Com Lab Manual

communication System Lab Aim: Study of delta modulation /demodulation system Objective:
(a) To understand the working principle of delta modulation and demodulation. (b) Study of slope overload distortion and methods those are used for minimising it.
Requirement: Trainer Kit ST2105

CRO
Introduction:
Delta modulation is a system of digital modulation developed after pulse code modulation. In this system, at each sampling time, say the kth sampling time, the difference between the sample value at sampling time K and the sample value at (the previous sampling time (k-1) is encoded into just a single bit), thus the O/P from the modulator is a series of zeros and ones. Working of delta modulator: The analog signal, which is to be encoded into a digital data, is applied to the positive input of the voltage comparator, which compares it with the signal applied to its negative input from the integrator O/P. The comparators O/P is logic High or Low depending on whether the input signal at the terminal is lower or greater than the negative terminal input signal. The comparators O/P is then lathed into a flip-flop, which is clocked by the transmitter clock. This binary data stream is transmitted to receiver and is also fed to the unipolar to bipolar converter block. This block convert logic 0 to voltage level of +4V and logic 1 to voltage level 4V Bipolar O/P is applied to the integrator, whose O/P is rising linear ramp signal when -4V is applied to it and falling linear ramp signal when +4V is applied to it. The integrator O/P is then connected to the negative terminal of voltage comparator. Delta demodulation consists of flip-flop, a unipolar to bipolar converter followed by an integrator and LPF.
Procedure:
1. Connect the mains supply 2. Connect the board as shown in diagram 1. 3. Select clock frequency selector block switches A & B are in A=0 & B=0 position. 4. The integrator 1 blocks switches are in following positions: a. Gain control switch in left-hand position b. Switch A & B are in A=0 & B=0 positions.
5. The integrator 2 block switches are in following positions: a. Gain control switch in right hand position b. Switch A & B are in A=0 & B=0 positions. 6. First connect the + input of the delta modulators VOLTAGE COMPARATOR to 0V and monitor on an oscilloscope O/P of integrator 1 (t.p.17) 7. Adjust the transmitters LEVEL CHANGER preset until the O/P of integrator (t.p.17) is triangle wave cantered around 0 volts. The peak to-peak amplitude of the triangle wave at the integrators O/P should be 0.5V(approx) this amplitude is known as the integrator STEP size. The O/P from the transmitter BISTABLE circuit (t.p.14) will now be stream of alternate 1 and 0s this is also the O/P of delta modulator itself. 8. Examine the signal at the O/P of the INTEGRATOR 2 (t.p.47) at the Receiver. This should be triangle wave, with step size equal to that of integrator 1, and ideally centred around 0 volt. otherwise remove it by adjusting the receivers LEVEL ADJUST preset. The receivers LPF leave a DC level at the filters O/P (t.p.51). 9. Now disconnect the voltage comparators + input from OV and reconnect it to the 250Hz O/P from the FUNCTION GENERATOR block; i.e 250Hz sine-wave. 10. Display the data of the transmitters BISTABLE (t.p.14) together with the analog input at t.p.9 and note that the 250Hz sine wave has effectively been encoded into a stream of data bits. 11. Now display the O/P of INTEGRATOR 2 (t.p.47) together with the O/P of the receivers LPF block (t.p.51), note that some ripple still remains at the filters O/P. This ripple is due to quantisation noise at the integrators O/P, which is caused by relatively large integrator step size. 12. This step size can be reduced by increasing the rate at which the system is clocked that is (the sampling frequency) since this reduces the sampling period and hence the time available between samples for the integrators to charges up and down. 13. Now increases the system clock frequency to 64KHz, by putting the switches in the A=0, B=1 position in CLOCK FREQUENCY SELECTOR block. 14. Now examine the ripple at the LPFs O/P (t.p.51) note that this is now less than it was before. 15. By changing the system Clock frequency to first 128KHz (A=1, B=0) and then to 256KHz (A=1, B=1 position) note the improvement in the LPFs O/P signal (t.p.51). Once again, it may be necessary to adjust slightly the transmitters LEVEL ADJUST preset in order to obtain a stable oscilloscope trace. 16. Now disconnect the comparators + input from 250Hz sine wave O/P and reconnect it to the 500Hz, 1KHz and 2KHz O/P in turn. Note that as the frequency of the analog signal increases so the LPFs O/P becomes more distorted and reduced in amplitude.
17. Now when the comparators + input is 2KHz sine wave then examine the O/P of INTEGRATOR 2 (t.p.47) which is no longer an approximation to the analog i/p signal but is instead somewhat triangular in shape. Since the analog signal is now changing so quickly that the integrators O/P cant ramp fast enough to Catch Up with it and the result is known as Slope overloading 18. Although the system clock frequency (i.e. the sampling freq.) determine how often the integrators o/p direction (up or down) can change, it does not affect how quickly the integrators o/p can ramp up and down. 19. Slope over loading can be avoided by a. Reducing the frequency of the analog input signal since there was no problem with the 250Hz analog i/p b. Reducing the amplitude of the i/p signal. This can be shown by slowly tuning the 2KHz preset anticlockwise. c. By increasing the gain of the integrators, so that they can ramp up and down faster. To illustrate this, first return to 2KHz preset to its fully clock wise (max amplitude) position, so that slope overloading can once again be seen on the scope. d. At both integrator blocks there are integrator gain switch A&B: now change the position A&B switches from A=0, B=0 to A=0, B=1, to double the gain of the two integrators, note that a slope overloading still occurs and continue to A=1, B=1 slope overloading can be eliminated. Result: Precautions (if any):
Communication System Lab

Aim: To generate an AM signal measuring its depth of modulation and demodulate it. Objective: (a) Study of AM modulation. (b) Measurement of depth of modulation.
Equipment Required: Trainer kit for AM modulation, function generator, CRO. Theory: Amplitude modulation is a process in which the amplitude of carrier is varied in accordance with the modulating signal. m= Procedure: 1. Switch on the experiment kit. 2. Connect the carrier signal generator to the terminals where carrier is written (1&2) 3. Connect the CRO on output terminals of modulator. 4. Connect the function generator oscillator to the modulating signal terminals i.e. between 3 & 4. 5. Trace the waveshapes, which are amplitude modulated wave. 6. Note the Vmax and Vmin of modulated wave. 7. Calculate the percentage modulation. 8 Observe the waveform on the CRO and compare it with the input waveform. 9 Trace the waveshapes, which is modulating signal. 10 Disconnect the CRO and AF oscillator from the circuit. 11 Connect the AF oscillator to CRO and trace the wave shape. V max v min 100% V max = V min
AF in Regulated Power Supply Amplitude Modulator
AM Signal
Carrier Generation
Amplitude Demodulator
Modulating Signal
Modulated Signal
Demodulated O/P
Result:
Precautions (If any):
Questions: 1. Indicate the false statement. Modulation is used to a. Amplify lower frequency signals b. Allow the use of practicable antennas c. Increase the rate of transmission d. Reduce the bandwidth of transmission. 2. Draw the diode detector circuit and explain its action. 3. For an AM signal, modulated to a depth of 100% by a sinusoidal, the total power is a. Same as that of the carrier b. Twice as large as that of the carrier c. 50% more than that of the carrier. 4. What are the limitations of amplitude modulation? 5. The modulation index of AM wave is changed from 0 to 1, the transmission power is a. Unchanged b. Doubled c. Increased by 50%. 6. Amplitude modulated signals are detected by a. a synchronous detector b. an envelope detector c. a ring demodulator. 7. Vestigial sideband transmission is used in a. TV Transmission b. RT System c. Mobile telephony
8. What do you understand by coherent and non-coherent detection of signals? 9. The O/P of a diode detector contains a. Modulating signal or b. dc voltage or c. Both (a) and (b). 10. What is the necessity of modulation in radio communication systems? 11. Explain the following terms as related to AM a. Depth of modulation b. Synchronous detection c. Diagonal clipping in demodulator.

Aim: Study of AM Transmitter and Receiver. Objective:
1. 2. 3.
Generate a DSB AM signal Generate a DSB-SC signal by removing the carrier component from the DSB AM signal To understand each and every block used in AM transmission and receptation
Equipments Required: Trainer Kit ST-2201 and ST-2202. Theory:

(1) The AM transmission Block consists of following functional blocks.. Audio Oscillator: This block comprises of monolithic function generator IC 8038, which generates sine wave output whose frequency can be adjusted from 300 Hz to 3.4 KHz by varying the frequency pot. This frequency will be used as our modulating signal, fed to audio amplifier balanced modulator & balanced modulator & band pass filter circuit. 455 KHz Oscillators: The circuit comprises of a coil and a transistor 2N3904 generates as frequency lightly less than 455khz and is used a carrier signal for a balanced modulator. 1MHz Crystal Oscillator: This crystal generate 1MHz frequency which is used as carrier for the two balanced modulators& band pass filter circuits, this is being amplitude modulated by the output from audio oscillator. Balanced Modulator: It generates two sidebands viz. upper sideband & lower side band with carrier, than it feeds back inverted variable carrier, which then generates an output, which contains only two side bands. Ceramic Band pass Filter: It is used to generate a SSB signal from the DSBSC signal obtained from the balanced modulator. Which is achieved by feeding the DSBSC signal to the input of ceramic band pass filter, whose band is so chosen as to pass only upper side band and reject lower side band. Output Amplifier: The input to this section may be a DSB or SSB signal depending on the position of mode switch. The output is an amplified signal, which will be transmitted via a cable or through antenna, which is selected.
(2) (3)
(4)
(5)
(6)
AMPLITUDE MODULATION (AM): The information signal should control the amplitude of the carrier wave. As the information signal increases in amplitude, the carrier wave is also made to increases in amplitude. Likewise, as the information signal decreases, then the carrier amplitude decreases.
Vmax Vmin Percentage modulation = ---------------------- X 100% Vmax +Vmin Main parts of the transmitter shown below; Information Signal Audio Oscillator Modulator Output Amplifier
Carrier Generator The Modulator: In this circuit the amplitude of the carrier is increased and decreased in sympathy with the incoming information signal. Information Signal
Modulator
AM Waveform
Carrier Wave Output Amplifier: This amplifier is used to increase the strength of the signal before being passed to the antenna for transmission.
DSB Receiver
The EM Wave from the transmitting antenna will travel to the receiving antenna carrying the information with it. The Receiving Antenna: It operates in the reverse mode to the transmitter antenna. The electromagnetic wave strikes the antenna and generates a small voltage in it. The Radio Frequency (RF) Amplifier: The first stage of amplification, which amplify the incoming signal above the level of the internally generated noise and also to start the process of selecting the wanted station and rejecting the unwanted ones. The Local Oscillator: An Oscillator producing a sinusoidal output similar to the carrier wave oscillator in the transmitter, however its output frequency is adjustable. It is always maintained at a frequency, which is higher, by a fixed amount, then the incoming RF signals, therefore it follows, or tracks, the RF amplifier frequency. Mixer: It performs a similar function to the modulator in transmitter. It combines the signal from the RF amplifier and the frequency input from the local oscillator to produce three frequencies: (i) A difference frequency of local oscillator frequency -RF signal frequency. (ii) A sum frequency equal to local oscillator frequency +RF signal frequency. (iii) A component at the local oscillator frequency.
From RF amplifier
Mixer
To IF amplifier
From local oscillator
Intermediate Frequency Amplifier (IF Amplifiers): It consists of two stages of amplification and provides the main signal amplification and selectivity, of this has removed the unwanted components generated by the mixing process. The Diode Detector: The function of the diode detector is to extract the audio signal from the signal at the output of the IF Amplifiers. It performs this task in a very similar way to a half wave rectifier converting an AC input to a DC output. Diode Input C R Output
The result is an output, which contains three components: (i) The wanted audio information signal. (ii) Some ripple at the IF frequency. (iii) A positive DC voltage level. The Audio Amplifier: at the input to the amplifier, a low pass filter is used to remove the IF ripple and a capacitor blocks the DC voltage level as shown in figure. The Automatic Gain Control (AGC): The AGC circuit is used to prevent very strong signals from overloading the receiver. It can also reduce the effect of fluctuations in the received signal strength.
Procedure:
Experiment 1 Double Sideband AM Generation This experiment investigates the generation of Double Sideband Suppressed Carrier (DSBSC) AM in such a way by removing the carrier from an AM waveforms. Procedure: Ensure the following conditions on ST-2201. 1. (a) AUDIO INPUT SWITCH INT. position (b) Select `DSB MODE`. (c) Output Amp. GAIN preset fully clockwise (d) Speaker Switch OFF position 2. Turn On power to the ST-2201 board 3. Turn the Audio Oscillator blocks Amplitude, Frequency and in BALANCED MODULATOR & BANDPASS FILTER CIRCUITS I block fully clockwise. 4. Monitor at t.p. 9 a sine wave frequency of 1MHz. This is the carrier input of our doublesideband modulator. 5. At t.p.3 the output of the BALANCED MODULATOR & BANDPASS FILTER CIRCUIT 1block is a double-sideband. AM waveform, which has been formed by amplitude-modulating the 1 Mhz carrier sine wave with the audio-frequency sine wave form the audio oscillator, by adjusting its preset we are removing the carrier component altogether means the carrier has been`balanced`out or suppressed to leave only the two sidebands. 6. Note that the DSBSC waveform appears, amplified slightly, at t.p. is the OUTPUT AMPLIFIERS output signal, which will be transmitted to the receiver 7. By using Audio Input Module, the human voice can be used as the modulating signal, instead of using ST-2201 Audio oscillator block. Connect the audio input module to the external audio input on the ST-2201 board, and put the Audio input select switch in the EXT position
Experiment 2
Double Sideband AM Reception This experiment investigates the reception and demodulation of AM waveforms. 1- Ensure the following conditions on ST-2201. (a) Audio oscillator Amplitude preset fully clockwise. (b) Audio input select switch in internal position. (c) Balance preset in BALANCED MOD & BPF CIRCUIT/block in fully clockwise. (d) Select DSB MODE. (e) O/P Amp. GAIN preset fully anti-clockwise. (f) TX output select switch in ANT position. (g) Audio amplifier VOLUME preset in counter clockwise. (h) SPEAKER switch in ON position (i) On board antennae in vertical position and fully extended 2- On ST-2202 (a) RX INPUT SELECT in ANT. position. (b) R.F.Amps.Tnd.Ckt.Swt.in ANT.psition. (c) R.F.Amps. GAIN preset fully clockwise. (d) AGC Switch IN position. (e) Detector Switch. in DIODE position. (f) Audio amplifier volume preset fully counter clockwise. (g) Speaker in ON position. (h) BEAT Frequency Oscillator Switch. In OFF position. (i) On board antennas in vertical position 3- Turn ON power to the modules 4- Now turn the GAIN preset in ST-2201 OUT AMPLIFIER block to its fully clockwise position, so that transmitter generates an AM signal 5- On the ST-2201/ ST-2202, adjust the volume preset so that receiver output clearly heard. Then adjust the receivers tunning dial, until the tone generated at the transmitters also clearly Audible at the receiver. (This should be when the tuning dial is set to about 55 65) 6- Now check the waveform at TP1 at ST-2201 and at TP39 at ST-2202. These two are approximately same Result: Precautions (if any): -
Aim: To measure the losses in an optical fiber communication link. Objective: (a) Propagation loss in the fiber (b) Bending loss Equipment Required: OFT, CRO, Function generator, 1 Hz 10 MHz Introduction: The fiber used in OFT is multimode plastic fiber with 1000 m core diameter. Unlike its Glass-Glass and Plastic Coated Silica fiber counterparts, this fiber has very low attenuation. This fiber has been selected for OFT because of the ease of handling it affords. Apart from the above propagation loss in a fiber, bending of fiber, connectors, splices and couplers may all contribute significantly to the losses in a fiber optic communication link. An Optical fiber is a circular waveguide. A small bend in a fiber will not significantly affect the propagation characteristics and therefore the losses in the fiber. However, if the fiber is bent with a radius of curvature smaller than a certain value, the propagating signal may suffer significant bending losses. Two optical fibres are joined using either a connector or a splice. The alignment of the cores of the two fibres is critical in both the situations, as even the minutes misalignment or gap between the fibres may cause significant coupling losses. Procedure: Set Up: 1. The interfaces used in the experiment should be as: a) At JP2, A1&B should be shorted b) At S6, A&B shortd c) At S26, A&B shorted The block diagram of the circuit used in this experiment is shown in Fig1 2. Set the switch SW8 to the ANALOG position and remove the shorting links S6 and S26. Attenuation at 850 nm: 3. Take the 1m fiber and set-up an analog link using LED1 and detector PD1
4. Drive a 1V p-p 10 KHz sinusoidal signal with zero d.c. at P11. 5. Observe the signal at P31 on the oscilloscope. 6. Adjust the GAIN such that the received signal is not saturated. Do not disturb the level of the signal at the function generator or the gain setting throughout the rest the experiment. 7. Note the peak value of the signal received at P31 and designate it as V1. Replace the 1m fiber by the 3m fiber between LED1 and PD1. 8. Note the peak value of the received signal and designate it as V3 . If is the attenuation in the fiber and l1 and l3 are the exact length of the 1m and 3m fibers in meters respectively, we have V3 = exp[ ( l 3 l1 ) ] V1 Where is in nepers/m. Compute in dB/m for 850 nm wavelength using = 4.343 . Bending Loss: 9. Repeat steps 3, 4 and 5 10. Now bend the fiber in a loop and reduce the diameter of the loop slowly and observe the reduction of the received signal at P31, and also note the diameter of loop. 11. Plot the amplitude of the received signal versus the diameter of the loop.
Result:
Precautions (if any):

Aim: To generate an FM signal and demodulate it. Equipments Required: FT-1502 Frequency Modulation and Demodulation kit, CRO. Theory: Frequency modulation is a system in which the amplitude of the modulation carrier is kept constant, while its frequency is varied as per the modulating signal. The modulation index is given by (FM) mf = Maximum frequency deviation Modulating Frequency or mf =
fm
Procedure: 1. Switch on the experiment kit. 2. Observe modulating signal having frequency of 8KHz and amplitude of 0-12VP-P. Carrier signal having frequency of 71kHz and amplitude is 5.6VP-P. 3. Connect Modulating signal to the modulator input as per figure given below and observe modulating signal and FM O/P on a dual trace CRO.
Modulating Signal Generator
FM Modulator
CH2 (CRO)
4. Adjust the amplitude of the modulating signal until we get undistorted FM output. And trace the signal ie. Modulating & modulated. 5. For demodulation connect the kit as per figure given below.
Modulating Signal Generator
CH1 (CRO)
FM Modulator
Demodulator
CH2 (CRO)
CH1 (CRO)
6.
Now decrease the amplitude of the modulating signal until we get undistorted demodulation output in this condition maximum signal generator output is VP-P is due to capture range restrictions of PLL in demodulator. Adjust the potentiometer in demodulation section until we get demodulated output. Trace it and compare with the input signal.
7. Result:
Precautions (If any):
Questions: 1. Compare FM system with AM system from bandwidth point of view. 2. With the help of a simple mathematical expression explain the basic difference between FM and PM. 3. What is meant by the following terms in connection with frequency modulation? a. Modulation index b. Frequency deviation 4. In frequency modulation for a given frequency deviation the modulation index varies. a. Inversely b. Directly c. Independently as the modulating frequency varies 5. In an FM waveform, the sidebands are spaced at interval equal to a. Twice the modulating frequency. b. Half the modulating frequency. c. Equal to modulating frequency. 6. In a system of modulation, increasing the depth of modulation increases the bandwidth of transmission, the system is a. AM b. FM c. PM 7. Indicate false statement, FM system a. Provides better noise immunity. b. Require lower bandwidth c. Require less modulating power 8. One of the following is an indirect way of generating FM this is a. Reactance tube modulator b. Armstrong modulator c. Varactor diode modulator 9. Pre-emphasis circuit in FM transmitter emphasises the a. Low frequency term b. Middle frequency term c. Both low and middle frequency terms d. High frequency terms 10. Demodulation of FM wave is effected by a. Envelope detector b. Discriminator c. Both (a.) or (b.) 11. Give the comparison between AM and FM systems.
Communication System Lab Aim: Study of FSK modulation and demodulation.

Objective: 1. To familiarisation with frequency shift keying method. 2. To observe the FSK modulator / demodulator output waveform.
Equipments Required: Trainer Kit FT-1506, CRO. Theory:

FSK is a system of frequency modulation in the nominal unmodulated carrier frequency corresponds to mark condition and a space is represented by a downward frequency shift. Fig. Shows the circuit diagram of a FSK modulation and demodulation system. Basically a 555IC is connected in astable multivibrator mode; generating a clock pulse of frequency determined by the values of RT and CT this clock signal is given by to a divided by 16 counter(74163) which generates divided by 2,4,8 and 16 outputs of the input clock signal. In this system divided by 2 and 8 output are given taken as two-carrier frequency, so these are given to a FSK modulator constructed by using NAND gates. Divided by 16 output is given to a decode counter (IC-7490) which generate the modulating data signals, so depending on the level of the modulating data signal given to the FSK modulator either divided by 2 or divided by 8 frequency output of the IC 74163 and transmitted to the output of the FSK modulator. In the demodulator section high-Q turned filter is used which is tuned to any frequency divided by 2 or 8 so the filter passes one frequency and stops the other frequency than we are using a envelop detector and comparator, its output is equivalent to the modulating data given at the input of the FSK modulator.
Procedure:
1. Switch on the experimental board. 2. Apply any one data output of the decade counter (IC-7490) to the data input point of FSK modulator and observe the same signal in one channel of a dual trace oscilloscope. 3. Observe the output of the FSK modulator on the second channel of the CRO. 4. During the demodulation connect the FSK output to the input of the demodulator. 5. Adjust the potentiometer P1 and P2 until we get the demodulated O/P equivalent to the modulating data signal. Result: Precautions (if any):

Aim: Measurement of Numerical Aperture of a Fiber using Optical Fiber Trainer Kit. Equipments Required: ST-2502 WB-Trainer, Numerical aperture Jig. Theory: Numerical aperture refers to the maximum angle at which the light incident on the fibre end is totally internally reflected and is transmitted properly along the fibre. It shows the light collection efficiency of the fiber. Its maximum value may be one. The cone formed by the rotation of this angle along the axis of the fibre is the cone of acceptance of the fibre. The light ray should strike the fibre end within its cone of acceptance else it is refracted out of the fibre. It is very important that the optical source should be properly aligned with cable and the distance from the launched point & cable be properly selected to ensure that the maximum amount of optical power is transferred to the cable. Procedure: (1) Connect power supply to the board. (2) Connect the frequency generators 1KHz sine wave output to input of emitter-1 Circuit. Adjust its amplitude at 5Vp-p(e.g.). (3) Connect one end of fiber cable to the output socket of emitter-1 circuit and the other end to the Numerical aperture measurement jig. Hold the white screen facing the fibre such that its cut face is perpendicular to the axis of the fibre. (4) Hold the white screen with 4 concentric circles (10,15, 20 & 25mm diameter) vertically at a suitable distance to make the red spot from the fibre coincide with 10mm circle.
Screen F.O.Cable W L Scale W
(5) Record the distances of screen from the fiber end L and note the diameter W of the spot. (6) Compute the numerical aperture from the formula given below: W 4 L2 + W 2
N .A =
= Sin max
(7) Vary the distance between the screen and fiber optic cable and make it coincide with one of the concentric circles and note its distance. (8) Tabulate the various distances and diameter of the circles made on the white screen and computes the numerical aperture from the formula given above. The N.A. recorded in the manufacturers data sheet is 0.5.
Result:

Aim: Study of an optical fiber kit and setting up of an analog/digital link. Equipment Required: ST2502WB Trainer Kit. Objective (a): To study an 650nm fiber optic analog link. Theory: Fiber optic links can be used for transmission of digital as well as analog signals. Basically a fiber optics link contains three main elements, a transmitter, an optical fiber and a receiver. The transmitter module takes the input signal in electrical form and then transforms it into optical (light) energy containing the same information. The optical fiber is the medium, which takes the energy to the receiver. At the receiver light is converted back into electrical form with the same pattern as originally fed to the transmitter. Transmitter: Fiber optic transmitters are typically composed of a buffer driver and optical source. The buffer provides both an electrical connection and isolation between the transmitter & the electrical system supplying the data. The driver provides electrical power to optical source. Finally the optical source converts the electrical current to the light energy with same pattern. Commonly used optical source are light emitting diodes and a laser beam. The transmitter section comprises of: (1) Function Generator. (2) Frequency Modulator & Pulse width modulator block. (3) The function generator generates the input signals that are going to be used as information to transmit through the fiber optic link. The output voltage available is 1KHz sinusoidal signal of adjustable amplitude and fixed amplitude 1KHz square wave signal. The modulator section accepts the information signal and converts it into suitable form for transmission through the fiber optic link. The fiber optic link: Emitter and detector circuit on board form the fiber optic link. This section provides the light source for the optic fiber and the light detector at the far end of the fiber optic links. The optics plugs into the connector provided in this part of the board. Two separate links are provided.
The receiver: The comparator circuit, low pass filter, phase Locked Loop, AC Amplifier Circuits form receiver on the board. It is able to undo modulation process in order to recover the original information signal. In this experiment the trainer board is used to illustrate one-way communication between digital transmitter and receiver circuits. Procedure: 1. Connect the power supply to the board. 2. Ensure that all switch faults are off. 3. Make the following connection a. Connect the F.G 1KHz sine wave output to emitter 1s input. b. Connect the Optical Fiber cable between emitter output and detectors input. c. Detector 1s output to Amplifier 1 input. 4. Switch emitter 1s driver to Analog mode. the two signals are same. Objective (b): The objective of this experiment is to study an 650nm fiber optic Digital link. Procedure: 1. Make the following connection (as shown in diagram-2) 2. Connect the F.G. 1 KHz square wave output to emitter 1s input. 3. Connect the fiber optic cable between emitter output and detectors input. 4. Detector1s output to comparator 1s input. 5. Comparator 1s output to Amplifier 1s input. 6. Switch emitter 1s driver to digital mode. 7. Switch ON the power. 8. Monitor both the inputs to comparator 1 (t.p.13 &14). Slowly adjust the comparators bias preset, until DC level on the input (t.p.13) lies mid way between the high and low level of the signal on the positive input (t.p.14). 9. Observe the input to emitter 1 (t.p.5) with the output from Amplifier 1(t.p.28) and note that the two signals are same. 5. Observe the input to emitter (t.p.5) with the output from amplifier 1(t.p.28) and note that
Result: Precautions (if any):
Communication System Lab Aim: Study of Pulse Code Modulation and Demodulation.
Objective: a. b. c. d. To observe multiplexed (TDM) signals To generate PCM wave and observe intermediate signals. To demodulate the PCM signal. To observe some of error correcting codes.
Equipment Required: Trainer kit ST2103 and ST2104, CRO. Introduction: The basis of digital modulation system lies on pulse modulation i.e. a particular characteristic of pulse is varied in accordance with the information signals. In PCM system the amplitude of the sampled waveform at define time intervals is represented as a binary code. The pulse code modulation/demodulation comprises of the following steps Sampling Quantisation Encoding Decoding Reconstruction filter
Coding allows us a great deal of detection and correction it generally cannot detect or correct all errors. Detection of errors allows the system to request the retransmission of data, but it doesnt solve the problem. A better solution would be to introduce a method of error detection and correction. Many different types of codes have been developed and are in use, commonly employed codes are Parity coding Hamming coding
Parity Coding: It is its simplest method of error coding. Parity is a method of encoding such that the number of ones in a codeword is either even or odd. If even parity is to be establish a 1 bit is added to each code word containing odd 1 and a 0 bit is added to each word containing even number of ones. Similarly the parity coding can ensure that the total numbers of ones in a encoded words is odd. In such cases it is called as odd parity. Parity coding is normally only used on transmission systems where the probability of error occurring is low.
Hamming Coding: Hamming coding recode each word at transmitter in two way new code by stuffing the word with extra redundant bits. Three bits Hamming code provide single bit error detection and correction. Therefore only four bits are used for transmitting data, if Hamming code is selected. For transmitting data than the format becomes D6, D5, D4, D3, C2, C1, C0 where C2, C1 and C0 are Hamming code bits. The code on this trainer is generated by adding parity check bit to each group as shown below Group1 Group2 Group3 D6, D5, D4 D6, D5, D3 D6, D4, D3 Parity Bit-C2 Parity BitC1 Parity Bit-C0
The groups and parity bits forms an even parity check group. If an error occurs in any of the digits, the parity is lost and can be detected at receiver e.g. Let us encode binary value D6, D5, D4, D3 of 1101 Group1 Group2 Group3 D6 1 D6 1 D6 1 D5 1 D5 1 D4 0 D4 0 D3 1 D3 1 C2 0 C1 1 C0 0
So, the data word after coding will be D6 D5 D4 D3 1 1 0 1
C2 0
C1 1
C0 0
At the receiver, the four digits representing a particular quantise value are taken in as three groups. The error detection/ Correction logic carries out parity checks on the three groups Group1 Group2 Group3 D6 D6 D6 D5 D5 D4 D4 D3 D3 C2 C1 C0
If none of them fails, than no errors has occurred in transmission and all bit values are valid. Suppose a case, where the following parity check was carried out and the listed groups failed. Group1 D6 D5 D4 C2 0 1 0 0 Failed Group2 D6 D5 D3 C1 0 1 1 1 Failed Group3 D6 D4 D3 C0 0 0 1 1 Passed
If we suppose only a single bit error the passing of group three means that all D6, D4, D3 and C0 are valid. Table given below gives the location of possible single bit errors.
Parity Check Results on ST2104 Trainer

Group1 D6D5D4C2 Pass Pass Pass Pass Fail Fail Fail Fail Group2 D6D5D3C1 Pass Pass Fail Fail Pass Pass Fail Fail Group3 D6D4D3C0 Pass Fail Pass Fail Pass Fail Pass Fail Location of Error No Error C0 C1 D3 C2 D4 D5 D6
Procedure: 1. Set up the following initial conditions on ST2103 Trainer. a. MODE switch in FAST position. b. DC1 & DC2 amplitude controls in function generator block in fully clockwise position. c. 1kHz & 2kHz signal levels in function generator block set to 1Vpp. d. PSEUDO RANDOM Sync code generator switched on. e. Error check code selector switches A&B in A=0&B=0 Position (OFF Mode) f. All switched faults off. 2. Set up following initial conditions on ST2104 Trainer. a. MODE switch in FAST position. b. PSEUDO RANDOM Sync code generator switched on c. Error check code selector switches A&B in A=0&B=0 Position (OFF Mode) d. All switched faults off. e. Pulse generator delay adjust control in fully clockwise position.
3. Make following connections on ST2103 Trainer (Diagram-1) a. DC1 output to CH.0 input (t.p.10) b. CH.0 input (t.p.10) to CH1 input (t.p.12) This ensures that the two channels contains the same information 4. Make following connection on ST2104 Trainer (Diagram-1) a. PCM data input (t.p.1) to clock regeneration circuit input (t.p.3) b. Output of clock regeneration circuit (t.p.8) to RX.clock input (t.p.46) 5. Make following connection between ST2103 and ST2104 (Fig.1)
a. PCM output (t.p.44) of ST2103 to PCM data input (t.p.1) of ST2104 b. Connect the grounds of both the trainers. 6. Turn on the power. Ensure that the frequency of the VCO(t.p.44) in the receiver clock regeneration circuit has been correctly adjusted. When it is correctly adjusted than LED of sync bit counter will turn on. 7. Connect channel one of CRO to t.p.10 on ST2103, Channel two of CRO to t.p.33 on ST2104 By varying DC1 we can verify data is transferred correctly between two trainers, we can verify that the data in the A/D converter block of ST2103 Trainer is always same as the data in D/A converter block of ST2104. 8. Even Parity: select even parity with error check code selector switches A&B at A=0 & B=1 position, on both the trainers. a. Set up various codes from A/D converters output LEDs some containing even number of 1s & some odd. b. Check the error check code generator output on ST2103 Trainer, data latch output (t.p. 16 to 22) on ST2104 trainer & D/A converter input (t.p. 23 to 29) on ST2104 trainer. c. Notice the number of 1s in the transmitted data streams is it even or odd? d. ST2103 uses the least significant bit (LSB) of the 7 bit word to transmit the parity bit. Its value is changed to achieve the correct parity for each word. 9. Odd Parity: set up the error check selector A & B switches to A=1 & B =0 position on both Trainers to select the odd parity mode. Carry out steps 8&9 again, but odd parity selected this time. 10. Carry out the same experiment with 1kHz sine wave applied at CH.0 & CH.1 input of ST2103 transmitter Trainer.
11. Hamming code: -The position of A and B switches in error check code selector block is A=1 and B=1. 12. Switch ON the Hamming code error check mode on Trainer, connect DC1 output from function generator block to CH.0 and CH.1 input of the ST2103 Trainer. 13. Turn ON the power. Adjust the DC1 control such that such that the A/D converter blocks LEDs shows a data of 1101000 on D6-D0 Bits record the binary code on the error check code generator LEDs. Match this with your expected code. 14. Vary the DC1 control and not the error check code generator output.
15. Error checks code generator is only concerned with the Bits D6, D5, D4 and D3.Parity check bits C2, C1 and C0 depends on the value of bits D6, D5, D4 and D3. 16. Work out a table as shown below with different values at D6, D5, D4 and D3 Bits
S.No Data at A/D Converters Output Expected Results Check on ST2103 Trainer
D6 D5 D4 D3 1 2 3
C2 C1 C0
17. Assume that the error detection/correction logic has received a transmitted seven Bit word as shown in table below. Work out the error Bit (if any ) and than correct the data used output bits D6, D5, D4 and D3 were originally at the output by the ST2103 Trainer. S.No Data Received 1 2 3 4 0101011 0110001 1101101 1101001 Case1 D6D5D4 C2 Fail Case2 D6D5D3 C1 Fail Case3 D6D4D3 C0 Pass Bit in error D5 Corrected output 0001011
Note: How the particulate combination of passes or failures of parity check locates a single faulty Bit. 18. Set the switch position A=0 and B=0 position. Connect 2kHz signal from the function generator block to CH0 and CH1 input of ST2103 Trainer and repeat the same process.
Results:
Questions: 1. What are the advantages and applications of PCM? 2. Name the digital modulation systems other than PCM? 3. Draw an irregular waveform and show how it is quantised, using eight standard levels. 4. In what way PCM is different from other modulation systems, pulse or analog? 5. What makes PCM a digital system?
6. As compared to message bandwidth the PCM bandwidth is a. Much smaller b. Same c. Much larger 7. Compander performs a. Compression before modulation b. Expansion after demodulation c. Both of the above d. None of (a) and (b). 8. The quantisation error is a function of a. Intervals between levels b. Signal amplitude c. Both of these. 9. In PCM generation, the sampling of the signal tends to a. PWM b. PPM c. PAM d. PCM 10. The quantisation error changes with the number (n) of quantisation levels as a. n b. n2 c. 1/n d 1/n2

Aim: Generation and Demodulation of Pulse Position Modulation Signals. Equipments Required: Trainer kit FT1505, CRO Circuit Diagram: Introduction: Pulse Modulation may be used to transmit analog information. It is a system in which continuous signals are sampled at regular intervals. Information regarding the signal is transmitted only pulses. At the receiving end, the original signal may be reconstituted from the received samples. In Pulse Position Modulation, we have fixed amplitude of each pulse, but the position of each pulse is shifted, the shift being proportional to the amplitude of the modulating signal at that instant. Procedure: 1. Switch on the experimental kit. 2. Observe the clock generator output and modulating signal output. 3. Connect the clock generator output to the clock input point and modulator output to modulation input point of PPM modulator and observe the same clock on one channel of a dual trace CRO 4. Observe the PPM output on CRO 5. By varying the modulating voltage, PPM output clock position changed, but its width maintains constant. 6. If we observe the PWM output, its width varies according to the modulating voltage. 7. During the demodulation, apply PPM signal to the input of demodulator and observe its output. 8. Output of the demodulator is almost concides with the modulation signal but having same phase difference due to RC networks and amplifier, which are in the demodulator.
Results:
Questions:
1. What is the fundamental difference between pulse modulation, on the one hand and frequency and amplitude modulation on the other? 2. It is necessary to transmit a series sync pulse in a. PAM b. PDM c. PPM 3. In essence, practical sample is a. PDM b. PPMc. PAM 4. The pulse width modulation may be generated a. By differentiating pulse position modulation b. With a monostable multivibrator c. By integrating the signal 5. Define and describe pulse position modulation and explain with waveforms, how it is derived from PWM. 6. If the synchronisation between transmitter and receiver fails, which of the following pulse systems would be affected? a. PAM b. PDM c. PPM 7. In PPM, the message resides in a. Pulses b. Time location of Pulses edges 8. Which performs best in the presence of noise? a. PAM b. PWM c. PPM c. None of these

Aim: Study of PSK Modulation/Demodulation Objective: 1. To familiarisation with phase shift keying method. 2. To observe the PSK modulator/demodulator output waveform. Equipment Required: Trainer kit FT-1507, CRO. Theory: Modulation is defined as the process by which some characteristics of a carrier is varied in accordance with a modulating signal. In digital modulating wave consist of binary data or an M-ary encoded version of it with a sinusoidal carrier, the feature i.e used by modulator to distinguish one signal from another is a step change in amplitude, frequency or the phase of the carrier.The result of this modulation process is amplitude shift keying(ASK), frequency shift keying(FSK) or phase shift keying(PSK), respectively. Ideally, PSK and FSK signals have a constant envelop. In PSK the carrier may be phase shifted by +ive90 for a mark, and by ive90 for a space. (Fig-1) shows the circuit diagram of PSK modulator and demodulator. IC-8038 is abasic waveform generator, which generates Sine, Triangle and Square waveforms. Sine wave generated is used as carrier signal to the system. Square wave is used as clock input to a decade counter (IC-7490), which generate modulating data output. IC-CD4051 is a multiplexer to which carrier is applied with and without phase shift. Modulating data input is applied to its control input. The 180 phase shift to the carrier is created by an operational amplifier (IC-741). During the demodulation the PSK signal is converted into square wave signal and is applied to one input of an EX-OR gate, on second input of gate carrier signal is applied, so the EX-OR gate output is equivalent to the modulating data signal. Procedure: 1. Switch ON the experimental board. 2. Apply the carrier signal to the input of the modulator. channel-1 of CRO. 4. Observe the output of PSK modulator on the channel-2 of the CRO. 5. Apply this PSK output to the demodulator input and also apply the carrier input. 6. Observe the demodulator output and compare it with the modulating data signal to applied the modulator input. 3. Apply the modulating data signal to the modulator input and observe the signal on
Questions:1. By using non coherent detection complexity of the receiver. a. Increase b. Decrease c. Decrease but ate the expense of an interior d. Both (b) and (c) error performance 2. Mark true or false:Some times both amplitude and phase of the carrier are combined to produce amplitude phase keying (APK). 3. For transmission of normal speech signal the PCM channel leads a bandwidth of a. 64 kHz bandwidth of a. fm b. 2 fm 3. n fm Hz b. PCM c. All modulation system. 5. Quantising noise is produced in a. All pulse modulation system b. 8 kHz c. 4 kHz 4. To transmit n signals each band limits to fm Hz by TDM will require a minimum

Aim: Generation and Demodulation of Pulse Width Modulation Signal. Equipments Required: Trainer kit FT1504, CRO Circuit Diagram: Introduction: Pulse Modulation may be used to transmit analog information. It is a system in which continuous signals are sampled at regular intervals. Information regarding the signal is transmitted only pulses. At the receiving end, the original signal may be reconstituted from the received samples. In pulse width modulation, we have fixed amplitude and starting time of each pulse, but the width of each pulse is made proportion to the amplitude of the modulating signal at that instant. The width of each pulse is varied if we change the amplitude of the modulating signal. Procedure: 1. Switch on the experimental kit. 2. Observe the clock generator and modulating signal outputs working properly. 3. Connect clock generator output to the clock input point of the PWM modulator and observe the same clock on the CRO. 4. Observe the PWM output on CRO at pin no.3 of IC555. 5. If we observe the PWM output, its width varies according to the modulating voltages. 6. During the demodulation apply PWM signal to the input of demodulator and observe its output. 7. Output of the demodulator almost coincides with the modulating signal but having some phase difference due to RC networks and amplifiers which are in the demodulator.
Results:
Questions:
1. What is pulse width modulation? What other names does it have? 2. Indicate which of the following pulse modulation system is analog? a. PCM b. Differential PCM c. PWM d. Delta 3. Pulse width modulation may be generated a. By differentiating PPM b. With a monostable multivibrator c. By integrating the signal 4. Which of the following pulse systems require highest bandwidth? a. PAM b. PDM c. PPM 5. The major application of PWM is in a. TDM b. Detection of PPM c. Generation of PPM d. PAM
6. For nearly distortion less receives signal in pulse modulation, it is required that the speed should be a. Less then the signal frequencyb. Less then twice the signal frequency c. More then twice the signal frequency

Aim: Sampling and reconstruction of an analog signal.
Objectives:
a. Study of sampling and reconstruction. b. To verify the Nyquist rate and see the Aliasing effect. c. To observe the effect of duty cycle on the reconstruction waveform in Sample and Hold output. Equipment required: Trainer kit ST2101, CRO, Function Generator
Introduction:
The signal, which contains the information to be transmitted, is known as information signal. Our aim is to reproduce this information signal as accurately as possible at the receiving end of the communication system. Sampling can be defined as measuring the value of an information signal at predetermined time instants. The rate at which the signal is sampled is known as the sampling rate or sampling frequency. If the signal is sampled quite frequently (The limit being specified by Nyquist rate), then it can be reproduced exactly at the receiver with no distortion. Procedure: 1. Set that the Ext.int sampling selector switch in internal position. 2. Put the duty cycle selector switch in position 5. This gives a duty cycle of 0.5. 3. Link 1 kHz sinewave output to analog input. 4. Turn on the Trainer (Power ON). 5. Select 32 kHz sampling rate. 6. Display 1 kHz sinewave (t.p.12) and sample output (t.p.37) on CRO. The display shows 1 kHz sinewave being sampled at 32 kHz, so that there are 32 samples for every cycle of the sinewave (Fig.1). 7. Link the sample output to fourth order low pass filter. Display sample output (t.p.37) and the output of filter (t.p.46) on the CRO. The display shows the reconstructed original 1 kHz sinewave (fig.2). Repeat with the filters of 2nd order and observe the difference. 8. By successively pressing of frequency selector switch, change the sampling frequency to 2 kHz, 4 kHz, 8 kHz, 16 kHz and back to 32 kHz (Sampling frequency is 10% of the frequency indicated by the illuminated LED). 9. Observe how sampled output changes in each case and how the lower sampling frequencies introduce distortion in the reconstructed output. 10. So far, we have used sampling frequencies greater than twice the maximum input frequency. To study Nyquist criteria, set sampling rate of 8 kHz, 50% duty cycle. 11. Remove the link from 1 kHz sinewave output to the signal input.
12. Obtain a 2 Volt peak, 2 kHz sinewave from 50 output of the function generator to signal input. 13. Observe the waveform at signal input and fourth order LPF filter (t.p.46). 14. Decrease the sampling rate to 32 kHz and then to 2 kHz. Observe the distorted waveform at the filter output (t.p.46). This is due to the fact that we under sampled the input waveform overlooking the Nyquist criteria and thus the output was distorted even though the signal lie below the cut off frequency of the filter. This also describes the phenomenon of Aliasing. 15. To see the effect on reconstructed waveform of the use of sample/hold circuitry and effect of sampling pulse duty cycle on the reconstructed waveform in sample and sample/hold output. Process is as given below. a. Follow steps 1-5 as described above. b. Observe the sample output (t.p.37) and the fourth order LPF filter output (t.p.46). c. Vary the position of duty cycle selector switch from 0% to 90%, observe how the sample output changes and how the amplitude of filter output changes. d. Disconnect the sample output from filter input. e. Link sample and hold output to fourth order low pass filter input. Set the duty cycle selector switch to position 5 (fig.3). f. Observe the waveform at sampled and hold output (t.p.39) on CRO. Vary the sampling frequency to illustrate how each sample is held at the sample/ hold output. g. Observe the filter output at t.p.46 (fig.4).
Results:
Questions:
1. What is the Nyquist sampling rate of a given signal? 2. The minimum sampling frequency is called a. Carlson Frequency b. Baud rate c. Nyquist sampling rate d. Bit rate 3. Aliasing can be reduced by a. Improving the filter slope characteristic b. Increasing the sampling frequency c. Both of these 4. If the lower sideband overlaps the baseband, distortion is called a. Cross over distortion b. Slope overload distortion c. Cross-Talk d. Aliasing 5. By increasing the sampling frequency, the bandwidth requirement of the transmission medium a. Increases b. Remains same c. Decreases 6. What will be the effect of over sampling on the bandwidth? 7. Under sampling will lead to some problems, discuss. 8. Sampling theorem find application in a. b. c. d. AM FM PCM None of these.

Aim: -\ Study of TDM Pulse Amplitude Modulation/Demodulation
Objective:
1. 2. 3. 4. To observe individual PAM signals. To observe multiplexed (TDM) signals. To study sync and control signals. To observe synchronisation requirement in time division multiplexing
Equipment Required: Trainer kit ST2102, CRO 30MHz Circuit Diagram: Procedure: The circuit, which ensures precisely timed action, is the clock circuit for receiver clock must match with the transmitter clock. Besides clock signal, the receiver also requires information from the transmitter to identify one timeslot per frame. It is accomplished by frame synchronisation signal. These two signals namely the clock signal and frame synchronisation signal may be transmitted by the transmitter along with the information signal. The most vital requirement of a time division multiplexed system is synchronization. We can use three different modes of these information transfers. Clock signal and frame synchronisation signal should be transmitted by the transmitter along with the information signal. The receive clock must match with the transmitter clock, the receiver also required an information from the transmitter to identify one time slot per frame and so as to pass the time slot to correct output channel. This ensured by frame synchronisation signal. Mode1: Three links between transmitter and receiver (Fig.1) In this mode a separate transmission media are used to carry the information signal the clock signal and the frame synchronisation signal. Mode2: Two links between transmitter and receiver (Fig.2) The number of links between transmitter and receiver can be reduced to two, by extracting the clock information from the frame synchronisation signal. Mode3: One link between transmitter and receiver (Fig.3) The number of links connecting transmitter and receiver can be reduced to one the reduction in number of links is achieved by using one time slot to transmit synchronisation signal along with the information samples. The word Link refers to one set of dedicated connections.
Synchronisation signal must be different from the information samples for the receiver to distinguish it from the other samples. The distinction is achieved by fixing the amplitude of the synchronisation level samples considerably greater than maximum information signal amplitude. We are using de-multiplexer to be able to distinguish the synchronisation signal from the information samples. 1. Set the duty cycle control switch in position 5. 2. Turn the all potentiometers in function generator block viz., synk level, 250 Hz, 1KHz, 2 KHz fully clockwise. 3. Turn the potentiometer mark comparator threshold level in phase locked loop timing logic block fully clockwise. 4. Ensure that the delay control port in receiver timing logic block is fully anticlockwise. 5. Make following connections a. 250Hz to CH.0 input socket of Transmitter block b. 500Hz to CH.1 input socket of Transmitter block c. 1KHz to CH.2 input socket of transmitter block d. 2KHz to CH.3 input socket of Transmitter block 6. Turn on the power to the Trainer; The Transmitter circuit samples all channels at different time intervals. Time division multiplexed samples appear at the Tx. Out (t.p.20). 7. Make the following connections a. Tx. Output to Rx. Input b. Tx. Clock to Rx. Clock c. Tx. CH.0 to Rx. CH.0 8. This ensures Mode1 operation of the ST2102 trainer (See Diagram 1 for interconnections) Tx. Clock signal is used by the receiver to synchronise its activity & TX.CH0 signal is used by the receiver to know which sample belongs to channel 0. 9. With the help of oscilloscope, observe the Tx. Output signal (t.p.20) & the Receivers CH.0 Low Pass Filter input (t.p.41). The Oscilloscope displays the extracted sample corresponding to channel 0 from the time division multiplexed sample. 10. Display the Receivers Low Pass Filters input (t.p.41) & output (t.p.42) simultaneously on the oscilloscope. The signal at t.p. 42 shows the reconstructed ~ 250 Hz. sine wave which was transmitted at CH.0. Similarly view the outputs of all Receiver Low Pass Filters in turn (t.p.44, 46,48). Observe that each of the original sine wave have been correctly reconstructed. 11. The three links required between Transmitter & Receiver in Mode 1 of operation can be reduced to two in Mode2. To configure the trainer in Mode 2 of operation, discount TX.CH.0 & RX.CH.0 link & make following new links.
a. Tx. CH.0 to PLL I/P b. SYNC to RX.CH.0 c. PLL O/P to Rx clock Also ensure that level of toggle switch in Phase Locked Loop Timing Logic is in upward position. The configuration is given in Diagram 2. 12. The Phase Locked Loop locks onto the Tx.CH.0 signal & produces two outputs. a. SYNC. This serves the same purpose as TX.CH.0 RX.CH.0 link i.e. it tells the Receiver which of the transmitted signal belongs to channel 0. b. CLK It is used to clock the receiver. These signals can be examined on t.p. 28 & 26 respectively. 13. Observe the Receivers CH.0, CH.1, CH.2, CH.3 output on the oscilloscope (t.p.42, 44, 46, 48). Notice that the wave shapes are still preserved in Mode2. 14. The number of links can be further reduced to one in Mode3. In this operational mode, the sync pulse are transmitted along with the other samples in channel 0 time slot i.e. channel 0 is dedicated to carry sync pulses. To configure the trainer in Mode3 of operation, remove following links. a. ~250 Hz. To transmitters CH.0 Input. b. TX.CH.0 to PLL I/P socket. Now establish the following connections, a. SYNC Level in Function Generator Block to Transmitters CH.0 Input. b. Ensure that the toggle switch lever in Phase Locked Loop Timing Logic is in Upward Position. The configuration is as shown in Diagram 3. 15. Display Tx. Output & Transmitters CH.1 input (t.p.13) on the oscilloscope. 16. Sync Level Pot should be to fully clockwise position. These sync pulses are fed to the voltage comparator, which extracts the sync pulses. The threshold level of the comparator has been set such that it can easily distinguish between sync pulses & the other samples. These pulses are fed to the Phase Locked Loop Circuitry, which locks on to the sync pulse, and generates sync clock signal as in Mode 2. 17. Observe the Receivers CH.1, CH.2 & CH.3 Outputs (t.p. 44, 46, 48) on oscilloscope. Notice that the waveshapes still preserved in Mode3. Receiver CH.0 output (t.p.42) is a D.C. level related to the amplitude of the Transmitted sync pulses. Block
Results: Precautions (If any):
Questions: 2. Explain the principle of time division multiplexing. 3. What is multiplexing? Why is it needed? 4. What are the two basic forms of multiplexing? 5. State sampling theorem. 6. Quantisation noise occurs in a. TDM b. FDM c. PCM d. PWM 7. What will happen if clocks at Tx and Rx are different? 8. What will happen if synchronisation gets lost? 9. Is sampling a must for TDM technique? 10. How do you demodulate a PAM signal? 11. Can TDM be accomplished with our pulse modulation systems like PPM and PWM? 12. Will TDM work satisfactorily if the signals are under sampled?
III YEAR Electronics & Communication Communication System Lab-II
Aim: Harmonic analysis of a given signal using Spectrum Analyzer. Objective:

1. 2. 3. 4. 5. To understand the working of spectrum analyser. To measure frequency and level of a given unknown signal. To measure Harmonics of square wave and Triangular wave. To check the frequency response of a low pass filter. To check the frequency response of a high pass filter. Spectrum analyser, Function Generator and CRO
Equipments Required: Introduction:

The spectrum analyzer permits the detection of spectrum components of electrical signals in the frequency range of 0.15 to 1050MHz. In contrast to an oscilloscope where the amplitude is displayed on the time domain, the spectrum analyzer displays amplitude on the frequency domain. The individual spectrum components of a signal become visible on a spectrum analyzer. The oscilloscope would display the same signal as one resulting waveform. The HM5014 includes a tracking generator. This generator provides sine wave within the frequency range of 0.15 to 1050MHz. Spectrum analyzer and tracking generator are synchronized. Controls (Fig.1) (1) Power: After about 10 second the noise level will appear on the bottom base line. (2) INTENS: Beam intensity adjustment. (3) FOCUS: Beam sharpness adjustment. (4) A/B/A-B: The instrument has two memories, memory A and memory B. Actual measurement results are always stored in A, whereby memory B can only accept copies of memory A results, Function A-B allows for the subtraction of B results from updated measuring results stored in A. (5) AB: Allows for temporary storage of settings from memory A to memory B for comparison purposes. Push short to store the actual contents of A in B. the instrument will automatically display stored B memory. To get back to the actual signal, push, A/B/AB two times. If pushed one time, AB will be displayed. Memory contents of B will be deleted when power is turned off. (6) AVERAGE: This function allows the automatic storage of average signal level readings of the instruments. Using the AVERAGE function the displayed noise band can be reduced. Pushing the AVERAGE button for a short time activates the AVERAGE function. The respective LED will light to show function is activated. (7) CENTER FREQUENCY:
By pushing CENTER FREQ. Button, input for center frequency is being enabled and respective LED is lit. Now Center Frequency can be adjusted via tuning knob (14). The frequency is displayed in the upper left-hand corner behind the letter C.
(8) FINE: If the FINE button is pushed (LED is lit), a Marker movement is performed in very small steps. (9) MARKER: In order to evaluate measurement curves, the instrument is equipped with a running Marker (X). The Marker can be moved in X-orientation via the tuning dial and follows the measurement curve in Y-orientation. The Marker is activated (LED lit) by pushing MARKER button . The numeric indication of marker frequency and amplitude is displayed on-screen. Push CENTER to leave MARKER mode. (10) Tuning Dial: The tuning dial either selects Center Frequency or Marker position, depending on CENTER FREQUENCY or MARKER being activated. SPAN: The span of sweep of the analyzer is set via the two SPAN buttons. The SPAN is displayed in the upper right-hand corner of the screen and is marked with the letter S. At full SPAN (1000MHz) the frequency axis is scaled in 100MHz step per (vertical) graticule line. The frequency reduces accordingly when moving towards the left screen edge. ZERO SPAN: With the ZERO SPAN button a SPAN of 0 Hz is being selected. Pushing the ZERO SPAN button activates the ZERO SPAN mode. Press again to revert to previous span. Select the precise Center Frequency and the SPAN as low as possible (resolution of display) so the signal can be viewed easily. 5dB/Div. Py pushing the 5 dB/Div. Button, the vertical scale is set to 5 dB/Div. The respective LED will then be lit. To enter again in 10 dB/Div. Mode, push 5 dB/Div. again. (13) RBW (Resolution Bandwidth): The instrument is equipped with resolution filters of 9KHz, 120KHz, and 400KHz, which can be selected via the RBW buttons. The respective LED will indicate which resolution bandwidth is selected. (14) VBW-Video Filter:
(11)
(12)
The Video Filter has the purpose of reducing video and noise bandwidth and therefore reducing noise distribution. When measuring low-level values, which are situated within the regular noise level, the Video Filter (low pass) can be used to reduce the noise level. This allows for the detection of even small signals, which might otherwise not be visible. (15) ATTN:
The buttons to set the input attenuation are marked ATTN. By pushing the UP and DOWN buttons, the attenuation can be set from 0 db to 40 dB in 10 dB steps. To
enter 0 dB attenuation, it is necessary to push button long, this is for protection of the input stage, and so that this setting cant be set accidentally. (16) REFERENCE With rotary knob REFERENCE the so-called reference level is set, to this level all amplitude readings on screen are referenced. The reference level is always at the topmost horizontal graticule line. Depending also on attenuator setting, the reference level may be set between 99.8 dBm and +13 dBm.
(17)
INPUT:
Without attenuation of input signal the maximum allowable input voltage is +25V DC or +10dBm AC respectively. With a maximum attenuation of the input signal (40dB) +20dBm is allowed. These values should not be exceeded; otherwise the input stages will be damaged. (18) ATTN. The Track Generator output attenuator HM5014 has 5 positions, which can be chosen via the UP/DOWN buttons. The output attenuator is used for reducing the output level of the Tracking Generator. (19) LEVEL With the LEVEL knob the output level of the Tracking generator can be varied in steps of 0.2dB. The output level is shown in the readout, also dependent on the attenuator setting. (20) Tracking Generator After switching on the instrument, the Tracking generator will be inactive. This is a security measure to protect any loads connected. In the readout this is shown by the t. By pushing TRACK GEN. The Tracking Generator is activated. Now a T will appear in the readout, and one of the attenuator LEDs (26) will be lit. By again pushing TRACK GEN., the Tracking Generator is deactivated. (21) Output
50 output of Tracking Generators. The output level is set using the rotary knob LEVEL (27) and the attenuator buttons (26). If the noise band moves upward on screen when decreasing input attenuation, this indicates a possible other high-amplitude input signal present but not visible in the chosen frequency range. If no signal is visible, the attenuation can be consecutively decreased. Measuring in full-span mode serves mostly as a quick overview. To analyze the detected signals more closely, the span has to be decreased. Previous to this, the center frequency has to be set so the signal is at center of screen. Then span can be reduced. Then the resolution bandwidth can be decreased, and the video filter used if necessary. The warning uncal in the readout must not be displayed; otherwise measurement results may be incorrect. Obtaining values:
For a numerical value of a measurement result the easiest way is the use of the marker. The rotary dial, if necessary sets the marker frequency, uses the fine step mode. Then, read the value for the amplitude, which is shown in the readout. If a value is to be measured without using the marker, then measure the difference of the reference line to the signal. Observe that the scale may be either 5 dB/Div. in the reference level value the setting of the input attenuation is already included; it is not necessary to make a correction afterwards. The signal shown in the picture shows an amplitude difference of about 16dB to the reference line. Assuming that the reference level is 27 dBm, and the scale 10 dB/Div. Thus the signal has an amplitude of (-27 dBm) + (-16 dB) = -43dBm. In this value the setting of the input attenuator is already included.
Calculation Of Level Of Signal In Spectrum Analyzer 1. All levels are in RMS reading 2. The display has 8 * 10 divisions on graticule. 3. The top line has a level of 27 dBm or 10 mV (reference) 4. Each division of graticule (vertical) is 10 dBm. EXAMPLE (Fig.2) 1. Say signal is 2 divisions down from the top line. This means 20 dBm down with reference. 2. Suppose you have pushed 2 switches (attenuator 10 dB each) of spectrum analyzer. Now add to the reference the following Reference level display level + attenuator level = - 27 dBm (-27 dB) (20 dB) (20 dB) =10 mV rms. Or R - (Reading from top ref. Line)
EXPERIMENT NO:1
OBJECT: To measure Harmonics of SQUARE WAVE EQUIPMENT REQUIRED: 1. Spectrum analyzer 2. Signal source 3. BNC-BNC cable HM5005/HM5006 Function Generator HM5030-2 1 Nos.
PROCEDURE: A. Switch on the spectrum analyzer B. Switch ON the signal source (function generator) and set as given below: FUNCTION KNOB FREQUENCY KNOB 1 MHz
C. Set the spectrum analyzer as given below: CENTER FREQUENCY 000.0 ATTENUATION (-10 dB x 2) pressed SPAN WIDTH Adjust between 1to 1000 MHz/div D. Connect spectrum analyzer and signal generator as shown in the fig E. On connecting both the instruments you shall observe a spectral line other than the zero frequency Line. F. Now switch on the MARKER pushbutton .MK is lit and the display shows the MARKER frequency. G. Note down the level of the highest spectral line on the CRT display and calculate the level. H. Follow steps from (E) and (G) for different frequency on function generator and note the levels. OBSERVATION: S. NO. 1 2 3 4 5 Freq on Display Fundamental 1st Harmonic 2nd Harmonic 3rd Harmonic 4th Harmonic Level in dB
EXPERIMENT NO: 2 OBJECT: To measure harmonics of Triangular WAVE Procedure: - Same as used in Experiment no.1 OBSERVATIONS: S. NO. L 2 3 4 5 Freq on Display Fundamental 1st Harmonic 2nd Harmonic 3rd Harmonic 4th Harmonic Level in dB
EXPERIMENT NO: 3 OBJECT: To check the Frequency Response of a LOW PASS filter EQUIPMENT REQUIRED: Spectrum analyzer BNC-BNC cable
2Nos.
LOW PASS filter

PROCEDURE: A. Set the spectrum analyzer as given below: CENTER FREQUENCY 000.0 ATTENUATION (10 dB * 2) pressed SPAN WIDTH 20 MHz/div B. Set the tracking generator as given below: ATTENUATION ALL PRESSED ON /OFF Switch ON Level +1 dB (clockwise) C. Connect the given filter between the spectrum analyzer and tracking generator as shown in the fig.6.1 You will observe the curve as shown in the fig 6.2
D. Now switch on the MARKER pushbutton .MK is lit and the display shows the marker frequency . E. In order to find the bandwidth of the given filter start moving the marker knob on RHS and note its reading at -3 dB (Marked X in the fig. 6.2)
OBSERVATIONS: On higher side the 3dB frequency on display at X=38 MHz (approx).
EXPERIMENT NO: 4
OBJECT: To check the frequency response of a High Pass filter Procedure: A. Same as used in Experiment no.4 B. You will observer the curve as shown in fig. 7.2 C. In order to find the bandwidth of the given filter start moving the marker Knob and note its reading at 3dB as in fir. 7.2 OBSERVATIONS: -3 dB frequency reading on display X=5.5 MHz
Result:
EXPERIMENT NO: 5 OBJECT: To check frequency response of a BAND REJECT filter Procedure:A. Same as used in Experiment no.4 (Fig.8.1) B. Observe the curve as shown in fig. 8.2 C. In order to find the band width of the band reject filter start moving the marker knob and note its readings at 3dB point as shown in fig. 8.2 OBSERVATIONS: -3 dB frequency reading on display X= 2.6 MHz (approx) -3 dB frequency reading on display Y=4.6 MHz (approx) CALCULATIONS: Reject bandwidth of the given filter =Y-X =2 MHz (approx) With lower side at 2.6 MHz & higher side at 4.6 MHz EXPERIMENT NO: 6 OBJECT: Frequency response of a BAND PASS filter. Procedure:A. Same as used in Experiment no.3 (Fig.9.1) B. You will observe the curve as shown in fig. 9.2 C. In order to find the band width of the band reject filter start moving the marker knob and note its readings at 3dB point as shown in fig. 9.2
OBSERVATIONS: -3 dB frequency reading on display X=12 MHz (approx) -3 dB frequency reading on display Y=29 MHz (approx) CALCULATIONS: Pass bandwidth of the given filter =Y-X =17 MHz (approx)
Result: Precautions (if any):

Aim: To understand the basic concepts and working of a telephone exchange (EPABX). Objective: a. Understanding of telephone transmission. b. To understand the basic features of EPABX. c. Study of speech circuit using IC and its interface to the line. d. To observe the different signaling waveforms. e. Study of tone generation. EQUPMENTS: 20MHz Oscilloscope. Telecommunication Trainer Kit. Three Telephones Instrument. a. Understanding of telephone transmission. ABOUT TELEPHONE ELECTRONICS: The public switched telephone network is one of the true marvels of the modern word. It provides the ability to interconnect any two out of more than one hundred million telephones, usually with in a few seconds of the request for connection. It is controlled by the worlds largest network of interconnected and co-operating computers. Yet the telephones in this network are usable by unskilled operators without formal training. (Almost any child of four or five can make a telephone call). Let us understand the basics of telephone system THE TELEPHONE SET: Telephone sets like those used to originate and received telephone calls. It is simple in appearance and operation yet it performs a surprising number of functions. The most important one is: (1) It requests the use of the telephone systems when the handset is lifted. (2) It indicates that the system is ready for use by receiving a tone, called the dial tone. (3) It sends the number of the called telephone to the system. This number is initiated by the caller when the number is pressed or the dial is rotated. (4) It indicates the stage of the call in progress by receiving tones indicating the status (ringing, busy etc.). (5) It indicates the incoming call of the called telephone by ringing bells or other audible tones. (6) It changes speech of a calling party to electrical signal for transmission to distant party through the system. It changes electrical signals received from the distant party to speech for the called party.
Caller Phone
Called Phone
(7) It automatically adjusts for change in the power supplied to it. (8) It signals the system that a call is finished when a caller hangs up the handset. Of course, for a telephone to be of any use, it must be connected to another telephone. In the very early days of telephony, the phone was simply wired together with no switching. As the number of phones increased this became impractical, so the local exchange or central office was established to handle the switching and other functions. THE LOCAL LOOP: Each subscriber telephone is connected to a central office that contains switching equipment, signaling equipment, and batteries that supply direct current to operate the telephone. Each phone is connected to the central office through a local loop of two wire pair. One of the wires is called T (for tip) and the other is called R (for ring) that refers to the tip and ring parts of the plug used in the early manual switchboards. Switches in the central office respond to the dial pulses or tones from the telephone to connect the calling phone to the called phone. When the connection is established, the two telephones communicate over transformer-coupled loop using the current supplied by the central office batteries. INITIATING A CALL: When the handset of the telephone is resting in its cradle, the weight of the handset holds the switch hook buttons down and the switches are open. This is called the on-hook condition. The circuit between the telephone handset and the central office is open; however, the ringer circuit in the telephone is always connected to the central office is open; however, the ringer circuit in the telephone is always connected to the central office as shown. The capacitor, C, blocks the flow of DC from the battery but passes the AC ringing signal. (The ringer circuit presents high impedance to speech signals so it has no effect on them). When the handset is removed from its cradle, the spring-loaded buttons come up and the switch hook closes. This completes the circuits to the exchange and the current flows in the circuit. This is called off-hook condition. (The on-hook, off-hook, and hang-up terms came from the early days of telephony, when the receiver was separate and hung on the switch hook when not in use as shown. This also explains why many people still refer to the handset of as the receiver). The off-hook signal tells the exchange that some one wants to make a call. The exchange returns a dial tone to the calling to let the caller know that the exchange is ready to accept a telephone number. (The telephone number also may be referred to as an address). SENDING A NUMBER: Some telephone sets send the number by dial pulses while others send it by audio tone. DIAL PULSING: Telephone sets that use dial pulsing have rotary dial, which opens and closes the local loop circuit at a timed rate. The number of dial pulses resulting from one operation is determined by how far the dial is rotated before releasing it.
Although all network facilities are currently compatible with pulse dialing telephones with todays standard embraces the tone method of dialing. DUAL TONE MULTIFREQUENCY (DTMF): Most modern telephone sets employ the newer method of using audio tones to send the telephone number. These can be used only if the central office is equipped to process the tones. Instead of a rotary dial, these telephones set a push button keypad with twelve keys for the number 0 through 9 and the symbols (asterisk) and # (Pound sign). Pressing one the keys causes an electronic circuit in the keypad to generate two output tones that represent the number. CONNECTING THE PHONES: The central office has various switches and relays that automatically connects the calling and called phones. For now, that the connection has been made. If the called phone handset is off-hook when the connection is attempted, a busy tones generated by the central office is returned to the calling phone. Otherwise, a ringing signal is sent to the called phone to alert the called party that a call is waiting. At the same time, a ring back tone is returned to the calling phone to indicate that the called phone ringing. RINGING THE CALLED PHONE: Early telephone circuits were point-to-point (not switched), and the caller gained the attention of the party at the other end by picking up the transmitter and shouting Hello or Ahoy. This was not very satisfactory, and schemes based on the mechanical signaling arrangements were soon invented. The one common use today, called the polarized ringer, or bell, was patented in 1878 by Thomas A. Watson (Mr. Bells assistant). Electronic ringing circuits are quickly replacing polarized ringer in new telephone designs. ANSWERRING THE CALL: When the called party removes the handset in response to a ring, the loop to that phone is completed by its closed switch hook and loop current flows through the called telephone. The central office then removes the ringing signal and the ring back tone from the circuit.
TALKING: The part of the telephone in to which a person talks is called the transmitter. It converts speech (acoustical energy) in to variations in an electric current (electrical energy) by varying or modulating the loop current in accordance with the speech of the talker. The part of the telephone that converts the electric current variations in to sound that a person can hear is called the receiver. The signal produced by the transmitter is carried by the loop current variations to the receiver of the called party. Also, a small amount of the transmitter signal is fed back in to the talkers receiver. This is called the side tone. BEYOND THE LOCAL LOOP: Thus far, the discussion of connecting two telephones together has been limited to local loops and a central office exchange. Most central office exchange can handle up to 10,000 phones in different cities, different states, or countries? Over the years, a complex network of many telephone exchanges has been established to accomplish these requirements.
b. Basic features of EPABX (1) Access to Trunk Line (0) (Line hunting): Extensions may be programmed to have or all trunk lines by dialing 0.Extensions can also be denied this access. (2) Access to Reserved line (9): One or two lines may be kept reserved for certain extensions. Extension can access this line by dialing 9. (3) Extension to Extension Call (Ex): When one extension user wish to talk to another, the user has to proceed as follows: Lift hand set and hear dial tone and dial Extns. No.(30, 31,32,33), Wait for the internal ring tone. Speak when called party answers. Note: in case the called Extension is busy, use the call back facility explained in point 7. (4) Re-Dial (*): Any extension user can repeatedly dial the last no. (whether internal or external) without passing all the numbers again. For this follow the procedure given below. Disconnect previous call, lift handset and hear dial tone, Dial * key. The last dialed number will be redialed automatically. (5) Barge in (# Ex): If an extension is found busy, this feature allows the caller to interrupt the conversation of the busy extension. The feature can be used with a warning tone or without a warning tone. To use this feature operate as follows:Lift hand set, dial extension No. and on hearing a busy tone, disconnect. Lift hand set and hear dial tone, then Dial # Extension No., wait for second (If the feature is with a warning tone you will hear the same and if without a warning tone you will not.), Interrupt conversation.
(6)
Automatic Call Back on Busy Extension (# 13): If the called extension is busy, this feature automatically connects, as soon as the called
extension gets free. On hearing a busy tone wait for a few seconds for the dial tone to return then Dial # 13 and hang up. As soon as the called extension gets free, your extension will ring. Lift hand set, you will hear an internal ring tone, wait for the called party to answer. (7) Automatic Call Back on Busy Trunk Line (# 13): If all/any CO. Jn. Lines are/is busy, this feature inform the user as soon as the CO. Jn. (Trunk Line) Gets free. If a user gets a busy tone after attempting to seize any CO.Jn. Line, wait for a few seconds for the dial tone to return and then operate as follows. Dial (# 13) and hang up. Wait for the extension to ring, lift hand set and hear dial tine, dial 0 or access code for CO.Jn. Line. (8) Do not Disturb (# 14) : If an extension user does not wish to called, this allows the extension to prevent itself from being called. However, the extension user can call others. Lift set and obtain dial tone. Dial # 14 and hang on. To cancel this feature, Lift hand set and obtain dial tone. Dial # 0. (9) Extension Privacy (#15): The feature protects an extension user from being interrupted by any other extension during a conversation. Lift hand set and hear dial tone, Dial # 15 hang up. To cancel this feature, Lift hand set, hear dial tone and dial #0.
(10). Call Pick-up (8 or # 8 Ex. No.): If another extension is ringing, this feature allows the user to receive that call at his own extension with out physically moving to the ringing extension. Lift hand of your extension and hear the dial tone. If the user extension no. is from a different pick up group dial # 8 followed by the extension no. which is ringing. If from the same group, then just dial 8 and talk to the caller. (11). Follow Me (#16 Ex.): Incoming calls can be made to follow the extension user. In other words, the extension to receive incoming calls directed at his original extension. Lift hand set where user wants to receive calls, hear dial tone. Dial # 16 AB. (AB is the no. of original extension being used). All calls for AB will now ring at the extension from where the above code has been dialed. To cancel this feature, dial # 17AB.\ Note: After using the follow me feature care must be taken to cancel the feature otherwise all calls will be diverted to the extension, till the feature is cancelled. (12). Call Transfer (HF EX.):
Any
internal
or
external
call
received/originated
at
any
extension
can
be
transferred from that extension to any other extension. Hook Flash and heat feature mode tone. Dial the extension No. to which you want to transfer the call. Wait for the internal ring tone. If called extension is busy then use feature No. 8. The held party will hear music while on hold. (13). Conference (HF Extn.1 or TN 1 + HF EXT2 or TN 2): If while conversing with an outside line or an extension you want to arrange for a third or even a fourth party to conference, you may do it the following way. While conversing with party A use the hook flash, party A goes on hold. Listen to the dial tone and dial the extension number of party B. Speak to party B and hook flash to conference between yourself, party A and party B. Repeat the procedure to extend the conference to party C in the same manner. A maximum of 4 people can conference at any point of time.
c. Study of speech circuit using IC and its interface to the line. Block Diagram
THEORY: The Telephone system as it presently exists to a discussion of how it is changing and improving as electronic devices, usually in the form of integrated circuits, replace mechanical and conventional electrical devices. These Devices have most of the required components on the chip with connections provided for outboard components such as resistors and capacitors, which are used to program the chip. The emphasis is on the circuits that provided two-way speech in the telephone set. THE SPEECH CIRCUIT: A simplified block diagram of a speech circuits suitable for implementation as an integrated circuit is shown in built in transmit, receive , and side tone circuits, as well as a DC loop interface regulator, and equalizer circuit. It is connected to the telephone line by a conventional rectifier bridge. The bridge is dynamically equivalent to a small resistant in series with the signal path, and a high resistance in parallel to it. External components are used to adjust transmits, receive and side tone gains and frequency responses. The DC line interface controls the voltage and current characteristics of the entire speech network depending upon the value of loop current in the subscriber line. Regulator circuitry sets the operating voltage in the integrated circuit and biases the speech circuits.
PROCEDURE: (1) (2) (3) Pick up hand set of extension 30 and dial 31. You can hear ring at extension 31. Pick up hand set at extension 31 as you pick up hand set loop between two extension is Observe the wave form on C.R.O. at TP14 for extension 30 and at TP 15 for extension 31.
completed and speech path is established. You can observe ring signal at TP15 during ringing and after pick up you can observe voice signal (Voice Frequency) at test points. (4) Like above procedure we can establish speech path between CO line and extension also.
(d) To observe the different signaling waveforms at TP1 to TP31
TP1 TP2 TP3 TP4 TP5 TP7 TP8 TP9
RING VOLTAGE I/P 15V 20Hz CLOCK INTRRUPT SIGNAL 2MHz CLOCK CLOCK RING DETECTOR SIGNAL IN COI RING DETECTOR SIGNAL IN COII TONE RECEIVED I
TP12 OUTPUT OF DTMF TP13 OUTPUT OF DTMF TP14 RING SIGNAL OF EXTENSION 30 TP15 RING SIGNAL OF EXTENSION 31 TP16 RING SIGNAL OF EXTENSION 32 TP17 RING SIGNAL OF EXTENSION 33 TP18 EXCHANGE TONE TP19 DIAL TONE TP20 RING BACK SIGNAL TP21 MUSIC OUTPUT SIGNAL TP22 OUTPUT OF LATCH FR RING OF EXTENSION 30 TP23 OUTPUT OF LATCH FR RING OF EXTENSION 31 TP24 OUTPUT OF LATCH FR RING OF EXTENSION 32 TP25 OUTPUT OF LATCH FR RING OF EXTENSION 33 TP26 HOOKUP SIGNAL OF EXTENSION 1 TP27 HOOKUP SIGNAL OF EXTENSION 2 TP28 HOOKUP SIGNAL OF EXTENSION 4 TP29 HOOKUP SIGNAL OF EXTENSION 3 TP30 TRUNK ACCESS SIGNAL OF CO I TP31 TRUNK ACCESS SIGNAL OF CO II
e. Study of tone generation. Theory:
Various tones are used for both control and status indication. The tones may be single frequency or combinations of frequencies. These are analog signals that are either continuous tones or tone bursts (tones turned on and off at various rates). The call progress tones are sent by the exchange to the calling phone to inform the caller about the status of the call e,g, Dial tone is a continuous tone made by combining the frequencies of 350Hz and 440Hz. The busy signal that tells the caller that the called appears in bursts of 0.5 second on time separated by an off time of 0.5 second. All of these tones are in band signaling. Tone signaling between exchanges may be in-band or out-of-band. PROCEDURE AND OBERVATION: (1) Exchange dial tone: Lift the hand set of extension 30 and hear the dial tone is a
continuous sound which lasts for 8seconds during which exchange waits for dialing to be initiated this signal is as shown in figure and can be observed at TP18. If no dialing takes place during this period the EPABX times the user out and starts issuing a busy tone. (2) TP9. (3) Busy Tone: Lift the hand-set of extn.31 and keep it aside. Then lift the hand-set of extn.30 and dial the no.31 you will hear a busy tone which is a discontinuous sound(Du------Du). There are two types of busy tones one is a high speed busy tone which consists of equal duration ON and OFF signals. This tone indicates that the system is too busy. This tone consist of double duration ON and single duration OFF. (4) Internal Ring Back Tone: Lift the handset of extn.30 and dial any extn. Number like 31, 32 you will hear ring back tone that can be observed at TP14 for extn. E0. This is a discontinuous sound of two frequencies sounds for one second with a tw9o second silence interval. You will hear this tone till thee extension answers. This waveform is as shown in figure. (5) Ringing Tone: Two types of tings can be heard from the telephone instrument connected to the system. When your instrument is called by another internal instrument the ring will be continuous one with a one second ON and two second OFF waveform of ring signal is as shown in figure. A from a CO jn. Line will ring like a normal telephone. Result: P and Dial tone: On accessing a direct line by dialing 0 , you will obtain the normal P and T dial tone.( DOT line must be connected at COI and COII). This signal can be observed at
Precautions (if any) :
III Year Electronics & Communication Communication System Lab Aim: Familiarisation with transmission line trainer Kit.
Objective: a. Measuring the input impedance of a line b. Measuring the attenuation Equipment Required: Trainer Kit ST-2266 and CRO Introduction:
Transmission lines are means of conveying signals or power from one point to another. One of the simplest forms of the transmission line is the open wire line or the twisted pair. Since the two conductors of this type of line have same relationship with respect to ground. It is BALANCED LINE but this type of line has very poor shielding properties and has a tendency to radiate. Another co-axial line consists of a central conductor and an outer conductor with the outer conductor referred to as shield normally grounded. Due to shielding, co-axial line have extremely low radiation loss. Since each conductor has certain length and diameter, it must have resistance and inductance, since there are two wires close to each other, there must be capacitance between them. Finally the wires are separated by a medium called the dielectric, which cannot be prefect in its insulation, the current leakage through it can be represented by a shunt conductance. The ohmic resistance R & conductance G are responsible for energy disputation in the form of heat. The losses, which determines the attenuation characteristic are expressed in terms of attenuation a and can be calculated by a= 20Log(V2/V1) Where V1= amplitude of signal at i/p V2= amplitude of signal at o/p a= attenuation for given length. Here total effective resistance of line is 68, for optimum power transfer we should have the source resistance and terminating resistance also as 68. Assuming generator resistance Rg as 50. We must connect 18 Ri in series with the generator to match the line.
Procedure:
A. Measuring the Attenuation of a transmission line 1. Adjust Ri and RL for 18 and 68 respectively with the help of DMM. 2. Make the connections as shown in dig. 2. 3. Set the sine-wave frequency to approx. 100KHz and level to 0.4V. 4. Oscilloscope CH1 shows applied input CH2 shows outputs. 5. Measure signal level at Input and at 25, 50 75 and 100m length. 6. Tabulate as under:
Length (m) 25 50 75 100
V1 (input)
V2 (output)
7. Now calculate the attenuation in its at various length by for formulae given below a=20Log(V2/V1) Measuring the input impedance a line 8. Repeat step 1 9. Make a connection as shown in dig. 3 10. A 1 resistance in series between the generator & the transmission line as shown in fig.6 allows to measure the value of i/p current. 11. Set the i/p at 0.4VPP and frequency 100KHz of sine-wave (both measurement on CRO) 12. Take readings of Vin & Vm (across 1) on oscilloscope 13. Calculate the i/p impedance according to the following formula: Zi=Vin/I =Vin/Vm* 1 Note down this result Result: Precautions (if any): -

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Precautions (if any):

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Precautions (if any):

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III YEAR Electronics & Communication Communication System Lab-II

Aim: Harmonic analysis of a given signal using Spectrum Analyzer. Objective:

Equipments Required: Introduction:

LOW PASS filter

Precautions (if any):

Result: Precautions (if any):

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(d) To observe the different signaling waveforms at TP1 to TP31

TP1 TP2 TP3 TP4 TP5 TP7 TP8 TP9

e. Study of tone generation. Theory:

Precautions (if any) :

Length (m) 25 50 75 100

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