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TSSN Question Bank Ece IV NRCM

This document discusses trunking in telecommunication systems and networks. It defines trunking as a technique that allows multiple users to access a network by sharing transmission lines or frequencies. Trunking reduces the size of telecom networks and increases bandwidth. The document then discusses characteristics of trunked radio systems such as using a pool of frequencies and managing calls through a central controller. It notes benefits of trunked radio like improved frequency usage and greater system flexibility. The document also covers measuring telecom traffic and setting up both local and long distance calls. It defines concepts like busy hour, erlangs, and lost call assumptions in analyzing telecom networks.

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100% found this document useful (1 vote)
201 views17 pages

TSSN Question Bank Ece IV NRCM

This document discusses trunking in telecommunication systems and networks. It defines trunking as a technique that allows multiple users to access a network by sharing transmission lines or frequencies. Trunking reduces the size of telecom networks and increases bandwidth. The document then discusses characteristics of trunked radio systems such as using a pool of frequencies and managing calls through a central controller. It notes benefits of trunked radio like improved frequency usage and greater system flexibility. The document also covers measuring telecom traffic and setting up both local and long distance calls. It defines concepts like busy hour, erlangs, and lost call assumptions in analyzing telecom networks.

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books babu
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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TELE COMMUNICATION SYSTEM AND NETWORK

TSSN question bank


1. What do you mean by trunking?
Trunking is a technique used in data communications transmission systems to provide many users
with access to a network by sharing multiple lines or frequencies. As the name implies, the system
is like a tree with one trunk and many branches. Trunking is commonly used in very-high-
frequency (VHF) radio and telecommunication systems.
Trunking can also be defined as a network that handles multiple signals simultaneously. The data
transmitted through trunking can be audio, video, controlling signals or images.
Telecommunication networks all across the globe are based on trunking. Trunking reduces the size
of a telecom network and increases bandwidth. VHF radio used by police and control centers is
also based on trunking.

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Characteristics of Trunked Radio (repeater) system:


 Uses a group of similar radio frequencies to create a “pool” for radio system users to access
 Systems can be built using VHF, UHF or 800 Mhz
 A Channel is NOT a Frequency…(generally)
 A Channel (on your radio) is now a computer-generated code
 …and are now referred to as “Talk Groups”
 The entire system is managed by a computer, often referred to as the “Central Controller”
 All radios in the system also are computer-controlled, and communicate to the central
controller via one frequency known as the “Control Channel”
Trunked Radio Benefits:
 Greatly improved usage of scarce radio frequencies
 Greater overall radio system flexibility, including:
 Channel (Talk Group) capabilities
 Various user features, including Emergency Alarm, PTT ID, Channel Regrouping, Call
Alert, and Radio Inhibit
 Brings radio users together on a common radio system
 Consistency in radio coverage
2. Explain how traffic can be measured?
In telecommunication system, traffic is defined as the occupancy of the server in the network.
There are two types of traffic viz. voice traffic and data traffic. For voice traffic, the calling rate is
defined as the number of calls per traffic path during the busy hour. In a day, the 60 minutes
interval in which the traffic is highest is called busy hour (BH).
Average occupancy: If the average number of calls to and from a terminal during a period T second
is ‘n’ and the average holding time is ‘h’ seconds, the average occupancy of the terminal is given
by
nh
A
T
The average occupancy is also referred as traffic flow of traffic intensity. The international unit of
telephone traffic is the Erlang.
The erlang unit is defined as
(1) a unit of telephone traffic specifying the percentage of average use of a line or circuit (one
channel) or
(2) the ratio of time during which a circuit is occupied and the time for which the circuit is available
to be occupied. Traffic that occupies a circuit for 1 hour during a busy hour is equal to 1 erlang.
Consider these examples:

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• If the traffic intensity of a subscriber line is 1 erlang, the line is occupied for 60 minutes in an
hour.
• If a subscriber line is in use 6 minutes out of an hour (on average), the traffic intensity is 6
minutes/60 minutes or 100 mErl.
• The maximum traffic intensity of a 2-Mbps (30 PCM channels) line system is 30 erlangs, that is,
all channels are in use 60 minutes during the busy hour.
The typical average busy-hour traffic volume generated by one subscriber is in the range of 10 to
200 mErl. Low values are typical for residential use and high values for business subscribers.
3. Explain how a call is setup for both local and long distance call.
In telecommunications, a long-distance call (U.S.) or trunk call (U.K.) is a telephone
call made to a location outside a defined local calling area. Long-distance calls are typically
charged a higher billing rate than local calls. The term is not necessarily synonymous with placing
calls to another telephone area code.Long-distance calls are classified into two categories: national
or domestic calls which connect two points within the same country, and international calls which
connect two points in different countries. Within the United States there is a further division into
long distance calls within a single state (intrastate) and interstate calls, which are subject to
different regulations (counter-intuitively, calls within states are usually more expensive than
interstate calls). Not all interstate calls are long distance calls. Since 1984 there has also been a
distinction between intra-local access and transport area (LATA) calls and those between different
LATAs, whose boundaries are not necessarily state boundaries.
Before direct distance dialing (DDD), all long distance calls were established by
special switchboard operators (long distance operators) even in exchanges where calls within the
local exchange were dialed directly. Completion of long distance calls was time-consuming and
costly as each call was handled by multiple operators in multiple cities. Recordkeeping was also
more complex, as the duration of every toll call had to be manually recorded for billing purposes.
In many less-developed countries, such as Spain, Mexico, Brazil, and Egypt, calls were placed at
a central office the caller went to, filled out a paper slip, sometimes paid in advance for the call,
and then waited for it to be connected.[1] In Spain these were known as locutorios, literally "a place
to talk". In towns too small to support a phone office, placing long distance calls was a sideline for
some businesses with telephones, such as pharmacies.
In some countries, such as Canada and the United States, long-distance rates were
historically kept artificially high to subsidize unprofitable flat-rate local residential services.[citation
needed]
Intense competition between long-distance telephone companies narrowed these gaps
significantly in most developed nations in the late 20th century.
The cost of international calls varies dramatically among countries. The receiving country has
total discretion in specifying what the caller should be charged (by the originating company, who
in a separate transaction transfers these funds to the destination country) for the cost of connecting
the incoming international call with the destination customer anywhere in the receiving country.
This has only a loose, and in some cases no, relation to the actual cost. Some less-developed
countries, or their telephone company(s), use these fees as a revenue source. As pay-via-

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telephone schemes, such as the 976 and 900 numbers in the U.S., gradually rejected pornographic
materials, in some cases — Guyanais an example — U.S. phone sex companies have, by creative
contracts, used international calls as a means of being paid.
4. What is lost call assumption?
The lost call assumption implies that any attempted call which encounters congestion is
immediately cleared from the system. In such a case, the user may try again and it may cause more
traffic during busy hour.
a) The Erlang loss system may be defined by the following specifications.
b) The arrival process of calls is assumed to be Poisson with a rate of λ calls per hour.
c) The holding times are assumed to be mutually independent and identically distributed
random variables following an exponential distribution with 1/µ seconds.
Calls are served in the order of arrival. There are three models of loss systems. They are :
1. Lost calls cleared (LCC)
The LCC concept, which is used primarily in Europe or those countries accepting European
practice, assumes that the user will hang up and wait some time interval before reattempting if the
user hears the congestion signal on the first attempt.
Such calls, it is assumed, disappear from the system.
A reattempt (after the delay) is considered as initiating a new call.
The Erlang formula is based on this criterion.
2. Lost calls returned (LCR)
In LCC system, it is assumed that unserviceable requests leave the system and never return. This
assumption is appropriate where traffic overflow occurs and the other routes are in other calls
service. If the repeated calls not exist, LCC system is used. But in many cases, blocked calls return
to the system in the form of retries. Some examples are subscriber concentrator systems, corporate
tie lines and PBX trunks, calls to busy telephone numbers and access to WATS lines. Including
the retried calls, the offered traffic now comprise two components viz., new traffic and retry traffic.
The model used for this analysis is known as lost calls returned (LCR) model. The following
assumptions are made to analyse the CLR model.
a) All blocked calls return to the system and eventually get serviced, even if multiple retries
are required.
b) Time between call blocking and regeneration is random statistically independent of each
other. This assumption avoid complications arrising when retries are correlated to each
other and tend to cause recurring traffic peaks at a particular waiting time interval.
c) Time between call blocking and retry is somewhat longer than average holding time of a
connection. If retries are immediate, congestion may occur or the network operation
becomes delay system.

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3. Lost calls held (LCH)


The LCH concept assumes that the telephone user will immediately reattempt the call on receipt
of a congestion signal and will continue to redial.
The user hopes to seize connection equipment or a trunk as soon as switching equipment becomes
available for the call to be handled.
It is the assumption in the LCH concept that lost calls are held or waiting at the user’s telephone.
This concept further assumes that such lost calls extend the average holding time theoretically, and
in this case the average holding time is zero, and all the time is waiting time.
The principal traffic formula used in North America is based on the LCH concept.
In a lost calls held system, blocked calls are held by the system and serviced when the necessary
facilities become available. The total time spend by a call is the sum of waiting time and the service
time. Each arrival requires service for a continuous period of time and terminates its request
independently of its being serviced or not. If number of calls blocked, a portion of it is lost until a
server becomes free to service a call. An example of LCH system is the time assigned speech
interpolation (TASI) system.
LCH systems generally arise in real time applications in which the sources are continuously in
need of service, whether or not the facilities are available. Normally, telephone network does not
operate in a lost call held manner. The LCH analysis produces a conservative design that helps
account for retries and day to day variations in the busy horn calling intensities. A TASI system
concentrates some number of voice sources onto a smaller number of transmission channels. A
source receives service only when it is active. If a source becomes active when all channels are
busy, it is blocked and speech clipping occurs. Each speech segment starts and stops independently
of whether it is served or not. Digital circuit multiplication (DCM) systems in contrast with original
TASI, can delay speech for a small amount of time, when necessary to minimize the clipping.
5. Discuss about cross point technology.
The cost of crossbar system increases in proportion to the number of crosspoints.
Challenges
Reduction of the size of a crosspoint
Reduction of the cost of a crosspoint
Improvement of switching time

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Electromechanical Crosspoint Technology


The Electromechanical Crosspoint switches which are capable of making and breaking contacts in
1-10ms of time duration for several million times without any wear and tear are being extensively
used even today. The two types of switches widely used are Mini switches and Reed relay.
Mini Switches
These switches are made up of a precious metal like Palladium, which makes the contacts work
quieter, with their bifurcated design and high resistance to corrosion for long lasting design. These
mechanically latched switches use “V” notches for this purpose and are highly reliable in Crossbar
switching systems.
These switches mounted on Crossbars move horizontally and vertically to establish and release
contacts with a switching time of 8-10ms.
Reed Relay Switches
In order to reduce the usage of mechanical switches and increase the operating life of the switches
further, the Reed relay switches were introduced. These switches are made up of magnetic material
contacts sealed in a glass tube; this protects the contacts from getting contaminated. The following
figure illustrates the design of a reed relay switch.
A reed relay switch may be electrically or mechanically latched; it contains the contacts very close
to each other having a displacement of 0.2mm resulting in a fast switching speed of 1ms. The
construction of this relay is such that the glass tube is surrounded by a pair of coils and when
current is passed through both the coils simultaneously, a field is created. This further leads to the
reed contacts moving together. As long as it is switched on, the electrical connection is latched
and current passes through the coil.
In magnetic latching, the hysteresis of the magnetic material decides the performance. The
magnetic pole pieces required may be placed outside the glass or the contacts may act as poles by
choosing an appropriate ferromagnetic material. The reed relay is called the remreed due to the
remnance property of the contact strips. The residual magnetism lets the contacts stay intact even

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TELE COMMUNICATION SYSTEM AND NETWORK

after the currents are withdrawn and hence a demagnetizing current needs to be applied to open
the contacts.
These reed relays are placed at each Crosspoint to construct a Crosspoint matrix. Crosspoint
selection is achieved by connecting one of the coil windings of each relay in series with its vertical
neighbor and the other winding in series with its horizontal neighbor. The reed relay is excited
when the required Crosspoint is selected by pulsating the corresponding vertical and horizontal
bars simultaneously.
Electronic Cross point technology
Switching matrices employedsolid-state cross-points.
The call store is often referred to as the “scratch-pad” memory. This is temporary storage of
incoming call information ready for use, on command from the central processor. It also contains
availability and status information of lines, trunks, and service circuits under internal switch-circuit
conditions. Circuit status information is brought to the memory by a method of scanning. All
speech circuits are scanned for a busy/idle condition. The program store provides basic instructions
to the controller (central processor). In many installations, translation information is held in this
store (memory), such as DN to EN translation and trunk signaling information. A simplified
functional diagram of a basic SPC system is shown in Figure below.

6. Explain about the crossbar switching.


Principles of Crossbar Switching
The basic crossbar matrix requires atleast M × N sets of contacts and M + N or less activactors to
select one of the contacts. Fig. below illustrates the 3 × 4 crossbar switching. It contains an array
of horizontal and vertical wires (shown as a solid line). Both wires are connected to initially

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TELE COMMUNICATION SYSTEM AND NETWORK

separated contact points of switches. Horizontal and vertical bars (shown as dotted lines) are
mechanically connected to these contact points and attached to the electromagnets.

When both horizontal and vertical bars connected to the electromagnet are activated, the contact
of the intersection of the two bars will close together. Thus the contact is made and continues to
hold. When the electromagnets are deenergized both horizontal bar and vertical bars are released
from the contact. In order to prevent the catching of different crosspoint in the same circuit, a
procedure is followed to establish a connection. Accordingly, horizontal bar is energised first and
then vertical bar is energised to make contact or in reverse. But while removing contact horizontal
bar is deenergized first and then the vertical bar is deenergized.
Establishment of a connection
i) Scheme1
(a) Step1:Energize horizontal bar
(b) Step2:Energize vertical bar
(c) De-energize horizontal bar
ii) Scheme2
(a) Step1:Energize vertical bar
(b) Step2:Energize horizontal bar
(c) De-energize vertical bar

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TELE COMMUNICATION SYSTEM AND NETWORK

7. What is the function of a switching network?


Things to consider when selecting a Switch for a Network.
To select the appropriate switch for a layer in a particular network, you need to have specifications
that detail the target traffic flows, user communities, data servers, and data storage servers.
Company needs a network that can meet evolving requirements.
Traffic flow analysis is the process of measuring the bandwidth usage on a network and
analysing the data for the purpose of performance tuning, capacity planning, and making hardware
improvement decisions.
1. Future Growth
Switches comes in different sizes, features and function, choosing a switch to match a particular
network sometimes constitute a daunting task.
Consider what will happen if the HR or HQ department grows by five employees or more’ A solid
network plan includes the rate of personnel growth over the past five years to be able to anticipate
the future growth. With that in mind, you would want to purchase a switch that can accommodate
more than 24 ports, such as stackable or modular switches that can scale.
2. Performance
When selecting a switch for the* access, **distribution, or ***core layer, consider the ability of
the switch to support the port density, forwarding rates, and bandwidth aggregation requirements
of your network.
Access layer switches facilitate the connection of end node devices to the network e.g. PC,
Modems, IP phone, Printers etc. For this reason, they need to support features such as port security,
VLANs, Fast Ethernet/Gigabit Ethernet, PoE(power over Internet, and link aggregation. Port
security allows the switch to decide how many or what type of devices are permitted to connect to
the switch. This is where most Cisco comes in, they all support port layer security. Most renowned
network administrator knows this is the first line of defence.
Distribution Layer switches plays a very important role on the network. They collect the data from all
the access layer switches and forward it to the core layer switches.
Traffic that is generated at Layer 2 on a switched network needs to be managed, or
segmented into VLANs, Distribution layer switches provides the inter-VLAN routing functions so
that one VLAN can communicate with another on the network.
Distribution layer switches provides advanced security policies that can be applied to network
traffic using Access Control Lists (ACL). This type of security allows the switch to prevent certain
types of traffic and permit others. ACLs also allow you to control, which network devices can
communicate on the network.
Core layer switches: These types of switches at the core layer of a topology, which is the high-speed
backbone of the network and requires switches that can handle very high forwarding rates. The
switch that operates in this area also needs to support link aggregation (10GbE connections which
is currently the fastest available Ethernet connectivity.) to ensure adequate bandwidth coming into
the core from the distribution layer switches.
Also, core layer switches support additional hardware redundancy features like redundant
power supplies that can be swapped while the switch continues to operate. Because of the high
workload carried by core layer switches, they tend to operate hotter than access or distribution

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layer switches, so they should have more sophisticated cooling options. Many true, core layer-
capable switches have the ability to swap cooling fans without having to turn the switch off.
For example, it would be disruptive to shut down a switch at the core layer to change a power
supply or a fan in the middle of the day when the network usage is at its Peak. To perform a
hardware replacement, you could expect to have at least a 10 to 15 minute network shutdown, and
that is if you are very fast at performing the maintenance. In more realistic circumstances, the
switch could be down for 30 to 45 minutes or more, which most likely is not acceptable. With hot-
swappable hardware, there is no downtime during switch maintenance.

8. Explain in detail about two stage Network.

1) Theorem
a) For any single stage network, there exists an equivalent multistage network.
b) Simple Two-stage NxN network
c) ANxN single stage network with a switching capacity of K connections can be realized by
a two-stage network of NxK and KxN.
2) Simple Two-stage NxN network
a) First Stage: Any of the N inlets can be connected to any of the K outputs. NK switching
elements.
b) Second Stage: Any of the K inputs can be connected to any of the N outlets. NK switching
elements.
c) There are K alternative paths for any inlet/outlet pair connection.
3) Full connectivity/full availability
a) Any of the N inlets can be connected to any of the N outlets.
4) Terminology
a) Expanding network: M<N
b) Concentrating network: M>N
c) Square network: M=N
5) Architecture of General two-stage networks
a) Multiple small size matrices are used in each stage.

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b) Easy to be realized in practice.


c) Flexible in system design.
d) MxN two-stage network design
e) Decomposition: M = p x r, N = q x s
f) Switching matrices: p x s and r x q
6) Full availability: There must be at least one out let from each block in the first stage terminating
as inlet on every block of the second stage.
9. Explain the call processing functions.
CALL PROCESSING In this section, the basic steps involved in processing a call is discussed. Most digital
system follow a similar scheme. For any switching system design, the range of signals that has to be
interchanged between a terminal and system is considered. These signals described in signal exchange
diagram. The sequence of operation between subscribers and system are shown in state transition diagram
(s.t.d.).
Basic Steps to Process a Call The sequence of processing between subscribers are described below :
1. Idle state. At this state, the subscriber handset is in ‘on-hook’ condition. The
2.Call request identification.
The exchange identifies a line requiring for a service. When the handset is lifted, current flows in the line
called seize signal indicates the call request.
3.Providing dial tone. Once the seize signal is received, an exchange sends a dial tone to the calling
subscriber to dial the numbers.
4. Address analysis. Once the first digit received, the exchange removes the dialtone and collect all
numbers. Then the address is analysed for the validity of the number, local, STD or ISD etc. If the number
is invalid, a recorded message may be sent to the calling subscriber and terminates call request. 5. Called
line identification. The exchange determines the required outgoing line termination from the address that
it has received.
6. Status of called subscriber. The called line may be busy or free or unavailable or even out of service.
In the case of PBX, where the customer have a group of lines, the exchange tests each termination until
either it finds a free one or all one found busy. For busy, number unobtainable or the handset off hook, a
status signal or call progress signal is sent to the calling subscribers for line termination. Now the exchange
resumes idle state.
7. Ringing. Once, the exchange finds the called subscriber is free, power ringing is provided to the called
subscriber and audible ringing to the calling subscriber.
8. Path setup. When the called subscriber lifts his handset, the line is looped and ringing is removed. Once
the conversation started, the exchange completes the connections between the subscribers.
9. Supervision. The exchange supervises the connection to detect the end of the call for charging.
10. Clear signal. Once the need for connection is over, either customer may replace his handset. It causes
the line current seize and provides a clear signal to exchange. If the calling subscriber replaces his phone

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set, the clear signal sent to the exchange is called clear forward signal. If called subscriber do first, the clear
signal is called clear backward signal.

10. Explain the principle of grading?


In telecommunication engineering, and in particular teletraffic engineering, the quality of voice
service is specified by two measures: the grade of service (GoS) and the quality of service (QoS).
Grade of service is the probability of a call in a circuit group being blocked or delayed for more
than a specified interval, expressed as a vulgar fraction or decimal fraction. This is always with
reference to the busy hour when the traffic intensity is the greatest. Grade of service may be
viewed independently from the perspective of incoming versus outgoing calls, and is not
necessarily equal in each direction or between different source-destination pairs. "Grade of
Service" sometimes means a measure of inbound call center traffic to verify adherence to
conditions to measure the success of customers served.
On the other hand, the quality of service which a single circuit is designed or conditioned to
provide, e.g. voice grade or program grade is called the quality of service. Quality criteria for such
circuits may include equalization for amplitude over a specified band of frequencies, or in the case
of digital data transported via analogue circuits, may include equalization for phase. Criteria
for mobile quality of service in cellular telephone circuits include the probability of abnormal
termination of the call.
When a user attempts to make a telephone call, the routing equipment handling the call has to
determine whether to accept the call, reroute the call to alternative equipment, or reject the call
entirely. Rejected calls occur as a result of heavy traffic loads (congestion) on the system and can
result in the call either being delayed or lost. If a call is delayed, the user simply has to wait for the
traffic to decrease, however if a call is lost then it is removed from the system.
The Grade of Service is one aspect of the quality a customer can expect to experience when making
a telephone call.[2] In a Loss System, the Grade of Service is described as that proportion of calls
that are lost due to congestion in the busy hour. For a Lost Call system, the Grade of Service can
be measured using Equation.

For a delayed call system, the Grade of Service is measured using three separate terms:[1]

 The mean delay – Describes the average time a user spends waiting for a connection
if their call is delayed.

 The mean delay – Describes the average time a user spends waiting for a connection
whether or not their call is delayed.
 The probability that a user may be delayed longer than time t while waiting for a
connection. Time t is chosen by the telecommunications service provider so that they can
measure whether their services conform to a set Grade of Service.

11. Explain the Single stage network.

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In this case, a single stage of switching elements (SEs) exists between the inputs and the outputs of the
network. The simplest switching element that can be used is the 2 x 2 switching element (SE). The
connection in a 2×2 switch will either be straight, exchange,lower broadcast or upper broadcast as shown
in the Figure.

Straight; the upper input is transferred to the upper output and the lower input is transferred to the lower
output.
Exchange; the upper input is transferred to the lower output and the lower input is transferred to the upper
output.
Upper-broadcast; the upper input is broadcast to both the upper and the lower outputs.
Lower-broadcast; the lower input is broadcast to both the upper and the lower outputs. A two function
switch box can assume either the straight or exchange state A four function switch box can be any of the
four legitimate state .
In permutation interconnection networks the information exchange requires data transfer from input set of
nodes to output set of nodes and possible connections between edges are established by applying various
permutations in available links.

12.Explain in detail about time division space switching.


In space division switching, crosspoints are used to establish a specific connection between two subscribers.
The crosspoints of multistage space switches assigned to a particular connection is dedicated to that
connection for its duration. Thus the crosspoints can not be shared. Time division switching involves the
sharing of crosspoints for shorter periods of time. This paves way for the reassign of crosspoints and its
associated circuits for other needed connections. Therefore, in time division switching, greater savings in
crosspoints can be achieved. Hence, by using a dynamic control mechanisms, a switching element can be
assigned to many inlet-outlet pairs for few microseconds. This is the principle of time division switching.
Time division switching uses time division multiplexing to achieve switching. Two popular methods that
are used in time division multiplexing are (a) the time slot interchange (TSI) and (b) the TDM bus. In
ordinary time division mutliplexing, the data reaches the output in the same order as they sent. But TSI
changes the ordering of slots based on the desired connections. The demultiplexer separates the stots and
passes them to the proper outputs. The TDM uses a control unit. The control unit opens and closes the gates
according to the switching need. The principle of time division switching can be equally applied to analog
and digital signals. For interfacing sampled analog signals but not digitized, the analog time division
switches are attractive. But for larger switches, there are some limitations due to noise, distortion and
crosstalk which nomally occurs in PAM signals. Thus analog switching is now used only in smaller
switching systems. In this section, the analog time division switching and digital time division switching
are described briefly.

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Analog Time Division Switching .It shows a simple analog time division switching structure. The speech
is carried as PAM analog samples or PCM digital samples, occurring at 125 µs intervals. When PAM
samples are switched in a time division manner, the switching is known as analog time division switching.
If PCM binary samples are switched, then the switching is known as digital time division switching. A
single switching bus supports a multiple number of connections by interleaving PAM samples from receive
line interfaces to transmit line interfaces. There are two cyclic control stores. The first control store controls
gating of inputs onto the bus one sample at a time. The second control store operates in synchronism with
the first and selects the appropriate output line for each input sample

The selection of inlet/outlet is controlled by various ways. The (a) cyclic control and (b) memory based
control are the important controls and described in the following paragraphs. Cyclic control. The cyclic
control is organised by using Modulo-N counter and k to 2k decoder as shown in Fig
The numerator 125 µ sec indicates the time taken to scan inlet and outlet and the denomintor ts is the time
in µ sec to setup connection. Memory based control. Full availability can be achieved if any one control is
made memory based. If the input side is cyclically switched and the outlets are connected based on the
addresses of the outlets stored in contiguous location is referred as input controlled or input driven. If the
outlets are cyclically switched, the switch is referred as output controlled or output driven. As the physical
connection is established between the inlet and the outlet through the common bus for the duration of one
sample transfer, the switching technique is known as time division space multiplexing. with certain changes
in network). The switching capacity in the memory controlled is equal to N. The use of cyclic control in
input or output controlled switches restricts the number of subscribers on the system rather than the
switching capacity since all the lines are scanned whether it is active or not. No restictions on subscriber
number and full availability of the switching system can be achieved by designing a switching configuration
with control memory for controlling both inlets and outlets. This configuration referred to as memory
controlled time division space switch is shown in Fig. As each word of the control memory has inlet address
and an outlet address. The control memory words are readout one after another. The modulo counter is
updated at the clock rate. For the path setup of kth inlet and jth outlet, the addresses are entered in control
memory and path is made. Then the location is marked busy. When conversation is terminated, the
addresses are replaced by null values and location is marked free. Hence

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TELE COMMUNICATION SYSTEM AND NETWORK

C = 125/ ts µ sec,
where ts = ti + tm + td + tt.

What do you mean by customer line signaling?


In a local telephone network loop/disconnect signalling is used for sending customers call and clear signals
to the exchange. Since there is a minimum line current that the exchange can detect, there is a maximum
permissible line resistance. This limits the maximum length of line and the size of the area served by the
exchange. In addition to being limited by DC resistance, the length of lines is also limited by permissible
attenuation at voice frequencies. Ideally both limits be the same.

Call between two customers on the same exchange require a number of actions to be performed
inresponse to signals as shown. If the call setup between two subscribers are made through many
exchanges and trunks, the originating exchange where calling subscriber is connected sends the seize and
then address to the terminating exchange where the called subscriber is connected. Remaining signaling
are similar to the local call, but through the originating and terminating exchanges. In electromechanical
system, the signalling between exchanges are sent through same inter exchange circuits referred as
channel associated signaling. In SPC controlled exchanges, interexchange signals are generated at
orginating exchange, but processed at terminating exchange. The signals are transferred over high speed
data like instead of speech connections are referred as common channel signaling.

13. Write short note on Signaling?


A signalling system in a data communication networks exchanges signalling information effectively
between subscribers. The signalling systems are essential building blocks in providing the ultimate
objective of a worldwide automatic telephone services standardized. Signalling provides the interface
between different national systems. The introduction of signalling system was the big step in improving the
PSTN. The consultative committe on international telegraphy and telephony (CCITT) based in Geneva,
recommended seven formats related to signalling. The first five formats related to Inband signalling and
the last two in the category of common channel signalling. In In-band signalling, voice information and

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TELE COMMUNICATION SYSTEM AND NETWORK

signalling information travel on common paths, where as in common channel signalling, they travel on
separate paths.

14. Discuss the FDM carrier system.


FDM is a broad band analog transmission technique in which multiple signals are transmitted over a single
cable simultaneously as shown in Fig. 3.21 (a). FDM systems divides the available BW of the transmission
medium into a number of narrow band or sub channels. The channels are sent over a common path by
modulation each channel to different carrier frequency (higher frequency). The signal thus occupies a
relatively narrow bandwidth which is a part of a much wider bandwidth transmitted. Each speech channel
occupies 4 kHz of the available bandwidth. It shows FDM in telephone transmission.

All multiplex equipment in the FDM hierarchy uses SSB modulation. Each level of the hierarchy is
implemented using a set of standard FDM modules. The multiplex equipment is independent of particular
transmission media. FDM principle. FDM for 12 telephone channels (group multiplex level) is considered
for example. The signals pass through 12 low pass filters (LPF) to remove any high frequency components.
The LPF outputs are mudulated on 12 separate carrier signals separated by 4 kHz. The modulations in all
FDM hierarchy is single side band (SSB) modulation. The output of each of the 12 modulations must be
filtered (band pass filter) to avoid the interference with each other. BPF are used to restrict each signal to
the allocated 4 kHz band. The principle of operation is shown in Fig. 3.22. When the signal is received,
converse process takes place as shown. In fact, the modulators and demodulators are combined into 12
single units to permit two way transmissions.

15.Define signaling.
A signalling system in a data communication networks exchanges signalling information effectively
between subscribers. The signalling systems are essential building blocks in providing the ultimate

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TELE COMMUNICATION SYSTEM AND NETWORK

objective of a worldwide automatic telephone services standardized. Signalling provides the interface
between different national systems. The introduction of signalling system was the big step in improving the
PSTN. The consultative committe on international telegraphy and telephony (CCITT) based in Geneva,
recommended seven formats related to signalling. The first five formats related to Inband signalling and
the last two in the category of common channel signalling. In In-band signalling, voice information and
signalling information travel on common paths, where as in common channel signalling, they travel on
separate paths.

Points to remember:
1. The links that run between the switching systems are called as Trunks
2. Both Relays and Latches are adopted by cross bar systems with hard wired control subsystem.
3. In folded type of network, output lines are folded back to input lines.
4. One sample duration, 125/M microseconds, is usually referred to as a Time slot.
5. The incoming traffic = Incoming calls × (avg. time duration of incoming calls) / 60.
6. In Queuing systems, the trunks are often called servers.
7. This technique interconnecting the multiples of switches is called Grading.
8. For a digital signal, the number of signal elements transmitted per second is called Signaling
rate.
9. The measure of the ability of user to access a trunked system during the busiest hour is trunking.
10. The unit for the measure of traffic intensity is Erlang.
11. The kind of switching technique indicates the transfer of coded values from input to output
during the same interval of time is Space Switching.
12. Time slot interchange is possible in Time multiplexed time switching.
13. The control functions are performed by a computer or processor, and then it is called as stored
program control.
14. Networks with point- to - point links among all the entities are known as fully connected
networks.
15. Three stage network mainly consists of three stage concentrator networks.
16. A Telephone exchange is a circuit switched system and its switching network is a lost call
system.
17. FDM systems are primarily used for sending following types of signals Analog signals.
18. The sequence of operations in which PCM is done is Sampling, quantizing, encoding.
19. A crossbar switching with stored programmed control qualifies as Electronic exchange.
20. Erlang is a unit of traffic.

NRCM ,ECE Page 17

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