Вы находитесь на странице: 1из 24

CHAPTER 01

Introduction

In most of the practical situations single phone line is shared between many users.
(Ex. In different locations of a house, small office, etc) Normally when number of phones are
in parallel, when ring signal occurs, ring signal can be heard in all the phones. Except one
person all the others will find that the call is not for them. It is also possible that the phones
will not work due to the high load. The conversation secrecy is also another problem in this
type of parallel connection because, when two subscribers are in a conversation, if another
subscriber lifts another receiver (parallel connected) he also can listen to the conversation.
The aim of our project was to find solutions for these types of problems.

In the system we designed, number of phones are connected to a single phone line in
parallel, but it has the following features.

∗ Ring signal can be delivered to a particular phone connected to the system by pressing
an extension number.
∗ Conversation secrecy. ( When one call is connected others cannot listen it)
∗ When one subscriber is connected to another subscriber, if another subscriber wants to
call a subscriber within the system, that call is also can be connected and it does not
interfere to the other connected calls.

The system we designed has two main parts. They are


1.) Control unit
2.) Sub unit

Frequency Division Multiplexing (FDM) technique is used to separate different channels


of subscribers, therefore different frequency bands (Channels) are used. A block diagram of
the system is given in the figure 1.1

1
Telephone line Control Unit
(From PSTN)

Sub
Unit

Sub
Unit

Sub
Unit

Figure 1.1: Block diagram of the system.

The control unit has a connection to the PSTN and all the parallel phones are
connected to the control unit through the sub unit.

When an outside subscriber (in the PSTN) dials the number corresponds to the
system, he is first connected the control unit. Each parallel phone in the system is assigned an
extension number and the subscriber should press the extension number of the required phone
within 10 second after he is connected. If the extension number is not pressed within 10
seconds, the call will be disconnected.

If the extension number is pressed, the control unit identifies the phone corresponds to
that number and it gives ring signal only to that phone. If the call is not answered within 10
seconds, it will be automatically disconnected.

When one phone in the system, is connected to a call, if another subscriber (in the
system) wants to call another subscriber within the system, a different frequency channel will
be assigned for him and that call also can be connected without making interference to the
other connected calls.
More details about the implementation are given in the following chapters.

2
CHAPTER 02
Control Unit

The control unit responsible for controlling all the channels and giving ring signal to
each phone separately. A detailed illustration of the control unit is given from the following
figure.

Telephone line Ring & DTMF


(From PSTN) Identifier Unit

Ring Signal
Generator

Sub
Unit

Figure 2.1: Inside view of the control unit

As mentioned in the introduction, the external telephone line comes from the PSTN is
connected to the control unit. When an external subscriber dials the number which is assigned
to the system, he will be fist connected to the control unit. To achieve this target, ring signal
provided by the exchange should be identified and “off hook” condition (the outside
subscriber should feel that the call was answered by the system) should be created.

Under normal condition, the voltage across the telephone subscriber cable is
approximately about -48V (provided by the exchange). When a call is connected (off hook) it
falls to around 10V. Therefore, by short circuiting the telephone line through a resister
(around 1kΩ) will be identified by the exchange, as the call was answered (off hook
condition). To ring a particular phone the exchange provide about -120V AC signal to that
particular line. We used this condition to identify the ring signal and connect the outside
subscriber to our control unit.

3
2.1 Ring Identifier

As shown in the figure 2.1 the control unit has a ring identifier unit and it identifies
the 120V AC signal given by the exchange to ring the phone. It turns the relay ON, which
make the phone line short circuit across the resister (provide off hook condition to the
exchange side). An opto coupler is used to identify the AC ring signal and it provides a pulse
when it detects an AC signal on the line. This pulse is used to trigger a timer circuit which
was made by NE555 timer and it makes the relay ON and keeps 10 seconds.

Opto Coupler
Timer Circuit

Relay
Telephone line
from PSTN

Figure 2.2: Ring identifier unit

If the subscriber presses an extension number within 10 seconds, the DTMF identifier
unit will identify it and give ring signal to the required phone. If not, the subscriber will be
disconnected from the system after 10 seconds.
+v
555

Gnd Vcc
Trg Dis
BC558 Out Thr
Rst Ctl
+

Opto Coupler Relay

Telephone Line

Figure 2.3: Ring identifier unit with timer

4
2.2 DTMF Identifier Unit

Dual Tone Multi Frequency (DTMF) signaling is used to transfer phone number in
most of the telephone systems. In this system each key in the telephone, is assigned a unique
frequency (made by adding two frequencies) and the telephone is capable of generating
DTMF signals and exchange can identify them. table 2.1 shows the DTMF frequencies
assigned for different keys.

Table 2.1: DTMF frequencies


1209Hz 1336Hz 1477Hz 1633Hz
697Hz 1 2 3 A
770Hz 4 5 6 B
852Hz 7 8 9 C
941Hz * 0 # D

When a key is pressed, the addition two frequencies correspond to the row value and
the column value is generated and it represents the key.

For our project, DTMF signal identification must be used to identify the
corresponding phone and give ring signal to it, when a subscriber presses the extension
number. The DTMF identifier chip MC8870 was used to identify the extension number in our
design. It is one of the popular and standard DTMF identifier chip available in the market. It
can be connected to the telephone line through a capacitor as shown in the figure 2.4, when
the line is in off hook condition. It identifies the number corresponds to the DTMF frequency
and give the binary value of the number as the output. Therefore in our design, the binary
decoder IC CD4028 was used to decode the binary output value. The outputs of the binary
decoder IC cause to switch on relays in the ring signal generator unit and give ring signal to
the phone corresponds to the extension number.

5
Figure 2.4: DTMF identifier circuit

2.3 Ring Signal Generator Unit

To generate the ring signal, an AC supply should be given to a phone when it is in


“on hook” condition. In our ring generator unit, a 25V AC supply was given to the phones by
using 230V / 24V step down transformer and the AC supply was given to the required phone
by switching on relays as shown in the figure 2.5 The process of switching on relays is done
by the binary decider output signal, as mentioned early.

To CD4028 outputs

T1

24V AC 230V AC

Figure 2.5: The way of connecting ring signal to phones

6
Normally when number of phones connected in parallel, when the ring signal comes
all phones rings at the same time, or none of them ring due to low impedance in the line. In
our system we had to face the same problem. So instead of directly connecting the phones to
the main line, each line was connected across a diac and a diode as shown in the figure 2.6.
The AC signal was connected to the phone after the diac. Diac does not pass voltages less
than 35V, so the ring signal given to one phone is not transmitted to other phones.

Telehone Line

Diac Diac Diac

Relay Relay Relay


Diode Diode Diode

Transformer

Figure 2.6: Connecting parallel phones to the main line

7
CHAPTER 03
Sub Unit
As shown in the figure 1.1, the sub unit is located near each phone. Sub units connect
each parallel phone to the main telephone line. There are four components in a sub unit.
• Modulator
• De-modulator
• Square wave generator (oscillator)
• Low pass filter

In our project, we are going to extend a single telephone line for multiple users. When
one user is taking a call to outside, the other internal users should be able to use the same line
to communicate with each other. Normally the telephone band is defined as the band from
300Hz to 3000Hz. So, in order to accommodate the other users in the same line, we had to
use modulation techniques. Here we used AM modulation since it is simple and easy to build
the circuit.

3.1 Procedure
First we measured the band width of the normal telephone line. So that we can divide
the whole band width into smaller bands and allocate them for different users. The input
signal is a peak of 4v.

Table 3.1: Vpeak of the output signal

Frequenc 10 1 10 1 1 2 2 3 4 500k
y 0 k k 00k 50k 00k 50k 00k 00k
/(Hz)
Vpeak 4 4 4 4 3.8 3.2 3.2 2.8 2.4 2.2
/(V)

8
From our results we understood that the band width of the normal telephone line is
about 100kHz. So we decided to divide the frequency band into 10kHz sub-bands allowing to
accommodate 10 users in a single line.

Note: The initial circuit discussed in previous chapter can handle nine telephones.

The block diagram of the sub unit is given below.

Modulator

From the phone To the line


Oscillator

Low-pass
Filter De-modulator

Figure 3.1: Block diagram of the sub unit

9
3.2 Oscillator

The main function of the oscillator is generating the carrier signal for the modulator.
Here we use square wave carrier signal. Initially we made one oscillator with the carrier
frequency of 10 kHz. The IC NE555 was used to implement the oscillator.

Figure 3.2: Pin connection of NE555

The circuit diagram of the oscillator is depicted below.

Figure 3.3: Circuit diagram of the oscillator

10
The R and C values were calculated from the equation stated below. The frequency is
taken as 10 kHz.
F = 0.72 / (R1*C)

Table 3.2: Values of the timing components


component value
R1 47 kΩ
C 1 nF

3.3 The Modulator


The modulator is a circuit which multiplies two signals. One signal is the carrier signal
(here it is 10 kHz) and the other one is the modulating signal (i.e. the voice signal). In this
project we searched for many modulators which were feasible for us reminding that they
should be AM modulators. Finally we came up with two options. One is using a modulating
IC named MC 1496 and the other is a simple circuit consisting of a switch hitter with a JFET.

3.3.1 The modulator with the IC - MC 1496


Here the IC modulates the signal applied to it. It is an AM modulator IC. The circuit
diagram is given below.

11
Figure 3.4: Modulator with MC 1496

The main drawback of this circuit is that the input signal should be lower than the 60
mV. We implemented the circuit and tried to modulate the signal. But we did not succeed.
After number of tries we had abandon this option and switched to the JFET method.

3.3.2 The modulator with the JFET


This circuit was very simple and easy to implement. The circuit of the modulator is
given below.

Figure 3.5: Modulator circuit with JFET 2N5457

The audio input signal is applied to the Vin input. The square wave generated by the
oscillator (discussed previously) is applied to the Vsq input. The square wave behaves like a
switching signal to the JFET. It causes JFET -J1 to alternately switch between a short circuit
and open circuit.

When J1 is in the open circuit condition, the gain of the circuit is


Vout/Vin = +1

12
When it is in the short circuit condition, the gain of the circuit is
Vout/Vin = -1

Hence the polarity of the audio signal is reversed when the square wave switches it
state. So it is clear that the audio signal is multiplied by the square wave.

Resistors R1 and R2 and the diode D1 in the circuit prevent the gate of J1 from being
overdriven. When Vsq is positive, the gate-channel diode is forward biased and R1 limits the
current. When Vsq is negative, D1 is forward biased and voltage division between R1 and R2
limit the negative voltage applied to the gate of J1.

400mV

200mV

0V

-200mV

-400mV
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R2:1) V(U1:OUT)
Time

Figure 3.6: Modulator input and output signal

Here we use one modulator and one de-modulator for a single phone. The transmitting
signal is modulated and transmitted along the line. The de-modulation circuit is used to de-
modulate the receiving signal.

3.4 The de-modulator with the JFET


We use the same circuit as the de-modulator and the modulated signal is given as the
input for the de-modulator. The carrier signal is a square wave signal with the same frequency
which was used in the modulation.

13
Figure 3.7: De-modulator circuit with JFET 2N5457

400mV

200mV

0V

-200mV

-400mV
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R2:1) V(R24:2)
Time

Figure 3.8: Demodulator output.

14
3.5 Low pass filter

C1

R1
R2
+

-
C2

R4
R3

Figure 3.9: Block diagram of a low pass filter

An approximation for an ideal low pass filter is of the Form


1
AV ( S ) =
Pn( S )

Where, Pn(S) is a polynomial in the variable S with zeros in the left-hand plane. Active
filters permit the realization of arbitrary left hand poles for AV(S) using the operational
amplifier as the active element and only resistors capacitors for the passive elements

A common approximation for the Butterworth filter


AV 0
AV ( S ) =
Bn( S )
S=jω
2
2 AV 0
AV ( S ) =
1 + (ω / ω 0 ) 2 n

Magnitude of Bn(ω) is given by

Bn(ω ) = 1 + (ω / ω 0 ) 2 n

15
Normalized Butterworth polynomial for second order

Bn(S) = (S2+1.414S+1)

Typical second order Butterworth filter transfer function is of the form

AV ( S ) 1
=
AV 0 (S / ω0 ) 2 + 2K (S / ω0 ) + 1
AV ( S ) (1 / RC ) 2
=
AV 0 3 − AV 0 1 2
S2 +( )S + ( )
RC RC

2k = (3-AV0)
AV0= (3-2k)
For second order 2k= 1.414
AV0=1.586
( R + R4 )
AV 0 = 3
R3
R3=R4=10 k

ω 0 = 1 RC

f0 = 1
2πRC

Figure 3.10: low pass filter characteristics


Let us assume

16
R1 =R2=R, C1=C2=C
Randomly Select R=1.2 k
R1=R2=1.2 k
Cut Off Frequency f 0 = 5kHz

1
C=
2π * 5 * 1.2 * 10 6

C=26.526 nF

Available capacitor 27 nF

C1=C2=27 nF

Table 3.2: Low Pass Filter Components

COMPONENTS NOTATION VALUE


Resistors R1=R2=R 1.2 k
Resistors R3=R4 10 k
Capacitors C1=C2=C 27 nF
OP- AMP LM 741

17
C 4

0 .0 2 7 u
R 25 R 24 R 23 U 5

7
3 5
10k + O S2

V+
V2
1Vac 6
V1 1 .2 k 1 .2 k O U T
0Vdc
6Vdc 2 1
- O S1

V-
0 C 5L M 7 4 1 R 21
0 R 26
0 .0 2 7 u

4
10k 10k
R 22
10k
0

Figure 3.11: Schematic diagram of the filter

2.0V

1.5V

1.0V

0.5V
0Hz 1KHz 2KHz 3KHz 4KHz 5KHz 6KHz 7KHz 8KHz 9KHz 10KHz
V(R21:2)
Frequency

Figure 3.12: Frequency response of the designed low pass filter

18
3.6 Resister Bridge circuit
In order to separate the transmitting signal and the receiving signal hybrid transformers
are used in most of the telephone circuits. But for our design we used the resister bridge
circuit shown in the following figure because it is simple and easy to implement.

PHONE LINE TX1


R
LINE INPUT A
680
RX1
R
680 TX2
R
LINE INPUT B 680 RX2
Figure 3.13: Resistor bridge circuit

TX1, TX2 - Transmit Path


RX1, RX2 - Receive Path

This is a simplified circuit with 680 ohm resistors. The hybrid transformer type was
the most used to make telephone hybrids. Instead of that we are using this set up to isolate
transmit and receive path

This hybrid design can be used for simple experimenting when measuring telephone
equipments and such applications.

19
CHAPTER 04
Simulated results

R3

C4
1k
R2 U1

7
3 5
+ OS2 0 .0 2 7 u

V+
V2 1k
R V6 O F F = 0 6 12v V1 R 25 R 24 R 23 U5

7
V OUT
VAMPL = 1v R1 3 5
1F 0R k E Q = 5 0 0 10k + OS2

V+
2 1
- OS1

V-
6
R4 1k V22 OUT
J4 LM 741 1 .2 k 1 .2 k
0 0 6Vdc 2 1
- OS1

V-
V1 = 1v V3
V 2 = -5 v R5 D 1 J2N 5457 C 5L M 7 4 1 R 21
TD = 0 22k 0 R 26
D 1N 4148 0 .0 2 7 u

4
TR = 0
10k 10k
TF = 0 R 22
P W = 0 .1 2 5 m R7 18k
P E R = 0 .2 5 m 10k
10k 0
0

R 14

1k
R 13 U 11
7

3 5
+ OS2
V+

1k V11
R 11 6 12v
R 19 OUT
10k 2 1
- OS1
V-

R 16 1k
J3 LM 7411
0
4

V1 = 1v V13
V 2 = -5 v R 15 D 2 J2N 54571
TD = 0 22k
TR = 0 D 1N 41481
TF = 0
P W = 0 .1 2 5 m R 12 18k
P E R = 0 .2 5 m
10k

Figure 4.1: Schematic diagram of cascaded modulator, demodulator & the filter

In this figure the modulator, the demodulator and the low pass filter are depicted. The
output of the modulator was given as the input to de-modulator and the output of the
demodulator was given to the low pass filter. The waveforms of the signals at each stage are
given in the next pages.

20
1.0V

0.5V

0V

-0.5V

-1.0V
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R2:1)
Time

Figure 4.2: The audio input signal

400mV

200mV

0V

-200mV

-400mV
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R2:1) V(U1:OUT)
Time

Figure 4.3: The output signal of the modulator

21
400mV

200mV

0V

-200mV

-400mV
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R24:2)
Time

Figure 4.4: Demodulated signal input to the low pass filter

500mV

0V

-500mV
1.0ms 1.5ms 2.0ms 2.5ms 3.0ms 3.5ms 4.0ms 4.5ms 5.0ms
V(R21:2)
Time

Figure 4.5: The output signal from the low pass filter

22
CHAPTER 05
Difficulties

When we were doing the project, we had to face many difficulties. We were able to
overcome some of them. The difficulties were given below.

• We could not able to find a method to identify the ring signal and connect the
telephone line to our system.
We were able to overcome the problem by using an opto-coupler - MCT2E
• In the beginning, we were unable to make the off hook condition so that the
calling party could identify that the call is connected.
Ultimately, by connecting a resister parallel to the line, we were succeeded to
overcome this problem.
• Normally when the ring signal is given, all the phones which were connected in
parallel began to ring. We had to find a method to ring the phones separately.
By connecting phones through diacs-“DB3” we were able to solve this problem.
• The input to the modulating IC- MC 1496 is in the milivolt range (60 mV). But
our modulating signal was around 1V. So we had to switch to the JFET operated
modulator.
• We had to separate the transmitting and receiving signals from the two wire line.
Normally hybrid transformers are used. Since they are not widely available in
the market we had to find another method.
We used resister bridge circuit which behaves like a ‘Wheastone bridge’

23
CHAPTER 06
Conclusion

This project gave us hands on experience about the analog electronic circuit
applications. All the projects we did earlier were based on micro controllers, so this project
was a new experience for us. We learned about the different techniques used in telephone
systems (Ex. providing off hook condition, ring signal generation, DTMF signals, separating
transmit and receive signals from the two wire signal, etc).

We were able to use the theoretical knowledge we learned about the AM signal
modulation and demodulation practically, to design and implement the project. We learned
about different methods which can be used to implement modulators. By implementing a
butterworth low pass filter we could learn about theoretical background and designing
methods, also we had the chance to learned about the analog electronic circuit designing
methods.

Considering all these facts, the project helped us to improve our theoretical and
practical knowledge, also it allow us to use the knowledge practically for a useful purpose..

24

Оценить