Full
This document was uploaded by user and they confirmed that they have the permission to share it. If you are author or own the copyright of this book, please report to us by using this DMCA report form. Report DMCA
Overview
Download & View Full as PDF for free.
More details
- Words: 3,635
- Pages: 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 (From PSTN)
Control Unit
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 (From PSTN)
Ring & DTMF 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
BC558
Vcc Dis Thr Ctl
0
Opto Coupler
Relay
Telephone Line Figure 2.3: Ring identifier unit with timer 4
+
Gnd Trg Out Rst
+
555
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 697Hz 770Hz 852Hz 941Hz
1209Hz 1 4 7 *
1336Hz 2 5 8 0
1477Hz 3 6 9 #
1633Hz A B C 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
0
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
Diode
Diac
Diac
Relay
Relay
Diode
Diode
Transformer
Figure 2.6: Connecting parallel phones to the main line
7
Relay
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 y /(Hz) Vpeak /(V)
10 0
1 k
10 k
1 00k
1 50k
2 00k
2 50k
3 00k
4 00k
500k
4
4
4
4
3.8
3.2
3.2
2.8
2.4
2.2
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 R1 C
value 47 kΩ 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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
4.0ms
4.5ms
5.0ms
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 demodulate 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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
V(R24:2) Time
Figure 3.8: Demodulator output.
14
4.0ms
4.5ms
5.0ms
3.5 Low pass filter
C1 R1
R2
+ -
C2 R3
R4
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 ( S ) =
AV 0 Bn( S )
S=jω 2
AV 0 AV ( S ) = 1 + (ω / ω 0 ) 2 n 2
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 ) = AV 0
(1 / RC ) 2 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 C=
1 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 Resistors Resistors Capacitors OP- AMP
NOTATION R1=R2=R R3=R4 C1=C2=C LM 741
17
VALUE 1.2 k 10 k 27 nF
C 4
V1 6Vdc
1Vac 0Vdc
+
V2 1 .2 k
V+
10k
O S2
O U T
1 .2 k
2
0
-
C 5L M 7 4 1
R 26
6 1
R 21
0 .0 2 7 u
10k
O S1
5
4
0
U 5
3
V-
R 25
R 23
7
0 .0 2 7 u R 24
R 22
10k
10k
0
Figure 3.11: Schematic diagram of the filter
2.0V
1.5V
1.0V
0.5V 0Hz 1KHz V(R21:2)
2KHz
3KHz
4KHz
5KHz
6KHz
7KHz
8KHz
Frequency
Figure 3.12: Frequency response of the designed low pass filter
18
9KHz 10KHz
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
TX2
680 R
LINE INPUT B
RX2
680
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
0
R5 22k R7
7
2
1k J4
-
12v
V1
R 23
V22
LM 741
6Vdc
D 1 J2N 5457
0
D 1N 4148
1 .2 k
+
2
-
C 5L M 7 4 1
R 26
R 21
0 .0 2 7 u R 22
18k
0
10k
10k
0
R 14
R 19
10k
1k J3
R 15 22k
D 2 J2N 54571
7
+
OS2
OUT 2
-
OS1
5 6
12v
V11
1
LM 7411 4
R 16
1k U 11
V+
1k
R 11
0
3
V-
R 13
V1 = 1v V 2 = -5 v TD = 0 TR = 0 TF = 0 P W = 0 .1 2 5 m P E R = 0 .2 5 m
V13
R 12
D 1N 41481
18k
10k
0
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
OS2
OUT
1 .2 k
10k
10k
U5
3
10k
1
OS1
R 24
R 25
7
0 .0 2 7 u
6
V+
OS2
OUT V-
R1
+
V
V-
V3
V1 = 1v V 2 = -5 v TD = 0 TR = 0 TF = 0 P W = 0 .1 2 5 m P E R = 0 .2 5 m
1k
5
4
R4
0
V2
C4
1k
4
R V6 O F F = 0 VAMPL = 1v 1F 0R k E Q = 5 0 0
U1
3
V+
R2
OS1
5 6 1
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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
V(U1:OUT) Time
Figure 4.3: The output signal of the modulator
21
4.0ms
4.5ms
5.0ms
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
V(R21:2) Time
Figure 4.5: The output signal from the low pass filter
22
4.5ms
5.0ms
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
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 (From PSTN)
Control Unit
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 (From PSTN)
Ring & DTMF 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
BC558
Vcc Dis Thr Ctl
0
Opto Coupler
Relay
Telephone Line Figure 2.3: Ring identifier unit with timer 4
+
Gnd Trg Out Rst
+
555
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 697Hz 770Hz 852Hz 941Hz
1209Hz 1 4 7 *
1336Hz 2 5 8 0
1477Hz 3 6 9 #
1633Hz A B C 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
0
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
Diode
Diac
Diac
Relay
Relay
Diode
Diode
Transformer
Figure 2.6: Connecting parallel phones to the main line
7
Relay
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 y /(Hz) Vpeak /(V)
10 0
1 k
10 k
1 00k
1 50k
2 00k
2 50k
3 00k
4 00k
500k
4
4
4
4
3.8
3.2
3.2
2.8
2.4
2.2
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 R1 C
value 47 kΩ 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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
4.0ms
4.5ms
5.0ms
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 demodulate 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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
V(R24:2) Time
Figure 3.8: Demodulator output.
14
4.0ms
4.5ms
5.0ms
3.5 Low pass filter
C1 R1
R2
+ -
C2 R3
R4
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 ( S ) =
AV 0 Bn( S )
S=jω 2
AV 0 AV ( S ) = 1 + (ω / ω 0 ) 2 n 2
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 ) = AV 0
(1 / RC ) 2 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 C=
1 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 Resistors Resistors Capacitors OP- AMP
NOTATION R1=R2=R R3=R4 C1=C2=C LM 741
17
VALUE 1.2 k 10 k 27 nF
C 4
V1 6Vdc
1Vac 0Vdc
+
V2 1 .2 k
V+
10k
O S2
O U T
1 .2 k
2
0
-
C 5L M 7 4 1
R 26
6 1
R 21
0 .0 2 7 u
10k
O S1
5
4
0
U 5
3
V-
R 25
R 23
7
0 .0 2 7 u R 24
R 22
10k
10k
0
Figure 3.11: Schematic diagram of the filter
2.0V
1.5V
1.0V
0.5V 0Hz 1KHz V(R21:2)
2KHz
3KHz
4KHz
5KHz
6KHz
7KHz
8KHz
Frequency
Figure 3.12: Frequency response of the designed low pass filter
18
9KHz 10KHz
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
TX2
680 R
LINE INPUT B
RX2
680
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
0
R5 22k R7
7
2
1k J4
-
12v
V1
R 23
V22
LM 741
6Vdc
D 1 J2N 5457
0
D 1N 4148
1 .2 k
+
2
-
C 5L M 7 4 1
R 26
R 21
0 .0 2 7 u R 22
18k
0
10k
10k
0
R 14
R 19
10k
1k J3
R 15 22k
D 2 J2N 54571
7
+
OS2
OUT 2
-
OS1
5 6
12v
V11
1
LM 7411 4
R 16
1k U 11
V+
1k
R 11
0
3
V-
R 13
V1 = 1v V 2 = -5 v TD = 0 TR = 0 TF = 0 P W = 0 .1 2 5 m P E R = 0 .2 5 m
V13
R 12
D 1N 41481
18k
10k
0
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
OS2
OUT
1 .2 k
10k
10k
U5
3
10k
1
OS1
R 24
R 25
7
0 .0 2 7 u
6
V+
OS2
OUT V-
R1
+
V
V-
V3
V1 = 1v V 2 = -5 v TD = 0 TR = 0 TF = 0 P W = 0 .1 2 5 m P E R = 0 .2 5 m
1k
5
4
R4
0
V2
C4
1k
4
R V6 O F F = 0 VAMPL = 1v 1F 0R k E Q = 5 0 0
U1
3
V+
R2
OS1
5 6 1
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
V(R2:1)
2.0ms
2.5ms
3.0ms
3.5ms
V(U1:OUT) Time
Figure 4.3: The output signal of the modulator
21
4.0ms
4.5ms
5.0ms
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
V(R21:2) Time
Figure 4.5: The output signal from the low pass filter
22
4.5ms
5.0ms
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
