Methods for transmitting discrete data at the physical level. Methods of data transmission at the physical level. Public Switched Telephone Network

Two main types of physical encoding are used - based on a sinusoidal carrier signal (analog modulation) and based on a sequence of rectangular pulses (digital encoding).

Analog modulation - for transmitting discrete data over a channel with a narrow bandwidth - telephone networks voice-frequency channel (bandwidth from 300 to 3400 Hz) A device that performs modulation and demodulation - a modem.

Analog modulation methods

n amplitude modulation (low noise immunity, often used in conjunction with phase modulation);

n frequency modulation (complex technical implementation, usually used in low-speed modems).

n phase modulation.

Modulated signal spectrum

Potential code- if discrete data is transmitted at a speed of N bits per second, then the spectrum consists of a constant component of zero frequency and an infinite series of harmonics with frequencies f0, 3f0, 5f0, 7f0, ..., where f0 = N/2. The amplitudes of these harmonics decrease slowly - with coefficients of 1/3, 1/5, 1/7, ... from the amplitude f0. The spectrum of the resulting potential code signal when transmitting arbitrary data occupies a band from a certain value close to 0 to approximately 7f0. For a voice frequency channel, the upper limit of the transmission rate is achieved for a data transfer rate of 971 bits per second, and the lower limit is unacceptable for any speed, since the channel bandwidth starts at 300 Hz. That is, potential codes are not used on voice frequency channels.

Amplitude modulation- the spectrum consists of a sinusoid of the carrier frequency fc and two side harmonics fc+fm and fc-fm, where fm is the frequency of change of the information parameter of the sinusoid, which coincides with the data transmission rate when using two amplitude levels. The fm frequency determines the line capacity for a given coding method. With a small modulation frequency, the signal spectrum width will also be small (equal to 2fm), and the signals will not be distorted by the line if the bandwidth is greater than or equal to 2fm. For a voice frequency channel, this method is acceptable at a data transfer rate of no higher than 3100 / 2 = 1550 bits per second.

Phase and frequency modulation- the spectrum is more complex, but symmetrical, with a large number of rapidly decreasing harmonics. These methods are suitable for transmission over a voice frequency channel.

Quadrate Amplitude Modulation - phase modulation with 8 phase shift values ​​and amplitude modulation with 4 amplitude values. Not all 32 signal combinations are used.

Digital coding

Potential codes– to represent logical ones and zeros, only the value of the signal potential is used, and its drops, which formulate completed pulses, are not taken into account.

Pulse codes– represent binary data either as pulses of a certain polarity, or as part of a pulse - as a potential difference in a certain direction.

Requirements for the digital coding method:

At the same bit rate, it had the smallest spectrum width of the resulting signal (a narrower signal spectrum makes it possible to achieve a higher data transfer rate on the same line; there is also a requirement for the absence of a constant component, that is, the presence of a direct current between the transmitter and the receiver);

Provided synchronization between the transmitter and the receiver (the receiver must know exactly at what point in time to read the necessary information from the line, in local systems - clock lines, in networks - self-synchronizing codes, the signals of which carry instructions for the transmitter about at what point in time it is necessary to carry out recognition of the next bit);

Possessed the ability to recognize mistakes;

It had a low cost of implementation.

Potential code without returning to zero. NRZ (Non Return to Zero). The signal does not return to zero during the clock cycle.

It is easy to implement, has good error recognition due to two sharply different signals, but does not have the property of synchronization. When transmitting a long sequence of zeros or ones, the signal on the line does not change, so the receiver cannot determine when the data needs to be read again. Another disadvantage is the presence of a low-frequency component, which approaches zero when transmitting long sequences of ones and zeros. The code is rarely used in its pure form; modifications are used. Attractiveness – low frequency of the fundamental harmonic f0 = N /2.

Bipolar coding method with alternative inversion. (Bipolar Alternate Mark Inversion, AMI), modification of the NRZ method.

To encode zero, a zero potential is used, a logical unit is encoded either by a positive potential or a negative one, with the potential of each subsequent unit being opposite to the potential of the previous one. Partially eliminates the problems of constant component and lack of self-synchronization. In the case of transmitting a long sequence of units, a sequence of multi-polar pulses with the same spectrum as the NRZ code transmitting a sequence of alternating pulses, that is, without a constant component and a fundamental harmonic N/2. In general, the use of AMI results in narrower spectrum than NRZ and therefore higher link capacity. For example, when transmitting alternating zeros and ones, the fundamental harmonic f0 has a frequency of N/4. It is possible to recognize erroneous transmissions, but to ensure reliable reception it is necessary to increase the power by about 3 dB, since signal level adjustments are used.

Potential code with inversion at one. (Non Return to Zero with ones Inverted, NRZI) AMI-like code with two signal levels. When transmitting a zero, the potential of the previous cycle is transmitted, and when transmitting a one, the potential is inverted to the opposite. The code is convenient in cases where the use of the third level is not desirable (optical cable).

Two methods are used to improve AMI, NRZI. The first is adding redundant units to the code. The property of self-synchronization appears, the constant component disappears and the spectrum narrows, but the useful throughput decreases.

Another method is to “mix” the initial information so that the probability of the appearance of ones and zeros on the line becomes close - scrambling. Both methods are logical coding, since they do not determine the shape of the signals on the line.

Bipolar pulse code. One is represented by a pulse of one polarity, and zero by another. Each pulse lasts half a beat.

The code has excellent self-synchronizing properties, but when transmitting a long sequence of zeros or ones, there may be a constant component. The spectrum is wider than that of potential codes.

Manchester code. The most common code used in Ethernet networks is Token Ring.

Each measure is divided into two parts. Information is encoded by potential drops that occur in the middle of a clock cycle. A one is encoded by a drop from a low signal level to a high one, and a zero is coded by a reverse drop. At the beginning of each clock cycle, a service signal drop may occur if several ones or zeros need to be represented in a row. The code has excellent self-synchronizing properties. The bandwidth is narrower than that of a bipolar pulse; there is no constant component, and the fundamental harmonic in the worst case has a frequency of N, and in the best - N/2.

Potential code 2B1Q. Every two bits are transmitted in one clock cycle by a four-state signal. 00 - -2.5 V, 01 - -0.833 V, 11 - +0.833 V, 10 - +2.5 V. Additional means are required to deal with long sequences of identical bit pairs. With random alternation of bits, the spectrum is twice as narrow as that of NRZ, since at the same bit rate the clock cycle duration is doubled, that is, it is possible to transmit data over the same line twice as fast as using AMI, NRZI , but you need more transmitter power.

Logic coding

Designed to improve potential codes such as AMI, NRZI, 2B1Q, replacing long sequences of bits leading to a constant potential with interspersed units. Two methods are used - redundant coding and scrambling.

Redundant codes are based on breaking the original sequence of bits into portions, which are often called symbols, after which each original symbol is replaced by a new one that has more bits than the original.

The 4B/5B code replaces sequences of 4 bits with sequences of 5 bits. Then, instead of 16 bit combinations, you get 32. Of these, 16 are selected that do not contain a large number of zeros, the rest are considered forbidden codes (code violation). In addition to eliminating the DC component and making the code self-synchronizing, redundant codes allow the receiver to recognize corrupted bits. If the receiver receives prohibited codes, it means that the signal has been distorted on the line.

This code is transmitted over the line using physical encoding using a potential encoding method that is sensitive only to long sequences of zeros. The code guarantees that there will not be more than three zeros in a row on the line. There are other codes, such as 8B/6T.

To ensure a given throughput, the transmitter must operate at a higher clock frequency (for 100 Mb/s - 125 MHz). The signal spectrum expands compared to the original one, but remains narrower than the Manchester code spectrum.

Scrambling - mixing data with a scrambler before transmission from the line.

Scrambling methods involve bit-by-bit calculation of the resulting code based on the bits of the source code and the bits of the resulting code obtained in previous clock cycles. For example,

B i = A i xor B i -3 xor B i -5 ,

where B i is the binary digit of the resulting code obtained at the i-th clock cycle of the scrambler, A i is the binary digit of the source code received at the i-th clock cycle at the input of the scrambler, B i -3 and B i -5 are the binary digits of the resulting code , obtained in previous cycles of work.

For the sequence 110110000001, the scrambler will give 110001101111, that is, there will be no sequence of six consecutive zeros.

After receiving the resulting sequence, the receiver will transmit it to the descrambler, which will apply the inverse transformation

C i = B i xor B i-3 xor B i-5 ,

Different scrambling systems differ in the number of terms and the shift between them.

There are simpler methods of dealing with sequences of zeros or ones, which are also classified as scrambling methods.

To improve Bipolar AMI the following are used:

B8ZS (Bipolar with 8-Zeros Substitution) – corrects only sequences consisting of 8 zeros.

To do this, after the first three zeros, instead of the remaining five, he inserts five signals V-1*-0-V-1*, where V denotes a one signal that is prohibited for a given polarity cycle, that is, a signal that does not change the polarity of the previous one, 1* - the unit signal is of the correct polarity, and the asterisk sign marks the fact that in the source code there was not a unit in this clock cycle, but a zero. As a result, at 8 clock cycles the receiver observes 2 distortions - it is very unlikely that this happened due to noise on the line. Therefore, the receiver considers such violations to be an encoding of 8 consecutive zeros. In this code, the constant component is zero for any sequence of binary digits.

The HDB3 code corrects any four consecutive zeros in the original sequence. Every four zeros are replaced by four signals, in which there is one V signal. To suppress the DC component, the polarity of the V signal is alternated in successive replacements. In addition, two patterns of four-cycle codes are used for replacement. If before replacement the source code contained an odd number of units, then the sequence 000V is used, and if the number of units was even, the sequence 1*00V is used.

Improved potential codes have a fairly narrow bandwidth for any sequences of zeros and ones that occur in the transmitted data.

When transmitting discrete data over communication channels, two main types of physical coding are used - based on a sinusoidal carrier signal and based on a sequence of rectangular pulses. The first method is often called modulation or analog modulation, emphasizing the fact that encoding is carried out by changing the parameters of the analog signal. The second method is usually called digital coding. These methods differ in the width of the spectrum of the resulting signal and the complexity of the equipment required for their implementation.

When using rectangular pulses, the spectrum of the resulting signal is very wide. This is not surprising if we remember that the spectrum of an ideal pulse has an infinite width. The use of a sine wave results in a spectrum of much smaller width at the same information transfer rate. However, to implement sinusoidal modulation, more complex and expensive equipment is required than to implement rectangular pulses.

Currently, increasingly, data that was originally in analog form - speech, television images - is transmitted over communication channels in discrete form, that is, as a sequence of ones and zeros. The process of representing analog information in discrete form is called discrete modulation. The terms "modulation" and "coding" are often used interchangeably.

2.2.1. Analog modulation

Analog modulation is used to transmit discrete data over channels with a narrow frequency band, a typical representative of which is voice channel, made available to users of public telephone networks. A typical amplitude-frequency response of a voice frequency channel is shown in Fig. 2.12. This channel transmits frequencies in the range from 300 to 3400 Hz, so its bandwidth is 3100 Hz. Although the human voice has a much wider range - from approximately 100 Hz to 10 kHz - for acceptable speech quality, the 3100 Hz range is a good solution. Strict limitation of voice channel bandwidth is associated with the use of multiplexing and channel switching equipment in telephone networks.

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A device that performs the functions of carrier sinusoid modulation on the transmitting side and demodulation on the receiving side is called modem(modulator-demodulator).

Analog modulation methods

Analog modulation is a physical encoding method in which information is encoded by changing the amplitude, frequency or phase of a sinusoidal carrier signal. The main methods of analog modulation are shown in Fig. 2.13. On the diagram (Fig. 2.13, A) shows a sequence of bits of source information, represented by high-level potentials for a logical one and a zero-level potential for logical zero. This encoding method is called potential code, which is often used when transferring data between computer blocks.

At amplitude modulation(Fig. 2.13, 6) for a logical unit one level of the amplitude of the carrier frequency sinusoid is selected, and for a logical zero - another. This method is rarely used in its pure form in practice due to low noise immunity, but is often used in combination with another type of modulation - phase modulation.

At frequency modulation(Fig. 2.13, c) the values ​​0 and 1 of the source data are transmitted by sinusoids with different frequencies - fo and fi. This modulation method does not require complex circuitry in modems and is typically used in low-speed modems operating at 300 or 1200 bps.

At phase modulation(Fig. 2.13, d) data values ​​0 and 1 correspond to signals of the same frequency, but with different phases, for example 0 and 180 degrees or 0.90,180 and 270 degrees.

High-speed modems often use combined modulation methods, usually amplitude combined with phase.

Chapter 2. Basics of Discrete Data Transfer

Modulated signal spectrum

The spectrum of the resulting modulated signal depends on the type of modulation and the modulation rate, that is, the desired bit rate of the original information.

Let us first consider the spectrum of the signal during potential encoding. Let a logical one be encoded by a positive potential, and a logical zero by a negative potential of the same magnitude. To simplify the calculations, we assume that information is transmitted consisting of an infinite sequence of alternating ones and zeros, as shown in Fig. 2.13, A. Note that in this case the values ​​of baud and bits per second are the same.

For potential encoding, the spectrum is directly obtained from the Fourier formulas for the periodic function. If discrete data is transmitted at a bit rate of N bit/s, then the spectrum consists of a constant component of zero frequency and an infinite series of harmonics with frequencies fo, 3fo, 5fo, 7fo,..., where fo = N/2. The amplitudes of these harmonics decrease quite slowly - with coefficients of 1/3, 1/5,1/7,... from the amplitude of the harmonic fo (Fig. 2.14, A). As a result, the spectrum of potential code requires a wide bandwidth for high-quality transmission. In addition, you need to take into account that in reality the signal spectrum is constantly changing depending on what data is transmitted over the communication line. For example, transmitting a long sequence of zeros or ones shifts the spectrum towards lower frequencies, and in the extreme case where the transmitted data consists of only ones (or only zeros), the spectrum consists of a harmonic of the zero frequency. When transmitting alternating ones and zeros, there is no constant component. Therefore, the spectrum of the resulting potential code signal when transmitting arbitrary data occupies a band from a certain value close to 0 Hz to approximately 7fo (harmonics with frequencies above 7fo can be neglected due to their small contribution to the resulting signal). For a voice frequency channel, the upper limit for potential encoding is achieved for a data rate of 971 bps, and the lower limit is unacceptable for any speed, since the channel bandwidth starts at 300 Hz. As a result, candidate codes on voice channels are never used.

2.2. Methods for transmitting discrete data at the physical level 135

With amplitude modulation, the spectrum consists of a sinusoid of the carrier frequency f c and two side harmonics: (f c + f m) and (f c - f m), where f m is the frequency of change of the information parameter of the sinusoid, which coincides with the data transmission rate when using two amplitude levels (Fig. 2.14, 6). The frequency f m determines the line capacity for a given coding method. At a small modulation frequency, the signal spectrum width will also be small (equal to 2f m), so the signals will not be distorted by the line if its bandwidth is greater than or equal to 2f m. For a voice frequency channel, this modulation method is acceptable at a data transfer rate of no more than 3100/2=1550 bps. If 4 amplitude levels are used to present data, then the channel capacity increases to 3100 bps.

With phase and frequency modulation, the signal spectrum is more complex than with amplitude modulation, since more than two side harmonics are formed here, but they are also symmetrically located relative to the main carrier frequency, and their amplitudes quickly decrease. Therefore, these types of modulation are also well suited for data transmission over a voice frequency channel.

To increase the data transfer rate, combined modulation methods are used. The most common methods are quadrature amplitude modulation (QAM). These methods are based on a combination of phase modulation with 8 phase shift values ​​and amplitude modulation with 4 amplitude levels. However, of the possible 32 signal combinations, not all are used. For example, in the codes Trellis Only 6, 7 or 8 combinations are allowed to represent the original data, and the remaining combinations are prohibited. Such coding redundancy is required for the modem to recognize erroneous signals resulting from distortions due to interference, which on telephone channels, especially dial-up ones, are very significant in amplitude and long in time.

2.2.2. Digital coding

When digitally encoding discrete information, potential and pulse codes are used.

In potential codes, only the potential value of the signal is used to represent logical ones and zeros, and its drops, which form complete pulses, are not taken into account. Pulse codes allow you to represent binary data either as pulses of a certain polarity, or as part of a pulse - a potential difference in a certain direction.

Requirements for digital coding methods

When using rectangular pulses to transmit discrete information, it is necessary to choose a coding method that simultaneously achieves several goals:

At the same bit rate, it had the smallest spectrum width of the resulting signal;

Provided synchronization between the transmitter and receiver;

Possessed the ability to recognize mistakes;

It had a low cost of implementation.

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A narrower spectrum of signals allows one and the same line (with the same bandwidth) to achieve a higher data transfer rate. In addition, the signal spectrum is often required to have no DC component, that is, the presence of a DC current between the transmitter and receiver. In particular, the use of various transformer circuits galvanic isolation prevents the passage of direct current.

Synchronization of the transmitter and receiver is necessary so that the receiver knows exactly at what point in time it is necessary to read new information from the communication line. This problem is more difficult to solve in networks than when exchanging data between closely located devices, for example, between units inside a computer or between a computer and a printer. At short distances, a scheme based on a separate clock communication line works well (Fig. 2.15), so that information is removed from the data line only at the moment the clock pulse arrives. In networks, the use of this scheme causes difficulties due to the heterogeneity of the characteristics of conductors in cables. Over long distances, uneven signal propagation speed can cause the clock pulse to arrive so late or before the corresponding data signal that the data bit is skipped or read again. Another reason why networks refuse to use clock pulses is to save conductors in expensive cables.

Therefore, networks use so-called self-synchronizing codes, the signals of which carry instructions for the transmitter at what point in time it is necessary to recognize the next bit (or several bits, if the code is focused on more than two signal states). Any sharp change in the signal - the so-called edge - can serve as a good indication for synchronizing the receiver with the transmitter.

When using sinusoids as a carrier signal, the resulting code has the property of self-synchronization, since changing the amplitude of the carrier frequency allows the receiver to determine the moment the input code appears.

Recognition and correction of distorted data is difficult to carry out using physical layer means, so most often this work is taken over by higher-lying protocols: channel, network, transport or application. On the other hand, error recognition at the physical layer saves time, since the receiver does not wait for the frame to be completely placed in the buffer, but discards it immediately when it recognizes erroneous bits within the frame.

The requirements for encoding methods are mutually contradictory, therefore each of the popular digital encoding methods discussed below has its own advantages and disadvantages compared to others.

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Potential code without returning to zero

In Fig. 2.16, and shows the previously mentioned potential encoding method, also called encoding without returning to zero (Non Return to Zero, NRZ). The last name reflects the fact that when transmitting a sequence of ones, the signal does not return to zero during the clock cycle (as we will see below, in other encoding methods a return to zero occurs in this case). The NRZ method is easy to implement, has good error recognition (due to two sharply different potentials), but does not have the property of self-synchronization. When transmitting a long sequence of ones or zeros, the signal on the line does not change, so the receiver is unable to determine from the input signal the moments in time when it is necessary to read the data again. Even with a high-precision clock generator, the receiver may make a mistake with the moment of data collection, since the frequencies of the two generators are never completely identical. Therefore, at high data rates and long sequences of ones or zeros, a small clock mismatch can lead to an error of a whole clock cycle and, accordingly, an incorrect bit value being read.

Another serious disadvantage of the NRZ method is the presence of a low-frequency component that approaches zero when transmitting long sequences of ones or zeros. Because of this, many communication channels do not provide

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Those that provide a direct galvanic connection between the receiver and the source do not support this type of coding. As a result, the NRZ code in its pure form is not used in networks. Nevertheless, its various modifications are used, which eliminate both the poor self-synchronization of the NRZ code and the presence of a constant component. The attractiveness of the NRZ code, which makes it worthwhile to improve it, is the fairly low fundamental frequency fo, which is equal to N/2 Hz, as was shown in the previous section. In other encoding methods, such as Manchester, the fundamental harmonic has a higher frequency.

Bipolar coding method with alternative inversion

One of the modifications of the NRZ method is the method bipolar coding with alternative inversion (Bipolar Alternate Mark Inversion, AMI). In this method (Fig. 2.16, 6) Three potential levels are used - negative, zero and positive. To encode a logical zero, a zero potential is used, and a logical one is encoded either by a positive potential or a negative one, with the potential of each new unit being opposite to the potential of the previous one.

The AMI code partially eliminates the DC and lack of self-synchronization problems inherent in the NRZ code. This occurs when transmitting long sequences of ones. In these cases, the signal on the line is a sequence of oppositely polarized pulses with the same spectrum as the NRZ code, transmitting alternating zeros and ones, that is, without a constant component and with a fundamental harmonic of N/2 Hz (where N is the bit rate of data transfer) . Long sequences of zeros are just as dangerous for the AMI code as for the NRZ code - the signal degenerates into a constant potential of zero amplitude. Therefore, the AMI code requires further improvement, although the task is simplified - all that remains is to deal with sequences of zeros.

In general, for different bit combinations on a line, using the AMI code results in a narrower signal spectrum than the NRZ code, and therefore higher line capacity. For example, when transmitting alternating ones and zeros, the fundamental harmonic fo has a frequency of N/4 Hz. The AMI code also provides some capabilities for recognizing erroneous signals. Thus, a violation of the strict alternation of signal polarity indicates a false pulse or the disappearance of a correct pulse from the line. A signal with incorrect polarity is called a prohibited signal (signal violation).

The AMI code uses not two, but three signal levels on the line. The additional layer requires an increase in transmitter power of approximately 3 dB to provide the same bit fidelity on the line, which is a common disadvantage of codes with multiple signal states compared to codes that distinguish only two states.

Potential code with inversion at one

There is code similar to AMI, but with only two signal levels. When transmitting a zero, it transmits the potential that was set in the previous cycle (that is, does not change it), and when transmitting a one, the potential is inverted to the opposite one. This code is called potential code with inversion at one

2.2. Methods for transmitting discrete data at the physical level 139

(Non Return to Zero with ones Inverted, NRZI). This code is convenient in cases where the use of a third signal level is highly undesirable, for example in optical cables, where two signal states are consistently recognized - light and darkness. Two methods are used to improve potential codes like AMI and NRZI. The first method is based on adding redundant bits containing logical ones to the source code. Obviously, in this case, long sequences of zeros are interrupted and the code becomes self-synchronizing for any transmitted data. The constant component also disappears, which means the signal spectrum narrows even more. But this method reduces the useful capacity of the line, since redundant units of user information are not carried. Another method is based on preliminary “mixing” of the initial information so that the probability of the appearance of ones and zeros on the line becomes close. Devices or blocks that perform such an operation are called scramblers(scramble - dump, disorderly assembly). When scrambling, a well-known algorithm is used, so the receiver, having received binary data, transmits it to descrambler, which restores the original bit sequence. In this case, excess bits are not transmitted over the line. Both methods refer to logical rather than physical coding, since they do not determine the shape of the signals on the line. They are studied in more detail in the next section.

Bipolar pulse code

In addition to potential codes, pulse codes are also used in networks, when the data is represented by a full pulse or part of it - an edge. The simplest case of this approach is bipolar pulse code, in which one is represented by a pulse of one polarity, and zero by another (Fig. 2.16, V). Each pulse lasts half a beat. Such code has excellent self-synchronizing properties, but a constant component may be present, for example, when transmitting a long sequence of ones or zeros. In addition, its spectrum is wider than that of potential codes. Thus, when transmitting all zeros or ones, the frequency of the fundamental harmonic of the code will be equal to N Hz, which is two times higher than the fundamental harmonic of the NRZ code and four times higher than the fundamental harmonic of the AMI code when transmitting alternating ones and zeros. Due to its too wide spectrum, the bipolar pulse code is rarely used.

Manchester code

In local networks, until recently, the most common encoding method was the so-called Manchester code(Fig. 2.16, d). It is used in Ethernet and Token Ring technologies.

The Manchester code uses a potential difference, that is, the edge of a pulse, to encode ones and zeros. With Manchester encoding, each measure is divided into two parts. Information is encoded by potential drops that occur in the middle of each clock cycle. A one is encoded by an edge from a low signal level to a high one, and a zero is encoded by a reverse edge. At the beginning of each clock cycle, an overhead signal drop may occur if you need to represent several ones or zeros in a row. Since the signal changes at least once per clock cycle of transmitting one bit of data, the Manchester code has good

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self-synchronizing properties. The bandwidth of the Manchester code is narrower than that of the bipolar pulse. It also has no DC component, and the fundamental harmonic in the worst case (when transmitting a sequence of ones or zeros) has a frequency of N Hz, and in the best case (when transmitting alternating ones and zeros) it is equal to N / 2 Hz, like AMI or NRZ On average, the bandwidth of the Manchester code is one and a half times narrower than that of the bipolar pulse code, and the fundamental harmonic fluctuates around the value of 3N/4. The Manchester code has another advantage over the bipolar pulse code. The latter uses three signal levels for data transmission, while the Manchester one uses two.

Potential code 2B1Q

In Fig. 2.16, d shows a potential code with four signal levels for encoding data. This is the code 2В1Q the name of which reflects its essence - every two bits (2B) are transmitted in one clock cycle by a signal having four states (1Q). Bit pair 00 corresponds to a potential of -2.5 V, bit pair 01 corresponds to a potential of -0.833 V, pair I corresponds to a potential of +0.833 V, and pair 10 corresponds to a potential of +2.5 V. With this coding method, additional measures are required to combat long sequences of identical bit pairs, since in this case the signal turns into a constant component. With random interleaving of bits, the signal spectrum is twice as narrow as that of the NRZ code, since at the same bit rate the clock duration is doubled. Thus, using the 2B1Q code, you can transfer data over the same line twice as fast as using the AMI or NRZI code. However, to implement it, the transmitter power must be higher so that the four levels are clearly distinguished by the receiver against the background of interference.

2.2.3. Logic coding

Logic coding is used to improve potential codes like AMI, NRZI or 2Q1B. Logic coding must replace long sequences of bits that lead to a constant potential with interspersed ones. As noted above, logical coding is characterized by two methods - redundant codes and scrambling.

Redundant codes

Redundant codes are based on breaking the original bit sequence into chunks, often called symbols. Each original character is then replaced with a new one that has more bits than the original. For example, the 4V/5V logic code used in FDDI and Fast Ethernet technologies replaces the original 4-bit symbols with 5-bit symbols. Since the resulting symbols contain redundant bits, the total number of bit combinations in them is greater than in the original ones. Thus, in a 4B/5B code, the resulting symbols can contain 32 bit combinations, while the original symbols contain only 16. Therefore, in the resulting code, you can select 16 such combinations that do not contain a large number of zeros, and count the rest prohibited codes (code violation). In addition to eliminating the constant component and giving the code self-synchronizing properties, redundant codes allow

2.2. Methods for transmitting discrete data at the physical level 141

the receiver can recognize the corrupted bits. If the receiver receives an illegal code, it means that the signal has been distorted on the line.

The correspondence between the source and result codes 4B/5B is presented below.

The 4B/5B code is then transmitted over the line using physical encoding using one of the potential encoding methods, which is sensitive only to long sequences of zeros. The 4B/5B code symbols, 5 bits long, guarantee that no matter how they are combined, more than three zeros in a row cannot appear on the line.

The letter B in the name of the code means that the elementary signal has 2 states - from the English binary - binary. There are also codes with three signal states, for example, in the 8B/6T code, to encode 8 bits of source information, a code of 6 signals is used, each of which has three states. The redundancy of the 8B/6T code is higher than that of the 4B/5B code, since for 256 source codes there are 3 6 =729 resulting symbols.

Using a lookup table is a very simple operation, so this approach does not add complexity to the network adapters and interface blocks of switches and routers.

To ensure a given line capacity, a transmitter using a redundant code must operate at an increased clock frequency. So, to transmit 4B/5B codes at a speed of 100 Mb/s, the transmitter must operate at a clock frequency of 125 MHz. In this case, the spectrum of the signal on the line expands compared to the case when a pure, non-redundant code is transmitted along the line. Nevertheless, the spectrum of the redundant potential code turns out to be narrower than the spectrum of the Manchester code, which justifies the additional stage of logical coding, as well as the operation of the receiver and transmitter at an increased clock frequency.

Scrambling

Shuffling the data with a scrambler before passing it onto the line using a potential code is another way of logical encoding.

Scrambling methods involve bit-by-bit calculation of the resulting code based on the bits of the source code and the bits of the resulting code obtained in previous clock cycles. For example, a scrambler might implement the following relation:

Bi - Ai 8 Bi-z f Bi. 5,

where bi is the binary digit of the resulting code received at the i-th clock cycle of the scrambler, ai is the binary digit of the source code arriving at the i-th clock cycle at

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scrambler input, B^3 and B t .5 - binary digits of the resulting code obtained in the previous cycles of the scrambler, respectively 3 and 5 clock cycles earlier than the current clock cycle, 0 - exclusive OR operation (addition modulo 2).

For example, for the original sequence 110110000001, the scrambler will give the following result code:

bi = ai - 1 (the first three digits of the resulting code will coincide with the original one, since there are no necessary previous digits yet)

Thus, the output of the scrambler will be the sequence 110001101111, which does not contain the sequence of six zeros present in the source code.

After receiving the resulting sequence, the receiver transmits it to the descrambler, which restores the original sequence based on the inverse relationship:

Different scrambling algorithms differ in the number of terms that give the resulting code digit and the shift between the terms. Thus, in ISDN networks, when transmitting data from the network to a subscriber, a transformation with shifts of 5 and 23 positions is used, and when transmitting data from a subscriber to the network, it is used with shifts of 18 and 23 positions.

There are simpler methods of dealing with sequences of units, also classified as scrambling.

To improve the Bipolar AMI code, two methods are used, based on artificially distorting the sequence of zeros with illegal characters.

In Fig. Figure 2.17 shows the use of the B8ZS (Bipolar with 8-Zeros Substitution) method and the HDB3 (High-Density Bipolar 3-Zeros) method to adjust the AMI code. The source code consists of two long sequences of zeros: in the first case - from 8, and in the second - from 5.

The B8ZS code only corrects sequences consisting of 8 zeros. To do this, after the first three zeros, instead of the remaining five zeros, he inserts five digits: V-1*-0-V-1*. V here denotes a signal of a unit that is prohibited for a given polarity cycle, that is, a signal that does not change the polarity of the previous unit, 1* is a signal of a unit of correct polarity, and the asterisk sign marks that

2.2. Methods for transmitting discrete data at the physical level 143

The fact is that in the source code in this cycle there was not a unit, but a zero. As a result, at 8 clock cycles the receiver observes 2 distortions - it is very unlikely that this happened due to line noise or other transmission failures. Therefore, the receiver considers such violations to be an encoding of 8 consecutive zeros and, after reception, replaces them with the original 8 zeros. The B8ZS code is constructed in such a way that its constant component is zero for any sequence of binary digits.

The HDB3 code corrects any four consecutive zeros in the original sequence. The rules for generating the HDB3 code are more complex than the B8ZS code. Every four zeros are replaced by four signals, in which there is one V signal. To suppress the DC component, the polarity of the V signal is alternated in successive replacements. In addition, two patterns of four-cycle codes are used for replacement. If before replacement the source code contained an odd number of ones, then the OOOV sequence is used, and if the number of ones was even, the sequence 1*OOV is used.

Improved candidate codes have a fairly narrow bandwidth for any sequences of ones and zeros that occur in the transmitted data. In Fig. Figure 2.18 shows the spectra of signals of different codes obtained when transmitting arbitrary data, in which various combinations of zeros and ones in the source code are equally probable. When plotting the graphs, the spectrum was averaged over all possible sets of initial sequences. Naturally, the resulting codes may have a different distribution of zeros and ones. From Fig. 2.18 shows that the potential NRZ code has a good spectrum with one drawback - it has a constant component. Codes obtained from potential by logic coding have a narrower spectrum than Manchester, even at an increased clock frequency (in the figure, the spectrum of the 4B/5B code should approximately coincide with the B8ZS code, but it is shifted

144 Chapter2 Fundamentals of discrete data transmission

to the region of higher frequencies, since its clock frequency is increased by 1/4 compared to other codes). This explains the use of potential redundant and scrambled codes in modern technologies like FDDI, Fast Ethernet, Gigabit Ethernet, ISDN, etc. instead of Manchester and bipolar pulse coding.

2.2.4. Discrete modulation of analog signals

One of the main trends in the development of network technologies is the transmission of both discrete and analog data in one network. Sources of discrete data are computers and other computing devices, and sources of analog data are devices such as telephones, video cameras, audio and video playback equipment. In the early stages of solving this problem in territorial networks, all types of data were transmitted in analog form, while computer data that was discrete in nature was converted into analog form using modems.

However, as technology for collecting and transmitting analog data developed, it became clear that transmitting it in analog form does not improve the quality of the data received at the other end of the line if it was significantly distorted during transmission. The analog signal itself does not give any indication that distortion has occurred or how to correct it, since the signal shape can be any, including the one detected by the receiver. Improving the quality of lines, especially territorial ones, requires enormous effort and investment. Therefore, analog technology for recording and transmitting sound and images has been replaced by digital technology. This technique uses so-called discrete modulation of original time-continuous analog processes.

Discrete modulation methods are based on the sampling of continuous processes both in amplitude and time (Fig. 2.19). Let's look at the principles of spark modulation using an example pulse code modulation, PCM (Pulse Amplitude Modulation, PAM), which is widely used in digital telephony.

The amplitude of the original continuous function is measured with a given period - due to this, discretization in time occurs. Then each measurement is represented as a binary number of a certain bit depth, which means discretization by function values ​​- a continuous set of possible amplitude values ​​is replaced by a discrete set of its values. A device that performs a similar function is called analog-to-digital converter (ADC). After this, the measurements are transmitted over communication channels in the form of a sequence of ones and zeros. In this case, the same coding methods are used as in the case of transmitting initially discrete information, that is, for example, methods based on the B8ZS or 2B1Q code.

On the receiving side of the line, the codes are converted into the original bit sequence, and special equipment called digital-to-analog converter (DAC), demodulates the digitized amplitudes of a continuous signal, restoring the original continuous time function.

Discrete modulation is based on Nyquist-Kotelnikov mapping theory. According to this theory, an analog continuous-time function, transmitted as a sequence of its time-discrete values, can be accurately reconstructed if the sampling rate was two or more times higher than the frequency of the highest harmonic spectrum of the original function.

If this condition is not met, then the restored function will differ significantly from the original one.

The advantage of digital methods of recording, reproducing and transmitting analog information is the ability to control the accuracy of data read from a medium or received via a communication line. To do this, you can use the same methods that are used for computer data (and are discussed in more detail below) - calculating a checksum, retransmitting corrupted frames, using self-correcting codes.

For high-quality voice transmission, the PCM method uses a quantization frequency of the amplitude of sound vibrations of 8000 Hz. This is due to the fact that in analog telephony the range from 300 to 3400 Hz was chosen for voice transmission, which conveys all the basic harmonics of the interlocutors quite well. According to Nyquist-Koteltkov theorem for high-quality voice transmission

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it is enough to choose a sampling frequency that is twice the highest harmonic of the continuous signal, that is, 2 x 3400 = 6800 Hz. The actually chosen sampling rate of 8000 Hz provides some margin of quality. The PCM method typically uses 7 or 8 bits of code to represent the amplitude of a single sample. Accordingly, this gives 127 or 256 gradations of the sound signal, which is quite sufficient for high-quality voice transmission. When using the PCM method, a single voice channel requires a throughput of 56 or 64 Kbps, depending on how many bits each sample is represented by. If used for these purposes

7 bits, then with a measurement transmission frequency of 8000 Hz we get:

8000 x 7 = 56000 bps or 56 Kbps; and for the case of 8 bits:

8000 x 8 - 64000 bps or 64 Kbps.

The standard is a 64 Kbps digital channel, also called elementary channel of digital telephone networks.

Transmission of a continuous signal in discrete form requires networks to strictly adhere to a time interval of 125 μs (corresponding to a sampling frequency of 8000 Hz) between adjacent measurements, that is, it requires synchronous data transmission between network nodes. If the synchronization of arriving measurements is not maintained, the original signal is restored incorrectly, which leads to distortion of voice, image or other multimedia information transmitted over digital networks. Thus, a synchronization distortion of 10 ms can lead to an “echo” effect, and shifts between measurements of 200 ms lead to loss of recognition of spoken words. At the same time, the loss of one measurement, while maintaining synchronicity between the other measurements, has virtually no effect on the reproduced sound. This occurs due to smoothing devices in digital-to-analog converters, which are based on the inertia property of any physical signal - the amplitude of sound vibrations cannot instantly change by a large amount.

The quality of the signal after the DAC is affected not only by the synchronism of measurements arriving at its input, but also by the sampling error of the amplitudes of these measurements.

8 of the Nyquist-Kotelnikov theorem assumes that the amplitudes of the function are measured accurately, at the same time, the use of binary numbers with a limited bit capacity to store them somewhat distorts these amplitudes. Accordingly, the reconstructed continuous signal is distorted, which is called sampling noise (in amplitude).

There are other discrete modulation techniques that can represent voice measurements in a more compact form, such as a sequence of 4-bit or 2-bit numbers. In this case, one voice channel requires less bandwidth, for example 32 Kbps, 16 Kbps or even less. Since 1985, a CCITT voice coding standard called Adaptive Differential Pulse Code Modulation (ADPCM) has been used. ADPCM codes are based on finding the differences between successive voice measurements, which are then transmitted over the network. ADPCM code uses 4 bits to store one difference and transmits voice at 32 Kbps. A more modern method, Linear Predictive Coding (LPC), samples the original function more infrequently, but uses methods to predict the direction of change in signal amplitude. Using this method, you can reduce the voice transmission speed to 9600 bps.

2.2. Methods for transmitting discrete data at the physical level 147

Continuous data presented in digital form can be easily transmitted over a computer network. To do this, it is enough to place several measurements in a frame of some standard network technology, provide the frame with the correct destination address and send it to the recipient. The recipient must extract measurements from the frame and submit them at a quantization frequency (for voice, at a frequency of 8000 Hz) to a digital-to-analog converter. As the next frames with voice measurements arrive, the operation must be repeated. If the frames arrive synchronously enough, the voice quality can be quite high. However, as we already know, frames in computer networks can be delayed both in end nodes (while waiting for access to the shared medium) and in intermediate communication devices - bridges, switches and routers. Therefore, voice quality when transmitted digitally over computer networks is usually poor. For high-quality transmission of digitized continuous signals - voice, image - today special digital networks are used, such as ISDN, ATM, and digital television networks. Nevertheless, the transmission of intracorporate telephone conversations today is characterized by frame relay networks, the frame transmission delays of which fall within acceptable limits.

2.2.5. Asynchronous and synchronous transmission

When exchanging data at the physical layer, the unit of information is a bit, so the physical layer always maintains bit synchronization between the receiver and transmitter.

The data link layer operates on frames of data and provides frame-level synchronization between the receiver and transmitter. The receiver's responsibilities include recognizing the beginning of the first byte of the frame, recognizing the boundaries of the frame's fields, and recognizing the end of the frame.

Usually it is enough to ensure synchronization at these two levels - bit and frame - so that the transmitter and receiver can ensure a stable exchange of information. However, if the quality of the communication line is poor (usually this applies to telephone dial-up channels), additional means of synchronization at the byte level are introduced to reduce the cost of equipment and increase the reliability of data transmission.

This mode of operation is called asynchronous or start-stop. Another reason for using this mode of operation is the presence of devices that generate bytes of data at random times. This is how the keyboard of a display or other terminal device works, from which a person enters data for processing by a computer.

In asynchronous mode, each byte of data is accompanied by special “start” and “stop” signals (Fig. 2.20, A). The purpose of these signals is, firstly, to notify the receiver of the arrival of data and, secondly, to give the receiver enough time to perform some synchronization-related functions before the next byte arrives. The start signal has a duration of one clock interval, and the stop signal can last one, one and a half, or two clock periods, so it is said that one, one and a half, or two bits are used as the stop signal, although these signals do not represent user bits.

The described mode is called asynchronous because each byte can be slightly shifted in time relative to the bit clocks of the previous one

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byte. This asynchronous transmission of bytes does not affect the correctness of the received data, since at the beginning of each byte additional synchronization of the receiver with the source occurs due to the “start” bits. More “loose” time tolerances determine the low cost of asynchronous system equipment.

In synchronous transmission mode, there are no start-stop bits between each pair of bytes. User data is collected into a frame, which is preceded by synchronization bytes (Fig. 2.20, b). A sync byte is a byte containing a known code, such as 0111110, that notifies the receiver of the arrival of a data frame. Upon receiving it, the receiver must enter byte synchronization with the transmitter, that is, correctly understand the beginning of the next byte of the frame. Sometimes multiple sync bytes are used to provide more reliable synchronization between the receiver and transmitter. Since when transmitting a long frame the receiver may have problems with bit synchronization, in this case self-synchronizing codes are used.

» When transmitting discrete data over a narrowband voice-frequency channel used in telephony, the most suitable methods are analog modulation, in which the carrier sinusoid is modulated by the original sequence of binary digits. This operation is carried out by special devices - modems.

* For low-speed data transmission, a change in the frequency of the carrier sinusoid is used. Higher-speed modems operate using combined quadrature amplitude modulation (QAM) methods, which are characterized by 4 levels of carrier sinusoid amplitude and 8 levels of phase. Not all of the possible 32 combinations of the QAM method are used for data transmission; prohibited combinations make it possible to recognize distorted data at the physical level.

* On broadband communication channels, potential and pulse coding methods are used, in which data is represented by different levels of constant signal potential or polarities of a pulse or its front.

* When using potential codes, the task of synchronizing the receiver with the transmitter is of particular importance, since when long sequences of zeros or ones are transmitted, the signal at the receiver input does not change and it is difficult for the receiver to determine the moment of picking up the next data bit.

___________________________________________2.3. Data Link Layer Transmission Methods _______149

* The simplest potential code is the non-return-to-zero (NRZ) code, however it is not self-clocking and produces a DC component.

» The most popular pulse code is the Manchester code, in which the information is carried by the direction of the signal drop in the middle of each clock cycle. The Manchester code is used in Ethernet and Token Ring technologies.

» To improve the properties of a potential NRZ code, logical coding methods are used that eliminate long sequences of zeros. These methods are based on:

On the introduction of redundant bits into the source data (4B/5B type codes);

Scrambling of source data (2B1Q type codes).

» Improved potential codes have a narrower spectrum than pulse codes, so they are used in high-speed technologies such as FDDI, Fast Ethernet, Gigabit Ethernet.

When transmitting discrete data over communication channels, two main types of physical coding are used - based on sinusoidal carrier signal and based on a sequence of rectangular pulses. The first method is often also called modulation or analog modulation, emphasizing the fact that encoding is carried out by changing the parameters of the analog signal. The second method is usually called digital encoding. These methods differ in the width of the spectrum of the resulting signal and the complexity of the equipment required for their implementation.
Analog modulation used for transmitting discrete data over channels with a narrow frequency band, a typical representative of which is the voice-frequency channel provided to users of public telephone networks. A typical amplitude-frequency response of a voice-frequency channel is shown in Fig. 2.12. This channel transmits frequencies in the range from 300 to 3400 Hz, so its bandwidth is 3100 Hz. A device that performs the functions of carrier sinusoid modulation on the transmitting side and demodulation on the receiving side is called a modem (modulator - demodulator).
Analog modulation methods
Analog modulation is a physical encoding method in which information is encoded by changing the amplitude, frequency or phase of a sinusoidal carrier signal.
The diagram (Fig. 2.13, a) shows a sequence of bits of the original information, represented by high-level potentials for a logical unit and a zero-level potential for logical zero. This encoding method is called potential code, which is often used when transferring data between computer units.
With amplitude modulation (Fig. 2.13, b), one level of the amplitude of the carrier frequency sinusoid is selected for a logical unit, and another for logical zero. This method is rarely used in its pure form in practice due to low noise immunity, but is often used in combination with another type of modulation - phase modulation.
With frequency modulation (Fig. 2.13, c), the values ​​0 and 1 of the source data are transmitted by sinusoids with different frequencies - f0 and f1. This modulation method does not require complex circuitry in modems and is typically used in low-speed modems operating at 300 or 1200 bps.
With phase modulation, data values ​​0 and 1 correspond to signals of the same frequency, but with different phases, for example 0 and 180 degrees or 0,90,180 and 270 degrees.
High-speed modems often use combined modulation methods, usually amplitude combined with phase.
When using rectangular pulses to transmit discrete information, it is necessary to choose a coding method that simultaneously achieves several goals:
· had the smallest spectrum width of the resulting signal at the same bit rate;
· ensured synchronization between the transmitter and receiver;
· had the ability to recognize mistakes;
· had a low cost of sale.
A narrower spectrum of signals allows one and the same line (with the same bandwidth) to achieve a higher data transfer rate. In addition, the signal spectrum is often required to have no DC component, that is, the presence of a DC current between the transmitter and receiver. In particular, the use of various transformer galvanic isolation circuits prevents the passage of direct current.
Synchronization of the transmitter and receiver is necessary so that the receiver knows exactly at what point in time it is necessary to read new information from the communication line.
Recognition and correction of distorted data is difficult to carry out using physical layer means, so most often this work is taken over by higher-lying protocols: channel, network, transport or application. On the other hand, error recognition at the physical layer saves time, since the receiver does not wait for the frame to be completely placed in the buffer, but discards it immediately when it recognizes erroneous bits within the frame.
The requirements for encoding methods are mutually contradictory, therefore each of the popular digital encoding methods discussed below has its own advantages and disadvantages compared to others.

Plan

Theoretical basis of data transmission

Fourier series

Managed media

Magnetic media

twisted pair

Coaxial cable

Fiber optics

A special connector can be attached to the end of the cable, with which the cable is inserted into an optical socket. The loss is 10-20% of the light intensity, but it makes it easy to change the system configuration.
Splicing - two neatly cut ends of the cable are laid next to each other and clamped with a special coupling. Improved light transmission is achieved by aligning the cable ends. Loss - 10% of light power.
Fusion. There is practically no loss.

Table 1.
Comparison table of LED and semiconductor laser usage
The receiving end of an optical cable is a photodiode that generates an electrical pulse when light hits it.

Comparative characteristics of fiber optic cable and copper wire.

High speed.
Less signal attenuation, output from fewer repeaters (one per 50 km, not 5)
Inert to external electromagnetic radiation, chemically neutral.
Lighter in weight. 1000 copper twisted pairs with a length of 1 km weighs about 8000 kg. A pair of fiber optic cables weighs only 100 kg with greater bandwidth
Low installation costs

Complexity and competence during installation.
Fragility
More expensive than copper.
transmission in simplex mode, a minimum of 2 wires are required between networks.

Wireless communication

Electromagnetic spectrum

Radio communication

Microwave communications

Infrared and millimeter waves

Communications satellites

Public Switched Telephone Network

Telephone system structure

Local communication lines: modems, ADSL, wireless communication

amplitude modulation - 2 different signal amplitudes are used (for 0 and 1),
frequency - several different signal frequencies are used (for 0 and 1),
phase - phase shifts are used when transitioning between logical units (0 and 1). Shear angles - 45, 135, 225, 180.

Digital subscriber lines. xDSL technology.

POTS - regular telephone network range;
outgoing range;
incoming range.

DSLAM - DSL access multiplexer;
NID is a network interface device that separates the ownership of the telephone company and the subscriber.
Splitter (splitter) - a frequency divider that separates the POTS band and ADSL data.

Lines and seals

Switching

Circuit switching

Packet switching

Physical layer deals with the actual transmission of raw bits over

communication channel.

Data transfer in computer networks from one computer to another is carried out sequentially, bit by bit. Physically, data bits are transmitted over data links in the form of analog or digital signals.

The set of means (communication lines, data transmission and reception equipment) used to transmit data in computer networks is called a data transmission channel. Depending on the form of transmitted information, data transmission channels can be divided into analog (continuous) and digital (discrete).

Since data transmission and reception equipment works with data in discrete form (i.e., discrete electrical signals correspond to ones and zeros of data), when transmitting them through an analog channel, conversion of discrete data to analog (modulation) is required.

When receiving such analog data, an inverse conversion is required - demodulation. Modulation/demodulation – processes of converting digital information into analog signals and vice versa. During modulation, information is represented by a sinusoidal signal of the frequency that the data transmission channel transmits well.

Modulation methods include:

· amplitude modulation;

· frequency modulation;

· phase modulation.

When transmitting discrete signals via a digital data channel, coding is used:

· potential;

· pulsed.

Thus, potential or pulse coding is used on high quality channels, and modulation based on sine waves is preferable in cases where the channel introduces severe distortion into the transmitted signals.

Modulation is typically used in wide area networks to transmit data over analog telephone links, which were designed to carry voice in analog form and are therefore not well suited for direct pulse transmission.

Depending on the synchronization methods, data transmission channels of computer networks can be divided into synchronous and asynchronous. Synchronization is necessary so that the sending data node can transmit some signal to the receiving node so that the receiving node knows when to start receiving incoming data.

Synchronous data transmission requires an additional communication line to transmit clock pulses. The transmission of bits by the transmitting station and their reception by the receiving station is carried out at the moments of the appearance of clock pulses.

For asynchronous data transfer, no additional communication line is required. In this case, data transmission is carried out in fixed-length blocks (bytes). Synchronization is carried out by additional bits (start bits and stop bits), which are transmitted before and after the transmitted byte.

When exchanging data between computer network nodes, three data transfer methods are used:

simplex (unidirectional) transmission (television, radio);

half-duplex (reception/transmission of information is carried out alternately);

duplex (bidirectional), each node simultaneously transmits and receives data (for example, telephone conversations).