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Free-Space loss, that is, the ratio of transmitted power and the received power in the case of isotropic antennas

………………………………. (1)

Free-Space Loss of Radio Waves

………………………….. (2)

The maximum transmitting and receiving gain (to direction of maximum radiation or sensitivity) of an antenna with
effective aperture area Ae is

……………………… (3)
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Symbol Rate (Baud Rate) and Bandwidth

Figure D shows the shape of a rectangular pulse with duration T before and after it passed through an ideal lowpass
channel of bandwidth B. For example, if the duration of the pulse T = 1 ms, distorted pulses are shown in the figure
for the channel with bandwidths B = 2⋅1/T = 2 kHz, B = 1/T = 1 kHz, B =1/2⋅1/T = 500 Hz, and
B = 250 Hz. If the next pulse is sent immediately after the one in the figure, the detection of the pulse value will be
impossible if the bandwidth is too narrow. The spread of pulses over other pulses, which disturbs detection of other
pulses in the sequence, is called intersymbol interference (ISI).

In baseband transmission, a digital signal with r symbols per second, bauds, requires the transmission bandwidth B to
be in hertz:

B ≥ r /2 ……………………………………. (6)

Symbol Rate and Bit Rate

Generally, the bit rate depends on modulation rate according to rb = k · r bps ………………………………….. (7)
where k represents the number of bits encoded into each symbol. Then the number of symbol values is M = 2k and the
bit rate is given as rb = r log2 M bps. In the example of Figure E, the number of symbol values is M = 2k = 22 = 4, and
the bit rate rb = k · r bps = 2r bps. Then the symbol rate of 1 kbaud makes the bit rate 2 Kbps.
Table A. Bit Rate of a System Using Multiple Symbol Values

Maximum Capacity of a Transmission Channel

We saw previously that the bandwidth of a channel sets the limit to the symbol rate in bauds but not to the
information data rate. In 1948, Claude Shannon published a study of the theoretical maximum data rate in the case of
a channel subject to random (thermal) noise.
We measure a noise relative to a signal in terms of the S/N. Noise degrades fidelity in analog communication and
produces errors in digital communication. The S/N is usually expressed in decibels as

S /N dB = 10 log10 (S /N) dB ………………………….. (8)

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Taking both bandwidth and noise into account, Shannon stated that the error-free bit rate through any transmission
channel cannot exceed the maximum capacity C of the channel given by:

C = B log2 (1 + S/N) ………………………………. (9)

where C is the maximum information data rate in bits per second; B, the bandwidth in hertz; S, the signal power; N,
the noise power, and S/N, the S/N power ratio (absolute power ratio, not in decibels).

However, modulation moves the spectrum of the pulse from low frequencies to carrier frequencies, and the
bandwidth is typically doubled when compared with baseband systems as was shown in Figure 1.2 (under S & S).
This is why the symbol rate in radio systems is less than or equal to the transmission bandwidth, that is:

r ≤ BT …………………………………… (10)

where r is the symbol rate in bauds and BT is the transmission bandwidth in hertz.
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The accurate requirement of bandwidth depends on the modulation scheme in use, the study of which is beyond the
scope of this course.

Example 4.1
Assume that the transmission channel is an ideal lowpass channel with a bandwidth of 4 kHz. The maximum symbol
rate via this channel is r ≤ 2 · B = 8 kbauds; that is, we can transmit up to 8,000 independent signals, symbols, in a
second. [To transmit the same symbol rate through a bandpass channel, we would need a bandwidth of 8 kHz
according to (eqn 10); see also Figure 1.2.]

Example 4.2
Assume that the S/N of a lowpass channel is 28 dB and its bandwidth is 4 kHz. Then S/N dB = 10 log10 S/N, S/N =
102.8 ≈ 631. The maximum bit rate according to (eqn 9) is C = B log2 (1 + S/N) = 4,000 log2 (432) = 4,000 (log10
632)/log10 2 = 37.2 Kbps. In Example 4.1 we learned that the maximum symbol rate is 4 kbauds, which depends only
on the bandwidth. To achieve the maximum bit rate, we transmit 4,000 symbols in a second and each of them carries
3 bits (with 4 bits, the maximum bit rate would be exceeded). The number of different symbols that can be used is 2 3
= 8 and this depends only on the S/N maximum, not on bandwidth.
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CODING
Coding is a digital symbol processing operation in which the digital form of the information is changed for improved
communication. In general, coding contains many different processes, such as ciphering, compression, and error
control coding. For ciphering, the transmitter and the receiver may simply perform an exclusive-or operation with
data and a ciphering sequence known only by the transmitter and receiver. An eavesdropper is not able to detect
information content without knowing the ciphering sequence.

Most modern systems use error control codes for handling of transmission errors. By appending extra check digits
to the transmitted data, we can detect or even correct errors that occur on the line. Error control coding increases both
the required bandwidth (data rate increases) and the hardware and software complexity, but it pays off in terms of
nearly error-free digital communication even when the S/N is low.

Still another purpose for coding is for compressing information. By using data compression we can reduce the disk
space needed to store data in a computer. In the same way we can decrease the required data rate on the line to a
small fraction of the original information data rate.

The operation of line encoding transforms a digital message into a new sequence of symbols. Decoding is the
opposite process that converts the encoded sequence back into the original message (Figure G).

Fig. G Line Coding

PURPOSE OF LINE CODING

1. to make the form of the spectrum of a digital signal suitable for a certain communication media. The line
codes usually have no dc. We want to get rid of the dc that does not transmit any information but wastes power.
2. Another reason for line encoding is to help to synchronize the receiver.
In digital transmissions the receiver must be synchronized with the transmitter in order to receive the information in
proper manner. For this the data should be transmitted in a form that contains synchronization information so that
there is no need to transmit additional clock or timing signals.

The systems that use only line coding, but not modulation, are called baseband transmission systems. The
spectrum of the line signal is still in the frequency range of the original message’s “baseband.” In radio systems both
code and modulation is used.

SPECTRUM OF COMMON LINE CODES

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To determine what kind of impact line encoding has on the spectrum, we look at the characteristics of some common
line codes. Figure I presents their power density spectrums showing how the signal power of random data is
distributed over the frequencies.

Nonreturn to Zero (NRZ)

NRZ is the most common form of digital signal used internally in digital systems. Each symbol has a constant value
corresponding to binary symbol values 1 and 0. The spectrum has a high dc component, and there are no discrete
spectral components at the harmonic frequencies of the data rate.

The harmonic frequencies are multiples of the data rate. An external clock signal is always needed for the timing of
the receiver.

Return to Zero (RZ)

RZ each symbol is cut into two parts. The first half of the symbol represents the binary value and the rest of the
symbol is always set to zero. Because pulses are shorter than in the case of NRZ the spectrum is wider, as we saw in
Figure 1.2, and the spectrum of a random data has strong discrete frequency components at the harmonic frequencies
of the data rate. With the help of these components, timing information can be extracted from the signal spectrum and
an external clock is not necessarily needed. However, because RZ code has high low-frequency content and a
wide spectrum (see Figure I), it is never used in long-distance transmission. Another problem is that
synchronization is lost if the data content is all zero for a long period of time.
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Fig. I Common Line Codes and their power spectrums

Alternate Mark Inversion (AMI)

If every other mark or 1 of the NRZ or RZ symbols is transmitted as an inverted voltage polarity, an AMI signal is
produced. The advantage of this is that no dc component is present on the transmission line. The dc component is
unwanted because it does not carry any information; it merely wastes power. With the help of this kind of code we
can avoid the problem caused by transformers on the line
AMI code is used in American telecommunications network in primary rate 1.5-Mbps transmission systems.

High-Density Bipolar 3
HDB-3 was developed from AMI and standardized for European primary rate 2-Mbps systems. HDB-3 overcomes
the problem of the original AMI code that occurs in the timing when a data message contains long periods of
subsequent zeroes. In this coding scheme, a pulse with the same polarity as the previous one is added in such a way
that no more than three sequential zeroes are allowed. In the decoder these pulses are taken away according to the
AMI coding rule that they violate.

Manchester Coding
Manchester coding is used in LANs. Binary digit 1 is coded as a “+ to –” transition and binary 0 as a “– to +”
transition. The most important advantage of the Manchester code is that each symbol contains the timing information
and the receiver needs only to detect the transition in the middle of each received symbol to extract the clock signal.
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Its main disadvantage is a wide spectrum because of short pulses and this is why it is suitable for LANs but not for
long-distance transmission.

REGENERATION

In long-haul transmissions, the transmitted signal is attenuated and amplifiers or repeaters are needed. Analog
amplifiers amplify the signal at the input, and the signal contains both the desired message and channel noise. In
every amplifier and cable section some noise is added and the S/N decreases with distance.

Unlike analog amplifiers, digital repeaters are regenerative. A regenerative repeater station consists of an
equalizing amplifier that compensates the distortion and filters out the out-of-band noise and a comparator as shown
in Figure J. Output of the comparator is high if the input signal is above the threshold voltage Vref, and low if the
input is below the threshold value.

Table B Examples of Error Rates and Mean Times Between Errors for a 64-Kbps Channel

When the S/N of a digital system decreases, errors occur more and more frequently and when the error rate becomes
too high, information is lost. An error rate of 1 × 10–3 is standardized to be the worst allowed communication quality
for PCM speech in the telecommunications network. If the error rate becomes worse, ongoing calls are cut off. Data
are transmitted in large packets and if a packet contains one or more errors it needs to be retransmitted. As a rule of
thumb, we can say that data transmission requires an error rate of 1 × 10–5 or better, otherwise retransmissions slow
down the end-to-end transmission data rate.

MULTIPLEXING
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Multiplexing is a process that combines several signals for simultaneous transmission on one transmission channel.
Most of the transmission systems in the telecommunications network contain more capacity than is required by a
single user. It is economically feasible to utilize the available bandwidth of optical fiber or coaxial cable or a radio
system in a single high-capacity system shared by multiple users. The main principles of multiplexing are described
below.

Frequency-Division Multiplexing (FDM)

FDM modulates each message to a different carrier frequency. The modulated messages are transmitted through the
same channel and a bank of filters separates the messages at the destination (Figure K). The frequency band of the
system is divided into several narrowband channels, one for each user.
Each narrowband channel is reserved for one user all the time. FDM has been used in analog carrier systems in the
telephone network. The same principle is also used in analog cellular systems in which each user occupies one FDM
channel for the duration of the call. In such a case, we call the process frequency-division multiple access (FDMA)
because the frequency-division method is now used to allow multiple users to access the network at the same time.

Fig. K Multiplexing methods FDM and TDM

Time-Division Multiplexing (TDM)

A more modern method of multiplexing is TDM, which puts different messages, for example, PCM words from
different users, in nonoverlapping time slots. Each user channel uses a wider frequency band but only a small fraction
of time, one time slot in each frame as shown Figure K. In addition to the user channels, framing information is
needed for the switching circuit at the receiver that separates the user channels (time slots) in the demultiplexer.
When the demultiplexer detects the frame synchronization word, it knows that this is the start of a new frame and the
next time slot contains the information of user channel 1.

PCM Frame Structure

As an example of TDM and to get a clear view of TDM, we now look at the most common frame structure in
telecommunications networks, namely, the primary rate 2,048-Kbps frame used in the European standard areas.
This is the basic data stream that carries speech channels and ISDN-B channels through the network and it is called
E-1. The corresponding North American primary rate is 1.544 Mbps, which carries 24 speech channels and it is
known as DS1 or T1.

In the European scheme, the primary rate frame is built up in digital local exchanges that multiplex 30 speech or data
channels at bit rate of 64 Kbps into the 2,048-Kbps data rate. ITU-T defines this frame structure in Recommendation
G.704.
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The 2-Mbps Frame Structure

PCM-coded speech is transmitted as 8-bit samples 8,000 times a second, which makes up a 64-Kbps data rate. These
eight-bit words from different users are interleaved into a frame at a higher data rate.

The 2,048-Kbps frame in Figure L is used in the countries implementing European standards for telecommunications.
It contains 32 time slots, and 30 of them are used for speech or 64-Kbps data. The frame is repeated 8,000 times a
second, which is the same as the PCM sampling rate.
Each time slot contains an eight-bit sample value and the data rate of each channel is 64 Kbps. These voice channels
or data channels are synchronously multiplexed into a 2-Mbps data stream, which is often called E1 (first level in
European hierarchy). For error-free operation the tributaries (64-Kbps data streams of the users) have to be
synchronized with the clock signal of the 2-Mbps multiplexer. The data rate of 2,048 Kbps for the multiplexer is
allowed to vary by 50 parts per million (ppm), and as a consequence each user of the network has to take timing from
the multiplexer in the network and generate data exactly at the data rate of the multiplexer divided by 32.

Fig. L The 2,048-Kbps frame structure from Recommendation G.704.

(i) Frame Synchronization Time Slot

The frame alignment word is needed to inform the demultiplexer where the words of the channels are located in the
received 2-Mbps data stream. The frame synchronization time slot (TS0) includes frame alignment information and it
has two different contents that are alternated in subsequent frames (Figure M). The demultiplexer looks for this time
slot in the received data stream and, when it is found, locks onto it and starts picking up bytes from the time slots for
each receiving user. Each user receives 8 bits in 125-μs periods, which makes 64 Kbps. A fixed alignment word is not
reliable enough for frame synchronization because it may happen that a user’s data from one channel simulates the
synchronization word and the demultiplexer might lock to this user time slot instead of TS0. This is why there is one
alternating bit (D2) in time slot 0 (see Figure M) and due to this the demultiplexer is able to detect the situation where
one channel constantly transmits a word that is equal to the frame alignment word (FAW).
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(ii) Multiframe Structure of the Signaling Time Slot

Time slot 16 (TS16) is defined to be used for the channel associated signaling to carry separate signaling information
to all user channels of the frame. TS16 is a transparent 64-Kbps data channel like any other time slot in the frame.
Thirty channels share the signaling capacity of TS16. A frame structure is needed to allocate the bits of this time slot
to each of the 30 speech channels. The information about the location of the signaling data of each speech channel is
given to the signaling demultiplexer with the help of the multiframe structure containing a multiframe alignment
word for multiframe synchronization. The data rate available for each speech channel is 2 Kbps.