      frequency it attains, the reproduced signal will be a highly
      accurate reproduction of the original.

The highest frequency used in voice communications is 4000 Hz (4000
cycles per second).  Therefore, if a signal is sampled 8000 times per
second, the listener will never know they have been connected and
disconnected 8000 times every second!  They will simply recognize the
signal as the voice of the speaker.

To visualize this procedure better, consider how a movie works.  Single
still frames are sped past a light and reproduced on a screen.  Between
each of the frames is a dark space.  Since the frames move so quickly,
the eye does not detect this dark space.  Instead the eye perceives
continuous motion from the still frames.

PCM samples can be compared to the still frames of a movie.  Since the
voice signal is sampled at such frequent intervals, the listener does
not realize that there are breaks in the voice and good quality
reproduction of voice can be achieved.  Naturally, the higher the
sampling rate, the more accurate the reproduction of the signal.  Dr.
Nyquist was the one who discovered that only 8000 samples per second are
needed for excellent voice reproduction.

The 8000 samples per second are recorded as a string of voltage levels.
This string is called a Pulse Amplitude Modulation (PAM) signal.


Step 2 - Quantizing


Since analog waves are continuous and have an infinite number of values,
an infinite number of PAM voltage levels are needed to perfectly
describe any analog wave.  In practice, it would be impossible to
represent each exact PAM voltage level.  Instead, each level is rounded
to the nearest of 256 predetermined voltage levels by a method called
Quantizing.

Quantizing assigns each PAM voltage level to one of 256 amplitude
levels.  The amplitude levels do not exactly match the amplitude of the
PAM signal but are close enough so only a little distortion results.

This distortion is called quantizing error.  Quantizing error is the
difference between the actual PAM voltage level and the amplitude level
it was rounded to.  Quantizing error produces quantizing noise.
Quantizing noise creates an audible noise over the transmission line.

Low amplitude signals are affected more than high amplitude signals by
quantizing noise.  To overcome this effect, a process call companding is
employed.  Low amplitude signals are sampled more frequently than high
amplitude signals.  Therefore, changes in voltage along the waveform
curve can be more accurately distinguished.

Companding reduces the effect of quantizing error on low amplitude
signals where the effect is greatest by increasing the error on high
amplitude signals where the effect is minimal.  Throughout this process,
the total number of samples remains the same at 8000 per second.

Two common companding formulas are used in different parts of the world.
The United States and Japan follow a companding formula called Mu-Law.
In Europe and other areas of the world, the formula is slight different
and is called A-Law.  Although the two laws differ only slightly, they
are incompatible.  Mu-Law hardware cannot be used in conjunction with
A-Law hardware.


Step 3 - Encoding

Encoding converts the 256 possible numeric amplitude voltage levels into
binary  8-bit digital codes.  The number 256 was not arrived at
accidentally. The reason there are 256 available amplitude levels is
that an 8-bit code contains 256 (28) possible combinations of 1s and 0s.
These codes are the final product of Pulse Codes Modulation (PCM) and
are ready for digital transmission.

PCM only provides 256 unique pitches and volumes.  Every sound that is
heard over a phone is one of these 256 possible sounds.

Digital-Analog Conversion

After the digital bit stream is transmitted, it must be convert back to
an analog waveform to be audible to the human ear.  This process is
called Digital-Analog conversion and is essentially the reverse of PCM.

This conversion occurs in three steps.

Step 1 - Decoding

      Decoding converts the 8-bit PCM code into PAM voltage levels.

Step 2 - Reconstruction

      Reconstruction reads the converted voltage level and reproduces
      the original analog wave

Step 3 - Filtering

      The decoding process creates unwanted high frequency noise in the
      4000 Hz - 8000 Hz range which is audible to the human ear.  A
      low-pass filter blocks all frequencies above one-half the sampling
      rate, eliminating any frequencies above 4000 Hz.


	.----------------------.
5	| Digital Transmission |
	`----------------------'

Importance of Digital Transmission

Digital transmission is the movement of computer-encoded binary
information from one machine to another.  Digital information can
represent voice, text, graphics, and video.

Digital communication is important because we use it everyday.  You have
used digital communications if

      - your credit card is scanned at the checkout line of a department
        store.

      - you withdraw money from an automated teller machine. 

      - you make an international call around the world. 

There are a million ways digital communication affects us every day.

As computer technology advances, more and more of our lives are affected
by digital communication.  A vast amount of digital information is
transmitted every second of every day.  Our bank records, our tax
records, our purchasing records, and so much more is stored as digital
information and transferred whenever and wherever it is needed.  It is
no exaggeration to say that digital communications will continue to
change our lives from now on.


Digital Voice Versus Digital Data


The difference between voice and non-voice data is this:

      Voice transmission represents voice while data transmission
      represents any non-voice information, such as text, graphics, or
      video.  Both can be transmitted in identical format--as digitized
      binary digits

In order to distinguish digital voice binary code from digital data,
since they both look like strings of 1s and 0s, you must know what the
binary codes represent.

This leads us to another important distinction-- that between digital
transmission and data transmission.  Although these two terms are often
confused, they are not the same thing.

      Digital transmission describes the format of the electrical
      signal--1s and 0s as opposed to analog waves.

      Data transmission describes the type of information transmitted-
      -text, graphics, or video as opposed to voice.

Basic Digital Terminology

A bit is the smallest unit of binary information--a "1" or a "0"

A byte is a "word" of 7 or 8 bits and can represent a unit of
information such as a letter, a digit, a punctuation mark, or a printing
character (such as a line space).

BPS (bits per second) or bit rate refers to the information transfer
rate-- the number of bits transmitted in one second.  BPS commonly refers
to a transmission speed.

Example: 

      A device rated at 19,200 bps can process more information than one
      rated at 2,400 bps.  As a matter of fact, eight times more.  Bps
      provides a simple quantifiable means of measuring the amount of
      information transferred in one second.

Bits per second is related to throughput.  Throughput is the amount of
digital data a machine or system can process.  One might say a machine
has a "high throughput," meaning that it can process a lot of information.


Digital Data Transmission


Data communications is made up of three separate parts:

   1. Data Terminal Equipment (DTE) is any digital (binary code) device,
      such as a computer, a printer, or a digital fax.

   2. Data Communications Equipment (DCE) are devices that establish,
      maintain, and terminate a connection between a DTE and a facility.
      They are used to manipulate the signal to prepare it for
      transmission.  An example of DCE is a modem.

   3. The transmission path is the communication facility linking DCEs
      and DTEs.


The Importance of Modems


A pair of modems is required for most DTE-to-DTE transmissions made over
the public network.

The function of a modem is similar to the function of a codec, but in
reverse. Codecs convert information that was originally in analog form
(such as voice) into digital form to transmit it over digital
facilities.  Modems do the opposite.  They convert digital signals to
analog to transmit them over analog facilities.

It continues to be necessary to convert analog signals to digital and
then back again because the transmission that travels between telephone
company COs is usually over digital facilities.  The digital signals
travel from one telephone company Central Office to another over high
capacity digital circuits.  Digital transmission is so superior to
analog transmission that it is worth the time and expense of converting
the analog signals to digital signals.

Since computers communicate digitally, and most CO-to-CO facilities are
digital, why then is it necessary to convert computer-generated digital
data signals to analog before transmitting them?

The answer is simple.  Most lines from a local Central Office to a
customer's residence or business (called the local loop) are still
analog because for many years, the phone company has been installing
analog lines into homes and businesses.  Only very recently have digital
lines begun to terminate at the end user's premises.

It is one thing to convert a telephone company switch from analog to
digital. It is quite another to rewire millions of individual customer
sites, each one requiring on-site technician service.  This would
require a massive effort that no institution or even industry could
afford to do all at one time.

In most cases, therefore, we are left with a public network that is part
analog and part digital.  We must, therefore, be prepared to convert
analog to digital and digital to analog.


Modulation/Demodulation


To transmit data from one DCE to another, a modem is required when any
portion of the transmitting facility is analog.  The modem (modulater/
demodulater) modulates and demodulates digital signals for
transmission over analog lines.  Modulation means "changing the
signals."  The digital signals are changed to analog, transmitted, and
then changed back to digital at the receiving end.

Modems always come in pairs-- one at the sending end and one at the
receiving end.  Their transmission rates vary from 50 bps to 56 Kbps
(Kilobits per second).


Synchronous Versus Asynchronous


There are two ways digital data can be transmitted:

Asynchronous transmission sends data one 8-bit character at a time.  For
example, typing on a computer sends data from the keyboard to the
processor of the computer one character at a time. Start and stop bits
attach to the beginning and end of each character to alert the receiving
device of incoming information.  In asynchronous transmission, there is
no need for synchronization.  The keyboard will send the data to the
processor at the rate the characters are typed.  Most modems transmit
asynchronously.

Synchronous transmission is a method of sending large blocks of data at
fixed intervals of time. The two endpoints synchronize their clocking
mechanisms to prepare for transmission.  The success of the transmission
depends on precise timing.

Synchronous transmission is preferable when a large amount of data must
be transmitted frequently.  It is better suited for batch transmission
because it groups data into large blocks and sends them all at once.

The equipment need for synchronous transmission is more expensive than
for asynchronous transmission so a data traffic study must be made to
determine if the extra cost is justified.  Asynchronous transmission is
more cost effective when data communication is light and infrequent.


Error Control


The purpose of error control is to detect and correct errors resulting
from data transmission.

There are several methods of performing error control.  What most
methods have in common is the ability to add an error checking series of
bits at the end of a block of data that determines whether the data
arrived correctly. If the data arrived with errors, it will contact the
sending DTE and request the information be re-transmitted.  Today's
sophisticated error checking methods are so reliable that, with the
appropriate equipment, it is possible to virtually guarantee that data
transmission will arrive error-free.  There are almost no reported cases
of a character error in received faxes.

Error control is much more critical in data communication than in voice
communication  because in voice communication, if one or two of the 8000
PCM signals per second arrive with an error, it will make almost no
difference to the quality of the voice representation received.  But,
imagine the consequences of a bank making a funds transfer and
misplacing a decimal point on a large account.



	.--------------.
6	| Multiplexing |
	`--------------'


Function of Multiplexers

Analog and digital signals are carried between a sender and receiver
over transmission facilities.  It costs money to transmit information
signals from Point A to Point B.  It is, therefore, of prime importance
to budget conscious users to minimize transmission costs.

The primary function of multiplexers is to decrease network facility
line costs.

Multiplexing is a technique that combines many individual signals to
form a single composite signal.  This allows the transmission of
multiple simultaneous calls over a single line.  It would cost a lot
more money to have individual lines for each telephone than to multiplex
the signals and send them over a single line.

Typical transmission facilities in use today can transmit 24 to 30 calls
over one line.  This represents a significant savings for the end user
as well as for commercial long distance and local distance carriers.


Bandwidth

The bandwidth of a transmission medium is a critical factor in
multiplexing. Bandwidth is the difference between the highest and lowest
frequencies in a given range.  For example, the frequency range of the
human voice is between 300 Hz and 3300 Hz.  Therefore, the voice
bandwidth is

           3300 Hz - 300 Hz  =  3000 Hz

We also refer to the bandwidth of a transmission medium.  A transmission
medium can have a bandwidth of 9600 Hz.  This means that it is capable
of transmitting a frequency range up to 9600 Hz.  A medium with a large
bandwidth can transmit more information and be divided into more
channels than a medium with a small bandwidth.

We will investigate three different methods of multiplexing:

                Frequency Division Multiplexing (FDM)
                Time Division Multiplexing (TDM)
                Statistical Time Division Multiplexing (STDM)


Frequency Division Multiplexing (FDM)

FDM is the oldest of the three methods of multiplexing.  It splits up
the entire bandwidth of the transmission facility into multiple smaller
slices of bandwidth.  For example, a facility with a bandwidth of 9600
Hz can be divided into four communications channels of 2400 Hz each.
Four simultaneous telephone conversations can therefore be active on the
same line.

Logically, the sum of the separate transmission rates cannot be more
than the total transmission rate of the transmission facility:  the 9600
Hz facility could not be divided into five 2400 Hz channels because  5 x
2400 is greater than 9600.

Guard bands are narrow bandwidths (about 1000 Hz wide) between adjacent
information channels (called frequency banks) which reduce interference
between the channels.

The use of FDM has diminished in recent years, primarily because FDM is
limited to analog transmission, and a growing percentage of transmission
is digital.


Time Division Multiplexing (TDM)


Time division multiplexing has two main advantages over frequency
division multiplexing:

           - It is more efficient
           - It is capable of transmitting digital signals

Instead of the bandwidth of the facility being divided into frequency
segments, TDM divides the capacity of a transmission facility into short
time intervals called time slots.

TDM is slightly more difficult to conceptualize than FDM.  An analogy
helps.

The problem is

      We must transport the freight of five companies from New York to
      San Francisco.  Each company wants their freight to arrive on the
      same day.  We must be as fair as we can to prevent one company's
      freight from arriving before another company's.  The freight from
      each company will fit into 10 boxcars so a total of 50 boxcars
      must be sent.  Essentially, there are three different ways we can
      accomplish this.

        1. We can rent five separate locomotives and rent five
           separate railway tracks and send each company's freight on
           its own line.

        2. We can rent five separate locomotives, but only one track and
           send five separate trains along one line.

        3. We can join all the boxcars together and connect them to one
           engine and send them over a single track.

Obviously the most cost effective solution is Number 3.  It saves us
from renting four extra rail lines and four extra locomotives.

To distribute the freight evenly so that each company's freight arrives
at the same time, the could be placed in a pattern as illustrated below:

  Company A + Company B + Company C + Company A + Company B + Company C . . .

At San Francisco, the boxcars would be reassembled into the original
groups of 10 for each company and delivered to their final destination.

This is exactly the principle behind TDM.  Use one track (communication
channel), and alternate boxcars (pieces of information) from each
sending company (telephone or computer).

In other words, each individual sample of a voice or data conversation
is alternated with samples from different conversations and transmitted
over the same line.

Let's say we have four callers in Boston (1, 2, 3, and 4) who want to
speak with four callers in Seattle (A, B, C, and D).  The task is to
transmit four separate voice conversations (the boxcars) over the same
line (the track).

The voice conversations are sampled by PCM.  This breaks each
conversation into tiny 8-bit packets.  For a brief moment, caller 1
sends a packet to receiver A.  Then, caller 2 sends a packet to receiver
B-- and so on.  The result is a steady stream of interleaved
packets-- just like our train example except the boxcars stretch all
across the country.  Notice that every fourth packet is from the same
conversation.  At the receiving end, the packets are reassembled and
sent to the appropriate receiver at the rate of 8000 samples per
seconds.

Remember that if the receiver hears the samples at the rate of 8000
times per second, it will result in good quality voice reproduction.
Therefore, the packets are transmitted fast enough so that every 1/8000
of a second, a packet from each send arrives at the appropriate
receiver.  In other words, each conversation is connected 8000 times per
second-- enough to satisfy Nyquist's Theorem.

In FDM the circuit was divided into individual frequency channels for
use by each sender.  In contrast, TDM divides the circuit into
individual time channels.  For a brief moment, each sender is allocated
the entire bandwidth-- just enough time to send eight bits of
information.


TDM Time Slots


Because a version of the TDM process (called STDM) is the primary
switching technique in use today, it is important that this challenging
concept be presented as clearly and understandably as possible.  Here is
a closer look at TDM, emphasizing the "T"--which stands for time.

Each transmitting device is allocated a time slot during which it is
permitted to transmit.  If there are three transmitting devices, for
example, there will be three time slots.  If there are four devices
there will be four time slots.

Two devices, one transmitting and one receiving, are interconnected by
assigning them to the same time slot of a circuit.  This means that
during their momentary shared time slot, the transmitting device is able
to send a short burst of information (usually eight bits) to the
receiving device.  During their time slot, they use the entire bandwidth
of the transmission facility but only for a short period of time.  Then,
in sequence, the following transmitting devices are allocated time slots
during which they too use the whole bandwidth.

Clock A and Clock B at either end of the transmission must move
synchronously. They rotate in unison, each momentarily making contact
with the two synchronized devices (one sender and one receiver).  For
precisely the same moment, Clock A will be in contact with Sender 1 and
Clock B will be in contact with Receiver 1, allowing one sample (8 bits)
of information to pass through. The they will both rotate so that clock
A comes into contact with Sender 2 and Clock B with Receiver 2.  Again,
one sample of information will pass.  This process is repeated for as
long as needed.

How fast must the clocking mechanism rotate?  Again, the answer is
Nyquist's theorem.  If a signal is sampled 8000 times per second, an
accurate representation of voice will result at the receiving end.  The
same theory applies with TDM.  If the clocking mechanism rotates 8000
times per second, the rate of transfer from each sender and receiver
must also be 8000 times per second.  This is so because every revolution
of the two clocking mechanisms result in each input and output device
making contact once.  TDM will not work if the clocking mechanism
synchronization is off.

Each group of bits from one rotation of the clocking mechanism is called
a frame. One method for maintaining synchronization is inserting a frame
bit at the end of each frame.  The frame bit alerts the demultiplexer of
the end of a frame.


Statistical Time Division Multiplexing (STDM)


STDM is an advanced form of TDM and is the primary switching technique
is use now.  The drawback of the TDM process is that if a device is not
currently transmitting, its time slot is left unused and is therefore
wasted.

In contrast, is STDM, carrying capacity is assigned dynamically.  If a
device is not transmitting, its time slot can be used by the other
devices, speeding up their transmission.  In other words, a time slot is
assigned to a device only if it has information to send.  STDM
eliminates wasted carrying capacity.



	.--------------------.
7	| Transmission Media |
	`--------------------'


Voice and data information is represented by waveforms and transmitted
to a distant receiver.  However, information does not just magically
route itself from Point A to Point B.  It must follow some predetermined
path.  This path is called a transmission medium, or sometimes a
transmission facility.

The type of transmission medium selected to join a sender and receiver
can have a huge effect on the quality, price, and success of a
transmission. Choosing the wrong medium can make the difference between
an efficient transmission and an inefficient transmission.

Efficient means choosing the most appropriate medium for a given
transmission.  For example, the most efficient medium for transmitting a
normal call from your home to your neighbor is probably a simple pair of
copper wires.  It is inexpensive and it gets the job done.  But if we
were to transmit 2-way video teleconferencing from Bombay to Burbank,
one pair of wires might be the least efficient medium and get us into a
lot of trouble.

A company may buy all the right equipment and understand all the
fundamentals, but if they transmit over an inappropriate medium, they
would probably be better off delivering handwritten messages than trying
to use the phone.

There are a number of characteristics that determine the appropriateness
of each medium for particular applications:

                 - cost 
                 - ease of installation 
                 - capacity
                 - rate of error

In choosing a transmission medium, these and many other factors must be
taken into consideration.


Terminology


The transmission media used in telecommunications can be divided into
two major categories:  conducted and radiated.  Examples of conducted
media include copper wire, coaxial cable, and fiber optics.  Radiated
media include microwave and satellite.

A circuit is a path over which information travels.  All of the five
media serve as circuits to connect two or more devices.

A channel is a communication path within a circuit.  A circuit can
contain one or more channels.  Multiplexing divides one physical link
(circuit) into several communications paths (channels).

The bandwidth of a circuit is the range of frequencies it can carry.
The greater the range of frequencies, the more information can be
transmitted. Some transmission media have a greater bandwidth than
others and are therefore able to carry more traffic.

The bandwidth of a circuit is directly related to its capacity to carry
information.

Capacity is the amount of information that may pass through a circuit in
a given amount of time.  A high capacity circuit has a large amount of
bandwidth-- a high range of frequencies-- and can therefore transmit a 
lot of information.

Copper Cable

Copper cable has historically been the most common medium.  It has been
around for many years and today is most prevalent in the local loop--the
connection between a residence or business and the local telephone
company.

Copper cables are typically insulated and twisted in pairs to minimize
interference and signal distortion between adjacent pairs.  Twisting the
wires into pairs results in better quality sound which is able to travel
a greater distance.

Shielded twisted pair is copper cable specially insulated to reduce the
high error rate associated with copper transmission by significantly
reducing attenuation and noise.

Copper cable transmission requires signal amplification approximately
every 1800 meters due to attenuation.

Advantages of Copper Cable

There is plenty of it and its price is relatively low.

Installation of copper cable is relatively easy and inexpensive.


Disadvantages of Copper Cable

Copper has a high error rate.

Copper cable is more susceptible to electromagnetic interference (EMI) and 
radio frequency interference (RFI) than other media.  These effects can 
produce noise and interfere with transmission.
      
Copper cable has limited bandwidth and limited transmission capacity.

The frequency spectrum range (bandwidth) of copper cable is relatively low
-- approximately one megahertz (one million Hz).  Copper circuits can be 
divided into fewer channels and carry less information than the other media.


Typical Applications of Copper Cable

Residential lines from homes to the local CO (called the local loop).

Lines from business telephone stations to an internal PBX.

Coaxial Cable

Coaxial cable was developed to provide a more effective way to isolate
wires from outside influence, as well as offering greater capacity and
bandwidth than copper cable.

Coaxial cable is composed of a central conductor wire surrounded by
insulation, a shielding layer and an outer jacket.

Coaxial cable requires signal amplification approximately every 2000
meters.


Advantages of Coaxial Cable

Coaxial cable has higher bandwidth and greater channel capacity than 
copper wire.  It can transmit more information over more channels than 
copper can.

Coaxial cable has lower error rates.  Because of its greater insulation, 
coaxial is less affected by distortion, noise, crosstalk (conversations 
from adjacent lines), and other signal impairments.

Coaxial cable has larger spacing between amplifiers.

Disadvantages of Coaxial Cable

Coaxial cable has high installation costs.  It is thicker and 
less flexible and is more difficult to work with than copper wire.

Coaxial cable is more expensive per foot than copper cable.


Typical Applications

      - Data networks

      - Long distance networks

      - CO-to-CO connections

Microwave

For transmission by microwave, electrical or light signals must be
transformed into high-frequency radio waves.  Microwave radio transmits
at the high end of the frequency spectrum --between one gigahertz (one
billion Hz) and 30 GHz.

Signals are transmitted through the atmosphere by directly aiming one
dish at another.  A clear line-of-sight must exist between the
transmitting and receiving dishes because microwave travels in a
straight line.  Due to the curvature of the earth, microwave stations
are spaced between 30 and 60 kilometers apart.

To compensate for attenuation, each tower is equipped with amplifiers
(for analog transmission) or repeaters (for digital transmission) to
boost the signal.

Before the introduction of fiber optic cable in 1984, microwave served
as the primary alternative to coaxial cable for the public telephone
companies.


Advantages of Microwave


Microwave has high capacity.  Microwave transmission offers greater
bandwidth than copper or coaxial cable resulting in higher transmission
rates and more voice channels.

Microwave has low error rates.

Microwave systems can be installed and taken down quickly and inexpensively. 
They can be efficiently allocated to the point of greatest need in a
network. Microwave is often used in rural areas because the microwave
dishes can be loaded on trucks, moved to the desired location, and
installed quickly.

Microwave requires very little power to send signals from dish to dish
because transmission does not spread out into the atmosphere.  Instead
it travels along a straight path toward the next tower.

Microwave has a low Mean Time Between Failures (MTBF) of 100,000
hours-- or only six minutes of down time per year.

Microwave is good for bypassing inconvenient terrain such as mountains
and bodies of water.

Disadvantages of Microwave


Microwave is susceptible to environmental distortions.  Factors such as
rain, snow, and heat can cause the microwave beam to bend and vary.
This affects signal quality.

Microwave dishes must be focused in a straight line-of-sight.  This can
present a problem over certain terrain or in congested cities.
Temporary physical line-of-sight interruptions, such as a bird or plane
flying through the signal pathway, can result in a disruption of
signals.

Microwave usage must be registered with appropriate regulatory agencies.
These agencies monitor and allocate frequency assignments to prevent
systems from interfering with each other.

Extensive use of microwave in many busy metropolitan areas has filled up
the airwaves, limiting the availability of frequencies.


Typical Applications

      - Private networks

      - Long distance networks


Satellite


Satellite communication is a fast growing segment of the
telecommunications market because it provides reliable, high capacity
circuits.

In most respects, satellite communication is similar to microwave
communication.  Both use the same very high frequency (VHF) radio waves
and both require line-of-sight transmission.  A satellite performs
essentially the same function as a microwave tower.

However, satellites are positioned 36,000 kilometers above the earth in
a geosynchronous orbit,  This means they remain stationary relative to a
given position on the surface of earth.

Another difference between microwave and satellite communications is
their transmission signal methods.  Microwave uses only one frequency to
send and receive messages.  Satellites use two different
frequencies--one for the uplink and one for the downlink.

A device called a transponder is carried onboard the satellite.  It
receives an uplink signal beam from a terrestrial microwave dish,
amplifies (analog) or regenerates (digital) the signal, then retransmits
a downlink signal beam to the destination microwave dish on the earth.
Today's satellites have up to 48 transponders, each with a capacity
greater than 100 Mbps.

Because of the long distance traveled, there is a propagation delay of
1/2 second inherent in satellite communication.  Propagation delay is
noticeable in phone conversations and can be disastrous to data
communication.

A unique advantage of satellite communication is that transmission cost
is not distance sensitive.  It costs the same to send a message across
the street as around the world.

Another unique characteristic is the ability to provide
point-to-multipoint transmission.  The area of the surface of the earth
where the downlinked satellite signals can be received is called its
footprint.  Information uplinked from the earth can be broadcast and
retransmitted to any number of receiving dishes within the satellite's
footprint.  Television broadcast is a common application of
point-to-multipoint transmission.


Advantages of Satellite Transmission


Satellite transmission provides access to wide geographical areas (up to the 
size of the satellite's footprint), point-to-multipoint broadcasting, a large 
bandwidth, and is very reliable.


Disadvantages of Satellite Transmission


Problems associated with satellite transmission include: propagation delay, 
licensing requirement by regulatory agencies security issue concerning the 
broadcast nature of satellite transmission.  Undesired parties within a 
satellites footprint may illicitly receive downlink transmission.

Installation requires a satellite in orbit.


Fiber Optics


Fiber optics is the most recently developed transmission medium.  It
represents an enormous step forward in transmission capacity.  A recent
test reported transmission rates of 350 Gbps (350 billion bits), enough
bandwidth to support millions of voice calls.  Furthermore, a recently
performed record- setting experiment transmitted signals 10,000 Km
without the use of repeaters, although in practice 80 to 300 Km is the
norm.  Recall the need for repeaters every kilometer or so with copper
wire and coaxial.

Fiber optics communication uses the frequencies of light to send
signals.  A device called a modulator converts electrical analog or
digital signals into light pulses.  A light source pulses light on and
off billions and even trillions of times per second (similar to a
flashlight turned on and off-- only faster). These pulses of light are
translated into binary code.  The positive light pulse represents 1; a
negative light pulse (no light) represents 0.  Fiber optics is digital
in nature.

The light is then transmitted along a glass or plastic fiber about the
size of a human hair.  At the receiving end, the light pulses are
detected and converted back to electrical signals by photoelectric
diodes.

Advantages of Fiber Optics

Fiber optics has an extremely high bandwidth.  In fact, fiber optic
bandwidth is almost infinite, limited only by the ability of engineers
to increase the frequency of the pulses of light.  Current technology
achieves a frequency of 100 terahertz (one million billion).

Fiber optics is not subject to interference or electromagnetic
impairments as are the other media.

Fiber optics has an extremely low error rate-- approximately one error
per 1,000,000,000,000.

Fiber optics has a low energy loss translating into fewer
repeaters/regenerators per long distance transmission.

Fiber is a glass and glass is made of sand.  There will never by a
shortage of raw material for fiber.


Disadvantages of Fiber Optics


Installation costs are high for a fiber optic system.  Currently it
costs approximately $41,000 per km to install a fiber optic system.  The
expense of laying fiber is primarily due to the high cost of splicing
and joining fiber.  The cost will almost certainly decrease dramatically
as less expensive methods of splicing and joining fiber are introduced.

A potential disadvantage of fiber optics results from its enormous
carrying capacity.  Occasionally a farmer or construction worker will
dig into the earth and unintentionally split a fiber optic cable.
Because the cable can carry so much information, an entire city could
lose its telephone communication from just one minor mishap.


	.-----------.
8	| Signaling |
	`-----------'

Types of Signals

When a subscriber picks up the phone to place a call, he dials digits to
signal the network.  The dialed digits request a circuit and tell the
network where to route the call--a simple enough procedure for the
caller.  But in fact, it involves a highly sophisticated maze of
signaling to and from switches and phones to route and monitor the call.
Signaling functions can be divided into three main categories.


Supervisory

      Supervisory signals indicate to the party being called and the CO
      the status of lines and trunks--whether they are idle, busy, or
      requesting service.  The signals detect and initiate service on
      requesting lines and trunks.  Signals are activated by changes in
      electrical state and are caused by events such as a telephone
      going on-hook or off-hook. Their second function is to process
      requests for telephone features such as call waiting.


Addressing

      Addressing signals determine the destination of a call.  They
      transmit routing information throughout the network.  Two of the
      most important are

      Dial Pulse:     These address signals are generated by alternately
                      opening and closing a contact in a rotary phone
                      through which direct current flows.  The number of
                      pulses corresponds to the number of the dialed
                      digit.

      Tone:           These address signals send a unique tone or
                      combination of tones which correspond to the
                      dialed digit.


Alerting

      Alerting signals inform the subscriber of call processing
      conditions.. These signals include:

           Dial tone
           The phone ringing
           Flashing lights that substitute for phone ringing
           Busy signal

Let's take a look at how signaling is used to set up a typical call over
the public network.

Step 1 -   Caller A goes off-hook

Step 2 -   The CO detects a change in state in the subscriber's line.
           The CO responds by sending an alerting signal (dial tone) to
           caller A to announce that dialing may begin.  The CO marks
           the calling line busy so that other subscribers can not call
           into it.  If another subscriber attempts to phone caller A,
           he will get the alerting busy signal.  Caller A dials the
           digits using tones from the keypad or dial pulses from a
           rotary phone.

Step 3 -   The dialed digits are sent as addressing signals from caller
           A to CO A

Step 4 -   CO A routes the addressing signals to CO B.

Step 5 -   Supervisory signals in CO B test caller B to determine if the
           line is free.  The line is determined to be free.

Step 6 -   CO B sends alerting signals to caller B, which causes caller
           B's telephone to ring.


This is an example of a local call which was not billed to the customer.
If the call had been a billable, long distance call, it would have used
a supervisory signal known as answer supervision.  When the receiving
end of a long distance call picks up, it sends a signal to its local CO.
The CO then sends an answer supervision signal to the caller's CO
telling it that the phone was picked up and it is time to begin billing.


Where on the Circuit Does Signaling Occur?

There are only three places where signaling can occur: 

      In-band means on the same circuit as voice, within the voice
      frequency range (between 300 and 3400 Hz).

      Out-of-band means on the same circuit as voice, outside of the
      voice frequency range (3400 - 3700 Hz).

      Common Channel Signaling (CCS) means signaling occurs on a
      completely separate circuit.


The frequency range of human voice is approximately 0 - 4000 Hz.
However, most voice signals fall in the area between 300 and 3400 Hz.
Therefore, to save bandwidth, telephones only recognize signals between
300 and 3400 Hz. It is conceivable that someone with an extremely high
voice would have difficulty communicating over the telephone.


In-band and Out-of-band


In-band signaling (300 to 3400 Hz) can take the form of either a single
frequency tone (SF signaling) of a combination of tones (Dual Tone
Multifrequency - DTMF).  DTMF is the familiar touch tone.

Out-of-band signaling (3400 to 3700 Hz)  is always single frequency
(SF).


In other words, using the frequency range from 300 to 3700 Hz, there are
three methods of signaling.

      Method A:       In-band (300 to 3400 Hz) by a single frequency
                      (SF)

      Method B:       In-band (300 to 3400 Hz) by multifrequencies
                      (DTMF)

      Method C:       Out-of-band (3400 to 3700 Hz) by a single
                      frequency (SF)


Single Frequency (SF) Signaling

Methods A and C are examples of Single Frequency (SF) signaling.  SF
signaling is used to determine if the phone line is busy (supervision)
and to convey dial pulses (addressing).

Method A:  In-band SF signaling uses a 2600 Hz tone which is carried
           over the frequency bandwidth of voice (remember the frequency
           bandwidth of voice is between 300 and 3300 Hz), within the
           speech path.  So as not to interfere with speech, it is
           present before the call but is removed once the circuit is
           seized and speech begins.  After the conversation is over, it
           may resume signaling.  It does not, however, signal during
           the call because it would interfere with voice which also may
           transmit at 2600 Hz.  Special equipment prevents occasional
           2600 Hz speech frequencies from accidentally setting off
           signals.

Method C:  To improve signaling performance, SF out-of-band signaling
           was developed.  It uses frequencies above the voice frequency
           range (within the 3400 to 3700 Hz bandwidth) to transmit
           signals.


The problem with Methods A and C is that they are easily susceptible to
fraud. In the late 1960s, one of the most popular breakfast cereals in
America had a promotion in which they packaged millions of children's
whistles, one in each specially marked box.  Never did General Mills,
the producer of the cereal, anticipate the fraud they would be party to.
It turned out that the whistles emitted a pure 2600 Hz tone, exactly the
tone used in Method A.  It did not take long for hackers to discover
that if they blew the whistles into the phones while making a long
distance phone call, it tricked the telephone company billing equipment
and no charge was made.

This trick grew into its own little cottage industry, culminating in the
infamous mass produced Blue Boxes which played tones that fooled
telephone billing equipment out of millions of dollars.


Method B:  DTMF was introduced to overcome this fraud, as well as to
           provide better signaling service to the customer.  Instead of
           producing just one signaling frequency, DTMF transmits
           numerical address information from a phone by sending a
           combination of two frequencies, one high and one low, to
           represent each number/letter and * and # on the dial pad.
           The usable tones are located in the center of the voice
           communication frequencies to minimize the effects of
           distortion.

Drawbacks to SF and DTMF Signaling

There are drawbacks to both SF and DTMF signaling that are promoting
their replacement in long distance toll circuits.  The most important is
that these signals consume time on the circuit while producing no
revenues.  Every electrical impulse, be it a voice conversation or
signaling information, consumes circuit time.  Voice conversations are
billable.  Signaling is not. Therefore, it is in the best interest of
the phone carriers to minimize signaling.

Unfortunately, almost half of all toll calls are not completed because
the called party is busy, not available or because of CO blockage.
Nevertheless, signals must be generated to attempt to set up, then take
down the call. Signals are generated but no revenue is produced.  For
incompleted calls, these signals compete with revenue producing signals
(whose calls were completed) for scarce circuit resources.


CCS introduced several benefits to the public network:

      .    Signaling information was removed from the voice channel, so
           control information could travel at the same time as voice
           without taking up valuable bandwidth from the voice channel.

      .    CCS sets up calls faster, reducing signaling time and freeing
           up scarce resources.

      .    It cost less than conventional signaling.

      .    It improves network performance.

      .    It reduces fraud.


Signaling System 7  (SS7)

Today the major long distance carriers use a version of CCS called
Signaling System 7  (SS7).  It is a standard protocol developed by the
CCITT, a body which establishes international standards.


Common Channel Signaling (CCS)

Common Channel Signaling (CCS) is a radical departure from traditional
signaling methods.  It transmits signals over a completely different
circuit than the voice information.  The signals from hundreds or
thousands of voice conversations are carried over a single common
channel.

Introduced in the mid-1970s CCS uses a separate signaling network to
transmit call setup, billing, and supervisory information.  Instead of
sending signals over the same communication paths as voice or data, CCS
employs a full network dedicated to signaling alone.

Loop Start Versus Ground Start Signaling

Establishing an electrical current connection with a CO can be done in
several different ways.  Here are a few of the possibilities


Loop Start

Inside of the CO, there is a powerful, central battery that provides
current to all subscribers.  Loop start is a method of establishing the
flow of current from the CO to a subscriber's phone.


The two main components of a loop start configuration are

      The tip (also called the A line) is the portion of the line loop
      between the CO and the subscriber's phone that is connected to the
      positive, grounded side of the battery.

      The ring (also called the B line) is the portion of the line loop
      between the CO and the subscriber's phone that is connected to the
      negative, ungrounded side of the battery.


To establish a loop start connection with the CO, a subscriber goes
off-hook. This closes a direct current (DC) path between the tip and
ring and allows the current to flow in a loop from the CO battery to the
subscriber and back to the battery.  Once the current is flowing, the CO
is capable of sending alerting signals (dial tone) to the subscriber to
begin a connection.

The problem with loop start signaling is a phenomenon called glare that
occurs in trunks between a CO and a PBX.  When a call comes into a PBX
from CO trunk, the only way the PBX knows that the trunk circuit is busy
is the ringing signal sent from the CO.

Unfortunately the ringing signal is transmitted at six second intervals.
For up to six seconds at a time, the PBX does not know there is a call
on that circuit.  If an internal PBX caller wishes to make an outgoing
call, the PBX may seize the busy trunk call at the same time.  The
result is confused users on either end of the line, and the abandonment
of both calls.

Ground Start 

Ground start signaling overcomes glare by immediately engaging a circuit
seize signal on the busy trunk.  The signal alerts the PBX that the
circuit is occupied with an incoming call and cannot be used for an
outgoing call.

Ground start is achieved by the CO by grounding the tip side of the line
immediately upon seizure by an incoming call.  The PBX detects the
grounded tip and is alerted not to seize this circuit for an outgoing
call, even before ringing begins.

Because ground start is so effective at overcoming glare, it is commonly
used in trunks between the CO and a PBX.


E & M

E & M signaling is used in tie lines which connect two private telephone
switches.  In E & M signaling, information is transmitted from one
switch to another over two pairs of wires.  Voice information is sent
over the first pair, just as it would be in a Loop Start or Ground Start
trunk.  However, instead of sending the signaling information over the
same pair of wires, it is sent over the second pair of wires.





