# Communication Systems/Telephone System

From the western perspective, the telephone is ubiquitous. However, the
reality is that 80% of the world's population have limited access to
one.

To the average person, the telephone system is simply a 'black box'.
Relatively few people need to know how the PSTN works. The main
consideration is that it works, and is relatively inexpensive. The PSTN
has a designed reliability of 99.999% (called 5 nines reliability).

### A Telephone Call

To make a telephone call, one simply picks up the handset, enters a
number, and waits for the system to perform its magic:

::\*Lifting the handset from its cradle releases a hook switch and
causes a dc current to flow (20 - 120 ma). The central office monitors
this loop current and interprets it as a request for service.

::\*The office acknowledges the request for service by sending dial
tone. This normally occurs in less time than it takes to pickup the
handset and place it to the ear. Once dial tone has been received, the
subscriber starts to dial.

::\*In the past, when dialing, a rotary dial switch opened and closed
the loop in a predetermined manner. If one was very coordinated, it was
possible to perform the same task by flashing the hook switch. To assure
the customer that the system is responding, dial tone is removed once
dialing starts. In most systems today DTMF (**Dual Tone Multi
Frequency**) signaling is used How Telephones
Work

::\*Depending upon the office type and digits received, a number of
things might happen. In most cases, end-users are attached to what is
called a class 5 or end-office. These are the most common types of
telephone exchanges. Each class 5 office has one or more, three digit
exchange numbers. These are the first three digits in an ordinary
7-digit telephone number.

::\*If the central office includes the customer dialed exchange number,
it will know that the call is local and the other party is connected to
the same office. The office will therefor control the entire call setup
and takedown.

::\*If the first three digits do not correspond to an exchange handled
by the end-office, it will have to find a trunk line to an office that
can handle the call. This means that each office must know the exchange
numbers of all the offices within its calling area, and how to get to
them. The call setup and takedown will therefor be shared between the
two exchanges. They must monitor the call in progress and inform each
other of any change in call status.

::\*If the first digit dialed is a one, the office will recognize this
as a long distance call, and will start looking for a spare toll trunk.
A toll office has a greater knowledge base as to where distant exchanges
are located and how to get to them.

::\*The telephone system must be intelligent enough to recognize that in
a local call, only seven digits are usually required. Some very small
exchanges however, allow local calls by omitting the exchange number and
using only the last 4 digits or extension number. In large urban areas,
it may be necessary to prefix local calls with a 3-digit area code. An
international call may require up to 16 digits.

::\*Once the entire number has been received, the office at each end of
the connection must alert both parties as to what is happening. At the
originating end, a ringing tone is sent to the speaker in the handset.
At the terminating end, the office is generating a much larger ringing
voltage to activate a bell.

::\*The far-end-office monitors the line to determine if someone answers
the ringing phone. This is done by examining the DC current drawn when
the far-end customer lifts the handset, inducing loop current through
the hook switch. The far-end-office must then disconnect the ringing
before the handset reaches the ear, and signals back to the originating
office that someone has picked up the phone. The origination office must
then disconnect the ring back tone and complete the voice connection.

::\*Both end-offices monitor their respective loop currents during the
entire call to determine if one party hangs up. Once this happens, one
end-office signals the other, and dial tone is placed on the loop. This
alerts the remaining party that the connection has been terminated.

::\*If the line is in use, the central office will not set up the
connection and return a busy tone to the originator. By doing this
switching, call processing, and transmission resources are not being
tied up unnecessarily. However, there are a number of options such as
call forwarding and call waiting which modify this process.

::\*With call forwarding, a call to a busy number is routed to an
answering service. With call waiting, the calling party hears a ringing
tone, and the called party hears a beep, which they can either ignore or
signal back to the office that the new call should be given priority
over the existing call. If however, the call cannot be completed because
the system itself is too busy, it returns a fast busy tone to the
originator.

::\*In a touch-tone environment, the same procedure is followed, except
that tones are used to convey numbers to the local office instead of
interrupting loop current. Some calling features, generally known as
CLASS (Custom Local Area Signaling Services) , are available in areas
with touch-tone service.

In summary, the steps in completing a local telephone call are as
follows:

-   Initially calling party goes in off hook condition.
-   Once the DC current flows into the loop,the switching machine sends
    a dial tone to the calling party in order to make sure that it
    access to the switching machine.
-   Now the caller starts dialing the destination telephone number
    either by rotary dialer or touch tone keypad.
-   The switching machine will locate the corresponding local loop for
    the destination number.
-   After locating,the switching machine will check for on hook or off
    hook condition of the destination subscriber soas to make dc current
    to flow.Simultaneously the switching machine provide a signal path
    through it for the two local loops.
-   The switching machine sends a busy dial tone to the called
    subscriber when the destination party is off hook.
-   If the destination part is off hook when the destination party
    answers to the call and completes the loop.
-   When the destination party answers the called party,DC current flows
    through the loop.The flow of DC current through the loop indicates
    the switching machine that destination party answered the call.As a
    result,the switching machine remove the ringing and ring back
    signals.
-   If both stations are on hook,switching machine recognizes on open
    circuit on that loop and drops the connection through the switching
    machine.

#### But where do the telephone wires go?

The telephone line goes to a terminal block in a service area interface.
These are often located on a pole or small enclosure on the street. The
service area interface bundles the subscriber drop cables into a single
larger cable. These are in turn gathered together to form larger feeder
cables. The entire wiring system somewhat resembles a huge tree.

Cables coming out of a central office may have hundreds or even
thousands of pairs bundled together however by the time the cable gets
to the end user, it is generally down to about 50 pairs. An individual
subscriber consists of many cable sections spliced together. Bellcore
claims that the average U.S. subscriber line has twenty-two splices.

Feeder cables enter the central office in a large underground room
called a cable vault. Each feeder may contain hundreds of pairs of
wires, and be pressurized in order to prevent moisture or ground water
from entering and affecting the transmission characteristics of the
wire. A typical vault may contain tens of thousands of wire pairs.

The cables pass through the vault and are terminated on the vertical
side of the MDF (main distribution frame) . To protect the central
office equipment from high voltage transients, such as lightning
strikes, which may travel down the wire, the lines are surge protected
by carbon blocks or gas tubes.

The horizontal side of the MDF, connects the incoming telephone lines to
the peripheral equipment. All that is required to connect a line
appearance to a specific interface is to place a jumper between the
vertical and horizontal sides of the MDF.

Signals coming from an end-user are generally analog in nature.
Consequently, the peripheral equipment converts the signals to digital
form before passing them on to the rest of the network.

Incoming trunks from other central offices are comprised of specialized
carrier systems. They may be either analog or digital, but all new
systems are strictly digital.

Most end-user voice & data interfaces are multiplexed on to high-speed
paths, which pass through the internal switching, network before being
routed to outgoing lines or trunks. Incoming digital carrier systems may
be accepted directly into the switching network through a cross-connect
or may be demultiplexed prior to switching.

## PSTN Hierarchy

:   ![](Local_network.gif "fig:Local_network.gif")

Historically the telephone network was composed of a hierarchical
structure consisting of 5 different office types. The most common of
these is the class 5 end-office. An end-office connects directly to
subscriber telephone sets and performs switching functions over a
relatively small area. Telephone exchanges connect to subscribers by
means of local loops or lines, generally one per customer. Telephone
offices connect to each other by means of trunks.

A class 5 or end-office interconnects telephones throughout a small
service area. Each end-office may contain several three-digit exchange
numbers and is aware of other local exchange numbers held by other
offices.

Calls between offices are routed over interoffice or tandem trunks. Long
distance calls are routed to toll offices via toll trunks. The average
class 5 office serves approximately 41,000 subscribers, and covers 30
square km in an urban environment.

Some nodes may have no customers at all, and may be connected only to
other nodes. These inter-node or trunk connections are usually made by
FDM or TDM transmission links.

### Exchange Area Network

:   ![](Trunk_connection.gif "fig:Trunk_connection.gif")

An exchange network consists of local and tandem exchanges connected by
trunks. A tandem office interconnects class 5 offices by means of
twisted pair, coax, microwave, or fiber optic carriers. Alternate
routing paths between local exchanges are provided if the direct trunks
are occupied.

An exchange area includes all of the offices, which are aware of each
other, but do not involve long distance charges. In very large urban
areas, there is an overlap between exchange areas, which may also cross
over area code boundaries.

### Long Haul Network

:   ![](Long_haul_network.gif "fig:Long_haul_network.gif")

A long haul network consists of exchanges interconnected by toll
offices. Toll offices keep track of long distance charges and are
typically confined to national boundaries. These trunks consist of high
capacity coax, microwave, or glass fiber.

Messages used to control the call setup and takedown can be sent by two
basic methods. Traditionally, inter-office messages are sent over the
same channel that will carry the voice path, but in newer systems,
common channel signaling is being employed. In this method, the offices
have dedicated facilities, which are used to send inter-office messages.
There are some advantages to this, perhaps the notable being the added
degree of difficulty encountered if one wants to defraud the system.
When in-band signaling was used, it was possible for people to dial long
distance calls without being charged, if they created the tones used to
disable the toll circuit.

### Interoffice Signaling

Trunks are used to interconnect the various levels of telephone
exchanges. It is necessary for these links to exchange on a wide range
of information including:

:   

    :   • Call related signaling messages
    :   • Billing information
    :   • Routing and flow control signals
    :   • Maintenance test signals

There are two ways for telephone offices to communicate with each other
and pass on routing information. Information can be conveyed in the same
channel that will be used to convey the voice signal, or it may be
completely disassociated with it.

#### CAS

The CAS (Channel Associated Signaling) approach uses the voice channel
to send information through a trunk. For example, a 2600 Hz tone is used
in interoffice trunks to signal on-hook. A major disadvantage of this
system is that subscribers can bypass toll centers by injecting the
appropriate tones. One way to avoid this problem is by using out-of-
band signals on toll trunks. Since the customer's signal must pass
through an audio anti-aliasing filter, it is not possible to inject the
out-of-band signaling tone.

A principle advantage of in-channel signaling is that the integrity of
the voice path is checked each time a connection is established.
Out-of-band signaling allows for continuous supervision of the
connection throughout the call.

#### CCIS

The CCIS (Common Channel Interoffice Signaling) approach has the
signaling information conveyed on a facility completely separated from
the customer's voice path. This allows for a faster, more efficient
control, however the reliability of the CCS network must be considerably
greater than that of the individual voice paths. The signaling channel
may follow the same route as the final connection path, or it may be
completely disassociated with it. STPs (Switch Transfer Point) are need
in the network if the signaling path is disassociated, thus effectively
creating two networks: a speech network and a signaling network overlay.

#### SS7

Virtually all calls requiring tandem or toll office routing are
established and controlled by the SS7 signaling network.

The SS7 signaling network is a packet switching facility comprised
primarily of STPs (Signaling Transfer Point) and SCPs (Service Control
Point) connected to the PSTN SSP (Signal Switching Point). STPs are
deployed in pairs and are the brains of the system. They determine which
trunks and offices should be used in establishing inter-office
connections.

The SCP is a database that keeps track of such things as: credit card
authorization, virtual network subscriber listings, 800 number
conversion tables, billing, and other special services.

## Class 5 Office

A telephone central office
is often referred to as a switch because it switches or routes calls.
Regardless of who makes them, all class 5 offices have the same
objectives, and therefor have similar structures. The three major
components found in any modern switching systems are; the central
control, network, and peripherals.

### A Generic Communication Switch

:   ![](Generic_switch.gif "fig:Generic_switch.gif")

The internal architecture of a telecommunications switch is somewhat
like the organization of the entire system. The internal structure is
often illustrated by the traditional pyramid or hierarchical
arrangement. The control or brains of the operation are shown at the
top, and the peripheral units that connect to the outside world are
placed at the bottom.

Physically, the switch is simply a series of boxes, full of electronics:

:   ![](Nortel_DMS100_public_voice_exchange.jpg "fig:Nortel_DMS100_public_voice_exchange.jpg")

The MTBF (mean time between failure) for any PSTN switch must be very
long, since business would soon grind to a halt if telephone traffic was
interrupted for a prolonged period, but more importantly, emergency
services would be severely curtailed. For these reasons, large public
switches have a great deal of redundancy built in. Redundancy is
provided in two basic ways; hot standby and load sharing.

In the hot standby arrangement, two or more processors are fed with the
identical information and are making decisions, however, only one of
these processors is in charge and is executing decisions. In the event
of a failure, the healthy unit assumes the full load. There is no
degradation in performance, and no calls in progress should be lost.

In a load-sharing configuration, all processors are actively working but
not to their full capacity. In the event of a failure, the defective
processor is isolated from the system, and the others pick up the slack.
There may be degradation in performance, and calls in progress on the
defective processor may be lost.

#### Peripheral Layer

The periphery is the outermost layer in any system. It contains the
interface to all outgoing lines and trunks.

#### Network Layer

The network switch does the actual routing of signals from one customer
or port to another. Switching can be done in either the space or time
domain. Initially, all switching was done by mechanical contacts in the
space domain. Today most switches are digital in nature and operate
mainly in the time domain. However, some degree of space switching is
always required since signals must ultimately be routed from one line to
another.

Networks always provide some form of concentration. That is to say, not
all customers can be handled simultaneously. Statistical analysis shows
that in the majority of cases, the switch needs only to handle about 20%
of all the subscribers at one time. This is similar to highway systems
where the roads are designed to handle not all vehicles simultaneously,
but only a certain peak load.

#### Central Control

This contains the system intelligence and customer database. It knows
who the customers are, what they want, and how to provide the service
they require.

In a step x step \[step by step\] switch, the intelligence is fully
distributed and there is no central control, whereas in a crossbar
facility, all of the intelligence is resident in a central controller or
computer. In all modern systems the intelligence is somewhat
distributed, with various functional blocks contributing to the decision
making process.

At onetime there was a sharp distinction between computers and telecom
switches, but today this division is less clear, and central controllers
may be regarded as a specialized computer.

## Specialized Components

There are some components associated with telephone systems that deserve
special consideration.

### Line Cards

:   ![](LineCard2.jpg "fig:LineCard2.jpg")

Line cards are the single most common component in a telephone office.
It is a very complex device that contains a wide range of technologies.

The (telephone) line interface is often referred to as a BORSCHT
circuit.

This acronym describes the functional requirements of a standard
telephone line interface. The tip and ring leads of the telephone set
are wired through some protection devices to the line interface located
in the peripheral module. This interface must perform the following
functions:

:   

    :   B Battery feed
    :   O Over voltage protection
    :   R Ringing
    :   S Supervision & Signaling
    :   C Coding
    :   H Hybrid
    :   T Test

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:   ![](Telephone_line.gif "fig:Telephone_line.gif")

Many of these functions can be integrated into a single IC, often called
a SLIC chip (subscriber line interface chip). SLICs have been available
for the PBX market for over a decade. Recently however, they have also
become available for the central office environment as well.

:   **B - Battery Feed**

Most domestic appliances are powered from an electric utility grid. The
notable exception to this is the telephone. This is because the
telephone should still operate in the event of a power failure. Indeed,
the telephone is vital in case of disaster or emergency.

The telephone office provides a nominal -48 volt dc feed to power the
phone. This magnitude is considered the maximum safe dc operating
potential. It would not be in the telephone company's best interest to
provide a dc voltage, which could electrocute its customers, or it's own
employees. A negative potential was chosen to reduce corrosive action on
buried cables.

Multi-function telephones cannot always be powered from the telephone
exchange and often require an alternate power source. For this reason,
sophisticated line interfaces such as ISDN SAA interfaces have a 'fail
to POTS' mode. If the electric power fails, the complex phone cannot
function to full capacity. The telephone exchange can sense the local
power outage through the telephone loop and switches to POTS only
service.

The POTS loop requires a nominal -48 v at 20 -- 100 ma dc to maintain a
voice and signaling path. The earpiece in the handset does not require
biasing, but the carbon microphone does. Subscriber signaling is
performed by temporarily placing a short circuit on the loop thus
changing the loop current, which is then sensed at the central office.

:   ![](Voltage_feed.gif "fig:Voltage_feed.gif")

There are several ways to provide loop current, the simplest being a
resistor in series with a battery.

:   ![](Current_feed.gif "fig:Current_feed.gif")

Another way to provide loop current is by an electronic current source.

Although this method is quite complex, it has become quite popular with
the advent of high voltage bipolar technology. One of the more difficult
requirements to meet is the 60-dB longitudinal line balance requirement.
To achieve this, the impedance to ground on each side of the loop, must
match within 0.1%. This is easy to do with laser trimmed thick film
resistors, but a bit tricky with current sources.

:   ![](AWG_loop_length.gif "fig:AWG_loop_length.gif")

A standard telephone requires a minimum of about 20 ma. This means that
the maximum possible loop resistance is about 2000 $\Omega$. In actual
practice, the loop is generally limited to 1250 W. The maximum loop
length is determined by the wire gauge.

:   **O - Over-voltage Protection**

The two major types of over-voltage that can occur are lightning strikes
and power line contact. In both cases, the circuit must either recover
or fail-safe. Under no circumstances can a surge be allowed to propagate
further into the system, or create a fire.

Initial surge protection is provided at the MDF by gas tubes and/or
carbon blocks, which arc if the applied voltage exceeds a few hundred
volts. Since these devices take a finite time to respond, high-speed
diodes are also used at the line circuit inputs.

:   **R - Ringing**

Ringing is often provided by means of a dedicated ringing generator that
is connected onto the loop by means of a relay. It is possible to
generate ringing voltages at the line interface if the current
generators have a high enough voltage source available to them. Or
alternately, a switching converter with step up capability can be place
on the interface.

In Canada, the ringing voltage is a nominal 86 Vrms at 20 Hz, with a 2
second on and 4 second off cycle. On rural party lines, ringing codes of
long and short rings are sometimes used.

In the U.S. there are a number of fully selective and semi-selective
ringing methods used on party lines. One employs different frequencies
ranging from about 16 -- 66 Hz. In such cases, each telephone ringer is
tuned to its own frequency. Other methods use positive and negative
battery voltages or apply ringing on either the tip or ring side of the
line with respect to ground.

:   **S - Supervision & Signaling**

The central office must supervise the loop in order to identify customer
requests for service. A request for service is initiated by going
off-hook. This simply draws loop current from the CO.

Loop current at the far-end is monitored during ringing to enable the CO
to disconnect the ringing generator when the phone is answered. The
office continues to monitor the loop current at both ends of the
connection throughout the call, to determine when the call is terminated
by hanging up.

Signaling is a way to inform the CO what the customer wants. The two
basic signaling methods used in customer loops are dial pulse and
touch-tone. It is interesting to note that preferred customer loop
signaling method in analog exchanges is digital, while the preferred
method in digital exchanges is analog!

:   

    :   **MF Signaling Tones**

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:   ![](DTMF_keypad.gif "fig:DTMF_keypad.gif")

Two tones are used to perform the signaling function to eliminate the
possibility that speech be interpreted as a signal. At one time DTMF
decoders were costly and bulky devices located in a common equipment
bay, but today with the advent of LSI technology, this function can be
performed on a chip. An example is the Mitel MT8865 DTMF filter, and
MT8860 DTMF decoder.

Positions 11 to 14 are not presently being used.

:   **C - Coding**
:   ![](Companding_curves.gif "fig:Companding_curves.gif")

Telecommunications signals are seldom linearly encoded, but rather are
companded (a combination of compression & expansion). This allows for a
more uniform S/N ratio over the entire range of signal sizes. Without
companding, a 12 bit linear encoding scheme would be needed to obtain
the same S/N ratio at low volume levels. It also reduces the noise and
crosstalk levels at the receiver.

There are two principal international standards used to implement coding
algorithms today: A-law, which is used in Europe, and µ-law, which is
used in North America. These both involve 8 kHz sampling, and 8 bit A/D
& D/A conversion, thus resulting in 64 kbps digital bit streams.

Although the actual compression algorithms are continuous functions, the
codec approximates them by linear segments. A-law has 13 linear
segments, and µ-law has 15 linear segments or cords.

:   ![](Codec-segments.gif "fig:Codec-segments.gif")

Another important difference between the European and North American
codecs, can be seen by the position of the decision threshold and its
digital value.

:   ![](Codec_steps.gif "fig:Codec_steps.gif")

When a telephone call is placed between Europe and North America, it is
essential that all of these differences be accounted for. It is possible
to regenerate the analog voice by passing it through the same type of
codec that originally processed it, and then re-code with the other. An
alternate approach is to use lookup tables that translate the binary
values of one system to the other.

:   ![](Simplified_codec.gif "fig:Simplified_codec.gif")

Since the highest frequency passed is about 3.4 kHz, a great deal of
ingenuity is required to pass data at 4.8, 9.6 kbps or even higher. Note
that these are well above the Nyquist rate but considerably below the
Shannon-Hartley limit.

All modern telephone systems today employ codecs in the BORSCHT
interface to digitize the incoming analog signals. It is ironic that
although the telephone system has been updated to digital technology,
the telephone set and loop has remained analog.

By international agreement, all voice codecs use an 8 kHz sampling rate.
Since each transmitted sample is 8 bits long, the analog voice signal is
encoded into a 64 kbps binary steam. This rate determines the basic
channel data rate of most other digital communications systems.

By bypassing the codec, it is possible to send 64 kbps customer data
through the telephone system. However, because of old style signaling
schemes still in use, digital data rates are often limited to 56 kbps.

:   **H - Hybrid**

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:   ![](2-4_wire_hybrid.gif "fig:2-4_wire_hybrid.gif")

A diplexer performs a bi-directional 2-wire to 4-wire conversion. It
allows two unidirectional electrical paths to be combined into a single
bi-directional one, and vice versa. It is advantageous to separate
transmit and receive portions of the signal since it is easier to make
unidirectional amplifiers, filters, and logic devices.

One of the simplest ways to create an audio band hybrid is to use a
transformer hybrid.

:   

    :   **Single Core Transformer Hybrid**

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:   ![](Single_core_hybrid.gif "fig:Single_core_hybrid.gif")

There are several ways to split transmit and receive paths, the simplest
method uses a single core hybrid transformer.

The basic defining transformer equations are:

$$\frac{{V_1 }}
{{n_1 }} = \frac{{V_2 }}
{{n_2 }} = \frac{{V_3 }}
{{n_3 }} =  \cdots \quad \quad \quad and\quad \quad \quad I_1 n_1  + I_2 n_2  + I_3 n_3  +  \cdots  = 0$$

For a single core hybrid with a center-tapped secondary, the impedance
relationships for proper operation (conjugate matching) are:

$$for\quad \quad \quad n_2  = n_3 \quad \quad \quad Z_2  = Z_3  = 2Z_4 \quad \quad \quad and\quad \quad \quad Z_1  = \left( {\frac{{n_2 }}
{{n_1 }}} \right)^2 Z_4$$

:   ![](Hybrid_operation.gif "Hybrid_operation.gif") Note what happens
    if the transformer is driven from one of the secondary windings:

$$let\quad \quad \quad n_1  = n_2  = n_3 \quad \quad \quad \therefore I_3  =  - I_1  - I_2$$

:   

    :   But *I~1~* and *I~2~* flow in the opposite directions, therefor:

$$I_3  =  - \left( { - I} \right)_1  - I_2 \quad \quad \quad and\;if\quad \quad \quad \left| {I_1 } \right| = \left| {I_2 } \right|\quad \quad \quad then\quad \quad \quad I_3  = 0$$

This last requirement can be satisfied by adjusting the impedances
Z~1~ - Z~4~ to make the currents equal. From this we observe that
signals injected into any port emerge only at adjacent ports but not at
the opposite one.

:   ![](THL.gif "fig:THL.gif")

In a properly balanced single core hybrid the typical throughput or
insertion loss is about 3.5 dB and the THL (trans hybrid loss) is about
25 dB.

:   

    :   **Double Core Hybrid**

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:   ![](Double_core_hybrid.gif "fig:Double_core_hybrid.gif")

When properly balanced, a 2-core network can achieve a THL of 50 dB
while the insertion loss remains at about 3.5 dB. It has better
performance than the single core device, but is bulkier and more
expensive.

:   

    :   **Balancing Networks**

All telecom equipment is tested and characterized against standard
impedance terminations. These impedances are based on line surveys and
are approximate equivalent circuit representations of the outside
cabling plant. For this reason, these networks vary from country to
country. In North America, IRL (input return loss) is measured against:

:   **T - Testing**

In order to maintain a high degree of service (99.999%), the equipment
must be capable of detecting and repairing faults before the customer is
even aware that there may be a problem. As a result, a separate test
buss and access relay is provided on a line interface. Tests may be
performed in a bridged mode or with the loop and line card disconnected
from each other.

Testing can be done in three basic directions:

:   

    :   • From the line interface looking out towards the subscriber
        loop
    :   • From the loop connection looking into the line card
    :   • From the central office side of the line card

These tests are generally automated and are conducted late at night when
there is little chance that the customer will request service, thus
interrupting the test. Some of the scheduled tests may include:

:   

    :   • Transmit and receive levels
    :   • Transmit and receive frequency response
    :   • Insertion loss
    :   • Trans-hybrid loss
    :   • Quantization distortion
    :   • Aliasing distortion

Some other tests that may be performed when commissioning a line or when
a complaint is lodged, include:

:   

    :   • Impulse noise test
    :   • C-message noise
    :   • Longitudinal balance

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:   **Repeaters**

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:   ![](2_wire_repeater.gif "fig:2_wire_repeater.gif")

By placing two hybrids back to back, it is possible to create a
bidirectional amplifier or repeater. The total gain in the 4-wire path
within the repeater must not exceed the combined transhybrid loss of the
transformers. If this happens, the circuit will oscillate or sing.

The total gain in the 4-wire path within the repeater must not exceed
the combined transhybrid loss of the transformers. If this happens, the
circuit will oscillate or sing.

### Space Division Switching

The physical path between any two customers on a space switch is not
shared with anyone else. Crosspoints made from electromechanical relays
have been used to perform the interconnection, but newer systems use
semiconductors.

In a crossbar matrix, the number of inputs and outputs do not have to be
equal thus facilitating either concentration or expansion. In any case,
a total of NxM crosspoints are required. Although only one contact is
shown, many systems require two contacts if they keep the integrity of
the tip and ring leads throughout the system.

:   ![](Space_division_switch.gif "fig:Space_division_switch.gif")

The folded matrix treats inputs and outputs identically and requires
only *N(N-1)/2* crosspoints. The major disadvantage of this type of
switch is the rapid increase of crosspoints as the system size
increases. The number of crosspoints increases as *N^2^* and yet only a
maximum of *N* crosspoints can be active at any given time. Failure of a
single crosspoint prevents communication between the two devices sharing
that crosspoint.

**Multiple Stage Space Switch**

One way to avoid the cost penalties associated with full matrices, is to
organize the contacts into smaller groups. This impacts the call
processing since the switch controller must manage several contacts per
connection. Furthermore, the connection between any two subscribers may
take any one of a number of paths, thus further complicating the
decision making process.

The following sketch shows a simple three-stage space switch. It should
be remembered that since a full duplex connection is required, a second
switch supplying the return path must be provided.

:   ![](3_stage_space_switch.gif "fig:3_stage_space_switch.gif")

The total number of crosspoints in the above illustration is:

PLACE EQUATIONS HERE

The number of crosspoints required is dramatically reduced in this
multiple stage switch in comparison to a square matrix. The central
stage allows several ways to make a connection between two subscribers
therefor, single crosspoint failures can be bypassed. This results in a
more flexible and reliable system, but demands a more complex control
structure.

**Blocking**

A non-blocking network is capable of finding a path between any idle
input to any idle output. This does not mean that the system be able to
handle simultaneously all customer requests for service. In such a case,
the system may overload but the customer is not able to distinguish the
difference between blocking and overload.

The following illustration shows how blocking can occur. The solid lines
represent connections in service.

PLACE ILLUSTRATION HERE

The connections in service are not necessarily the optimum routing and
may have been forced by the prior connections. Note that in this case,
it is not possible for the last customer on switch 1:1 to contact the
last customer on switch 1:3 because there is not a free center switching
stage common to both. To overcome this, an additional center stage can
be added. To prevent blocking, 2n-1 center arrays are necessary.

The total number of crosspoints (NoC) in a non-blocking network is
therefore:

PLACE EQUATIONS HERE

As the total number of lines in increase, the total number of
crosspoints can be approximated by:

A non-blocking network of this type is known as a Clos switch, after its
inventor. Its basic characteristics include:

:   

    :   • Expansion in the first switching stage
    :   • An odd number of center stages
    :   • Concentration in the last stage

### Time Division Switching

Time domain switching is simply an application of time domain
multiplexing and may be performed on analog or digital signals. Any
number of inputs may be sequentially routed to a single output.

:   ![](TDM1.gif "fig:TDM1.gif")

This technique increases the transmission link utilization and can be
modified to support circuit switching. If a multiplexer is placed at the
input, a demultiplexer is placed at the output. This system can be used
to multiplex either analog or digital signals.

:   ![](TDM2.gif "fig:TDM2.gif")

Each customer is assigned a unique switch, but all customers share the
same internal signal path. For N customers there are 2N switches. TDM
bus switching occurs when the input sequence is not the same as the
output sequence.

Each customer is given access to the common structure for a brief
moment. If higher data rates are needed, multiple inputs can be
assigned, thus giving the customer more time to transmit a signal.

**Time Slot Interchange \[TSI\]**

If information can be arranged into a sequence, it can also be
rearranged, much like shuffling a deck of cards. This is the task of the
time slot interchange unit. Full duplex operation is achieved by
interchanging time slots in both directions.

:   ![](TSI.gif "fig:TSI.gif")

The incoming TDM channels are mapped sequentially into RAM while the
outgoing channels are read out non-sequentially. This output address
generator is simply a memory-mapped pointer governed by a central
controller.

The required memory access time can be approximated as the inverse of
the channel rate. The RAM width is determined by the number of bits in a
channel, and the length by the number of channels in a frame.

**Time Multiplexed Switching**

Combining TDM and TSI allows a channel from one digital bit stream to be
switched to any channel on another digital bit stream. A multistage time
switch can consist of cascaded switching modules. To prevent blocking, 3
or more stages are required.

:   ![](TMS_switch.gif "fig:TMS_switch.gif")

Although it appears that only time domain switching is used in this
example, it is also known as a Time-Space-Time switch or simply *TST*.
This is because the center stage is actually switching different input
lines in space to a common output line. Since there is an ambiguity in
the terminology, some manufacturers of telecommunications equipment may
refer to this as a *TTT* switch if all of the signals are digital.

For small-scale switches, space switching is most efficient. However, as
switch size increases, time domain switching tends to gain the
advantage.

It is sometimes difficult to make direct comparisons between various
telecommunications switches because the internal architectures may be
quite different. However, it is possible to compare BHCA (busy hour call
attempts) capacity or performance such as traffic intensity, under a
specified set of circumstances.

Traffic Intensity is the product of average holding time and the calling
rate and is expressed in CCS (hundred call seconds) or Erlangs.

CCS is defined as: (number of calls per hour)x(call holding time in
seconds)/100seconds Therefore, 36 CCS = 1 Erlang

An erlang is a measure of the traffic intensity and is equal to the
average number of simultaneous calls at any given moment. It represents
the total circuit usage during any time interval, divided by that
interval. It also corresponds to the minimum number of channels
necessary to carry the traffic, if it could be scheduled.

**Example**

:   

    :   Imagine for a moment that there are 8 customers in a small
        telephone system making random calls:

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```

:   ![](Erlang_example_a.gif "fig:Erlang_example_a.gif")

```{=html}
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```

:   

    :   These calls could conceivably be arranged as:
        ![](Erlang_example_b.gif "Erlang_example_b.gif")

Three, 1 hour channels could carry this traffic and still have 25
minutes left over to spare. The traffic intensity is therefor less than
3 Erlangs:

## Questions for Research

1\. What is the purpose of dial tone?

2\. What are the advantages of CAS and CCIS interoffice signaling?

3\. What is the difference between a line and a trunk? 4. How does the
line circuit monitor the state of the telephone?

4\. Why does the standard telephone exchange need to provide BORSCHT?

5\. List the differences between North American and European codecs.

6\. Under what circumstances can poor THL be tolerated?

7\. What is the difference between an echo suppresser and an echo
canceler?

8\. Where do the telephony terms tip and ring originate? Tip and Ring
come from switchboard days. The plug on operators cords had a center
\'tip\' for one conductor, and the outer \'ring\' for the other
conductor. 9. Define blocking.

10\. Why are multiple stage space (or time) switches more practical than
a single stage switch?
