1. Principles of constructing wireless telecommunication systems

1.1 Architecture of cellular communication systems.

1.2 Subscriber service by the network.

1.3 Methods for separating subscribers into cellular communications

1.4 DECT standard for communication.

1.5 Bluetooth, Wi-Fi standards (802.11, 802.16).

2. Complex signal systems for telecommunication systems.

2.1 Signal spectra

2.2 Correlation properties of signals

2.3 Types of complex signals

2.4 Derived signal systems

3. Modulation of complex signals

3.1 Geometric representation of signals

3.2 Methods of phase manipulation of signals (PM2, PM4, OFM).

3.3 Modulation with minimum frequency shift.

3.4 Quadrature modulation and its characteristics (QPSK, QAM).

3.5 Implementation of quadrature modems.

4. Characteristics of signal reception in telecommunication systems.

4.1 Probability of errors in distinguishing M known signals

4.2 Probability of errors in distinguishing M fluctuating signals.

4.3 Calculation of errors in distinguishing M signals from unknowns

non-energy parameters.

4.4 Comparison of synchronous and asynchronous communication systems.

5. Conclusion.

6. References

1. Principles of constructing wireless telecommunication systems

1.1 Architecture of cellular communication systems

A cellular communication system is a complex and flexible technical system that allows for great diversity, both in configuration options and in the range of functions performed. An example of the complexity and flexibility of the system is that it can transmit both speech and other types of information, in particular text messages and computer data. In terms of voice transmission, in turn, conventional two-way telephone communication, multi-way telephone communication (the so-called conference call - with more than two subscribers participating in a conversation at the same time), and voice mail can be implemented. When organizing a regular two-way telephone conversation starting with a call, auto-redial, call waiting, and call forwarding modes are possible.

A cellular communication system is built in the form of a collection of cells, or cells, covering the service area, for example, the territory of a city with suburbs. Cells are usually schematically depicted in the form of regular equal-sized hexagons (Fig. 1.1.), which, due to their similarity to a honeycomb, was the reason for calling the system cellular. The cellular, or cellular, structure of the system is directly related to the principle of frequency reuse - the basic principle of the cellular system, which determines the efficient use of the allocated frequency range and high system capacity.

Rice. 1.1. Cells (cells) of the system covering the entire serviced territory.

In the center of each cell there is a base station that serves all mobile stations (subscriber radiotelephone devices) within its cell (Fig. 1.2.). When a subscriber moves from one cell to another, his service is transferred from one base station to another. All base stations of the system, in turn, are connected to the switching center, from which there is access to the Interconnected Communications Network (ICN) of Russia, in particular, if it happens in a city, there is access to the regular city wired telephone network.

Rice. 1.2. One cell with a base station in the center serving all mobile stations in the cell.

In Fig. 1.3. a functional diagram corresponding to the described structure is shown.

Rice. 1.3. Simplified functional diagram of a cellular communication system: BS – base station; PS – mobile station (subscriber radiotelephone device).

In reality, cells never have a strict geometric shape. The actual boundaries of the cells have the form of irregular curves, depending on the conditions of propagation and attenuation of radio waves, i.e. on the terrain, the nature and density of vegetation and buildings, and similar factors. Moreover, the boundaries of cells are generally not clearly defined, since the boundary of handover of a mobile station from one cell to another can shift within certain limits with changes in the propagation conditions of radio waves and depending on the direction of movement of the mobile station. In the same way, the position of the base station only approximately coincides with the center of the cell, which is also not so easy to determine unambiguously if the cell has an irregular shape. If the base stations use directional (not isotropic in the horizontal plane) antennas, then the base stations actually end up at the cell boundaries. Further, a cellular communication system may include more than one switching center, which may be due to system evolution or switch capacity limitations. It is possible, for example, to have a structure of the type shown in Fig. 1.4. – with several switching centers, one of which can be conventionally called the “head” or “master”.

Rice. 1.4. Cellular communication system with two switching centers.

Let's consider a mobile station - the simplest element of the cellular communication system in terms of functionality and design, moreover, it is the only element of the system that is actually accessible to the user.

The block diagram of the mobile station is shown in Fig. 1.5. It includes:

Control block;

Transceiver unit;

Antenna block.

Rice. 1.5. Block diagram of a mobile station (subscriber radiotelephone device).

The transceiver block, in turn, includes a transmitter, receiver, frequency synthesizer and logic block.

The antenna unit is the simplest in composition: it includes the antenna itself and the transmit-receive switch. The latter for a digital station can be an electronic switch that connects an antenna either to the output of the transmitter or to the input of the receiver, since a mobile station of a digital system never operates to receive and transmit simultaneously.

The control unit includes a handset - microphone and speaker, keyboard and display. The keyboard (a dialing field with numeric and function keys) is used to dial the telephone number of the called subscriber, as well as commands that determine the operating mode of the mobile station. The display is used to show various information provided by the device and operating mode of the station.

The transceiver unit is much more complex.

The transmitter includes:

Analog-to-digital converter (ADC) – converts the signal from the microphone output into digital form, and all subsequent processing and transmission of the speech signal is carried out in digital form, up to the reverse digital-to-analog conversion;

The speech encoder encodes the speech signal - converts the digital signal according to certain laws in order to reduce its redundancy, i.e. in order to reduce the amount of information transmitted over the communication channel;

Channel encoder – adds additional (redundant) information to the digital signal received from the output of the speech encoder, designed to protect against errors when transmitting the signal over the communication line; for the same purpose, information undergoes a certain repackaging (multiplication); in addition, the channel encoder includes control information coming from the logical block into the transmitted signal;

Modulator – transfers the information of the encoded video signal to the carrier frequency.

The composition of the receiver basically corresponds to the transmitter, but with the reverse functions of its constituent blocks:

The demodulator extracts an encoded video signal carrying information from the modulated radio signal;

The channel decoder extracts control information from the input stream and directs it to the logical block; the received information is checked for errors, and the identified errors are corrected if possible; Before further processing, the received information is subjected to reverse (relative to the encoder) repacking;

The speech decoder restores the speech signal received from the channel decoder, converting it into a natural form, with its inherent redundancy, but in digital form;

A digital-to-analog converter (DAC) converts the received speech signal into analog form and supplies it to the speaker output;

The equalizer serves to partially compensate for signal distortion due to multipath propagation; Essentially, it is an adaptive filter, adjusted according to the training sequence of symbols included in the transmitted information; The equalizer block is not, generally speaking, functionally necessary and in some cases may be absent.

The combination of encoder and decoder is sometimes called codec.

In addition to the transmitter and receiver, the transceiver unit includes a logic unit and a frequency synthesizer. Logic block- this is, in fact, a microcomputer with its own RAM and permanent memory, which controls the operation of the mobile station. The synthesizer is a source of oscillations of the carrier frequency used to transmit information over a radio channel. The presence of a local oscillator and a frequency converter is due to the fact that different parts of the spectrum are used for transmission and reception.

The block diagram of the base station is shown in Fig. 1.6.

Rice. 1.6. Block diagram of a base station.

The presence of several receivers and the same number of transmitters allows simultaneous operation on several channels with different frequencies.

Receivers and transmitters of the same name have common tunable reference oscillators, ensuring their coordinated tuning when moving from one channel to another. To ensure simultaneous operation of N receivers per receiving antenna and N transmitters per transmitting antenna, a power divider with N outputs is installed between the receiving antenna and the receivers, and a power adder with N inputs is installed between the transmitters and the transmitting antenna.

The receiver and transmitter have the same structure as in the mobile station, except that there is no DAC or ADC because both the transmitter input and the receiver output are digital.

The communication line interface unit packs the information transmitted over the communication line to the switching center and unpacks the information received from it.

The base station controller, which is a fairly powerful and sophisticated computer, provides control over the operation of the station, as well as monitoring the performance of all its blocks and nodes.

The switching center is the brain center and at the same time the control center of the cellular communication system, to which information flows from all base stations are closed and through which access to other communication networks is provided - a landline telephone network, long-distance communication networks, satellite communications, and other cellular networks.

The block diagram of the switching center is shown in Fig. 1.7. The switch switches information flows between the corresponding communication lines. It can, in particular, direct the flow of information from one base station to another, or from a base station to a fixed communication network, or vice versa.

The switch is connected to communication lines through appropriate communication controllers that perform intermediate processing (packing/unpacking, buffer storage) of information flows. General control of the operation of the switching center and the system as a whole is carried out from a central controller, which has powerful mathematical support. The work of the switching center requires the active participation of operators, therefore the center includes the appropriate terminals, as well as means for displaying and recording (documenting) information. The operator enters data about subscribers and the conditions of their service, as well as initial data on the operating modes of the system.

Rice. 1.7. Block diagram of a switching center.

Important elements of the system are databases - home register, guest register, authentication center, equipment register. The home register contains information about all subscribers registered in this system and the types of services that can be provided to them. Here the location of the subscriber is recorded to organize his call, and the services actually provided are recorded. The guest register contains approximately the same information about subscribers - guests (roamers), i.e. about subscribers registered in another system, but currently using cellular communication services in this system. The Authentication Center provides subscriber authentication and message encryption procedures. The equipment register, if it exists, contains information about the mobile stations in operation regarding their serviceability and authorized use.

1.2 Subscriber service by the network

Interface is a system of signals through which cellular communication system devices connect to each other. Each cellular standard uses several interfaces (different in different standards).

Of all the interfaces used in cellular communications, one occupies a special place - this is the exchange interface between the mobile and base stations. It is called the air interface. The over-the-air interface is necessarily used in any cellular communication system, with any configuration and in the only option possible for its cellular communication standard.

The air interface of the D-AMPS system of the IS-54 standard is comparatively simple (Fig. 1.8.).

A traffic channel is a channel for transmitting voice or data. The transmission of information in the traffic channel is organized in successive frames with a duration of 40 ms. Each frame consists of six time intervals – slots; The slot duration (6.67 ms) corresponds to 324 bits. With full-rate coding, two slots are allocated for one speech channel in each frame, i.e. A 20-ms speech segment is packed into one slot, which is one-third as long. With half-rate coding, one slot in the frame is allocated to one speech channel, i.e. The packaging of the speech signal is twice as dense as with full-rate encoding.

Fig.1.8. Frame and slot structure of the D-AMPS system (traffic channel; IS-54 standard): Data – speech information; Sync(Sc) – synchronizing (training) sequence; SACCH – slow alignment control channel information; CDVCC(CC) – encoded digital color confirmation code; G – protective form; R – transmitter pulse front interval; V,W,X,Y – hexadecimal zeros; Res – reserve.

A slot has a slightly different structure in the forward traffic channel - from the base station to the mobile station and in the reverse traffic channel - from the mobile station to the base station. In both cases, 260 bits are allocated for speech transmission. Another 52 bits are occupied by control and auxiliary information. It includes: a 28-bit training sequence used to identify a slot within a frame, synchronize the slot in time and adjust the equalizer; 12-bit SACCH signaling (monitoring and control) message; A 12-bit coded digital color code (CDVCC) field that serves to identify the mobile station when its signal is received by the base station (the code is assigned by the base station individually for each channel, i.e., for each mobile station and is relayed by the latter back to the base station).

The remaining 12 bits in the forward channel are not used (reserve), and in the reverse channel they serve as a guard interval, during which no useful information is transmitted.

At the initial stage of communication establishment, a shortened slot is used in which the synchronization sequence and CDVCC code are repeated many times, separated by zero numbers of varying lengths. At the end of the shortened slot there is an additional protective form. The mobile station transmits shortened slots until the base station selects the required time delay, determined by the distance of the mobile station from the base station.

There are several communication channels: frequency, physical and logical.

A frequency channel is a frequency band allocated for transmitting information from one communication channel. Several physical channels can be placed in one frequency channel, for example, in the TDMA method.

A physical channel in a time division multiple access (TDMA) system is a time slot with a specific number in the sequence of air interface frames.

Logical channels are divided by the type of information transmitted in the physical channel to the traffic channel and control channel. The control channel carries signaling information, including control information and equipment status monitoring information, and the traffic channel carries speech and data.

(Traffic is a collection of messages transmitted over a communication line.)

Let us consider the operation of a mobile station within one cell of its (“home”) system, without handover. In this case, four stages can be distinguished in the operation of the mobile station, which correspond to four operating modes:

Power on and initialization;

Standby mode;

Communication (call) mode;

Communication mode (telephone conversation).

After turning on the mobile station, initialization is performed - initial start. During this stage, the mobile station is configured to work as part of the system - according to signals regularly transmitted by the base stations via the corresponding control channels, after which the mobile station goes into standby mode.

While in idle mode, the mobile station monitors:

Changes in system information - these changes can be associated both with changes in the operating mode of the system and with movements of the mobile station itself;

System commands - for example, a command to confirm its functionality;

Receiving a call from the system;

Initializing a call from your own subscriber.

In addition, the mobile station can periodically, for example once every 10...15 minutes, confirm its functionality by transmitting corresponding signals to the base station. In the switching center, for each of the switched-on mobile stations, the cell in which it is “registered” is fixed, which facilitates the organization of the procedure for calling a mobile subscriber.

If the system receives a call to the number of a mobile subscriber, the switching center routes this call to the base station of the cell in which the mobile station is “registered”, or to several base stations in the vicinity of this cell - taking into account the possible movement of the subscriber in the time that has elapsed since last “registration”, and the base stations transmit it over the appropriate calling channels. The mobile station in idle mode receives and answers the call through its base station, simultaneously transmitting the data necessary for the authentication procedure. If the authentication result is positive, a traffic channel is assigned and the corresponding frequency channel number is reported to the mobile station. The mobile station is tuned to a dedicated channel and, together with the base station, performs the necessary steps to prepare a communication session. At this stage, the mobile station tunes to a given slot number in the frame, clarifies the time delay, adjusts the level of emitted power, etc. The choice of time delay is made for the purpose of temporal coordination of slots in the frame when organizing communications with mobile stations located at different distances from the base station. In this case, the time delay of the packet transmitted by the mobile station is adjusted according to the commands of the base station.

The base station then issues a ringing message which is acknowledged by the mobile station and the caller is able to hear the ringing tone. When the called party answers the call, the mobile station issues a request to terminate the call. When the connection is completed, the communication session begins.

During a conversation, the mobile station processes transmitted and received speech signals, as well as control signals transmitted simultaneously with speech. At the end of the conversation, service messages are exchanged between the mobile and base stations, after which the mobile station's transmitter is turned off and the station goes into standby mode.

If the call is initiated from the mobile station, i.e. The subscriber dials the number of the called subscriber and presses the “call” button on the control panel, then the mobile station transmits a message through its base station indicating the called number and data for authenticating the mobile subscriber. After authentication, the base station assigns a traffic channel, and the subsequent steps to prepare a communication session are the same as when a call arrives from the system.

The base station then notifies the switching center that the mobile station is ready, the switching center transmits the call to the network, and the mobile station subscriber is able to hear “calling” or “busy” signals. The connection is terminated on the network side.

Each time a connection is established, authentication and identification procedures are performed.

Authentication is a procedure for confirming the authenticity (validity, legality, availability of rights to use cellular communication services) of a subscriber of a mobile communication system. The need to introduce this procedure is caused by the inevitable temptation to gain unauthorized access to cellular communication services.

Identification is a procedure for establishing that a mobile station belongs to one of the groups that have certain properties or characteristics. This procedure is used to identify lost, stolen or faulty devices.

The idea of ​​the authentication procedure in a digital cellular communication system is to encrypt some identifier passwords using quasi-random numbers periodically transmitted to the mobile station from the switching center, and an encryption algorithm individual for each mobile station. This encryption, using the same input data and algorithms, is performed at both the mobile station and the switching center, and authentication is considered successful if both results match.

The identification procedure consists of comparing the subscriber device identifier with the numbers contained in the corresponding “black lists” of the equipment register in order to remove stolen and technically faulty devices from circulation. The device identifier is made in such a way that it is difficult and economically unprofitable to change or fake it.

When a mobile station moves from one cell to another, its service is transferred from the base station of the first cell to the base station of the second (Fig. 1.9.). This process is called handover. It occurs only when the mobile station crosses the cell boundary during a communication session and the communication is not interrupted. If the mobile station is in the standby mode, it simply tracks these movements using system information transmitted over the control channel, and at the right time it switches to a stronger signal from another base station.

Rice. 1.9. Handover from cell A to cell B when a mobile station crosses a cell boundary.

The need for handover occurs when the quality of the communication channel, as measured by signal strength and/or bit error rate, falls below an acceptable limit. In the D-AMPS standard, the mobile station measures these characteristics only for the working cell, but if the communication quality deteriorates, it reports this through the base station to the switching center, and at the command of the latter, similar measurements are performed by mobile stations in neighboring cells. Based on the results of these measurements, the switching center selects the cell to which the service should be transferred.

Service is transferred from a cell with the worst quality of the communication channel to a cell with best quality, and the specified difference must be no less than a certain specified value. If this condition is not required, then, for example, when a mobile station moves approximately along the cell boundary, multiple handovers from the first cell to the second and back are possible, leading to the system being loaded with meaningless work and to a decrease in communication quality.

Having made a decision to hand over and select a new cell, the switching center informs the base station of the new cell about this, and the mobile station, through the base station of the old cell, issues the necessary commands indicating the new frequency channel, working slot number, etc. The mobile station is being rebuilt to new channel and is configured to work together with the new base station, performing approximately the same steps as when preparing a communication session, after which communication continues through the base station of the new cell. In this case, a break in a telephone conversation does not exceed a fraction of a second and remains invisible to the subscriber.

A cellular communication system can provide a roaming function - this is the procedure for providing cellular communication services to a subscriber of one operator in the system of another operator.

An idealized and simplified scheme for organizing roaming is as follows: a cellular subscriber who finds himself in the territory of a “foreign” system that allows roaming initiates a call as if he were in the territory of “his” system. The switching center, having made sure that this subscriber is not listed in its home register, perceives it as a roamer and enters it in the guest register. At the same time, it requests in the home register of the roamer’s “native” system information related to it, necessary for organizing service, and reports in which system the roamer is currently located; the latest information is recorded in the home register of the roamer’s “native” system. After this, the roamer uses cellular communications as if at home.

1.3 Methods for separating subscribers in cellular communications

Communication resource represents the time and bandwidth available for signal transmission in a particular system. To create an efficient communication system, it is necessary to plan resource allocation among system users so that time/frequency is used as efficiently as possible. The result of such planning should be equal access of users to the resource. There are three main methods of separating subscribers in a communication system.

1. Frequency division. Specific sub-bands of the frequency band used are allocated.

2. Temporal separation. Subscribers are allocated periodic time slots. Some systems provide users with a limited time to communicate. In other cases, the time users access a resource is determined dynamically.

3. Code separation. Stand out certain elements a set of orthogonally (or almost orthogonally) distributed spectral codes, each of which uses the entire frequency range.

With frequency division MA (FDMA), the communication resource is distributed according to Fig. 1.10. Here, the distribution of signals or users over a frequency range is long-term or permanent. A communication resource can simultaneously contain several signals spaced apart in the spectrum.

The primary frequency range contains signals that use the frequency interval between f 0 and f 1, the second - between f 2 and f 3, etc. The regions of the spectrum that lie between the bands used are called guard bands. Guard bands act as a buffer, which reduces interference between adjacent (by frequency) channels.

Rice. 1.10. Frequency division multiplexing.

To make the unmodulated signal use a higher frequency range, it is converted by superposing or mixing (modulating) this signal and a sine wave signal of a fixed frequency.

In time divisional MA (TDMA), the communication resource is distributed by providing each of the M signals (users) with the entire spectrum for a short period of time called a time slot (Fig. 1.11.). The time periods separating the used intervals are called guard intervals.

The guard interval creates some temporal uncertainty between adjacent signals and acts as a buffer, thereby reducing interference. Typically, time is divided into intervals called frames. Each frame is divided into time slots that can be distributed among users. The overall structure of frames is periodic, so that TDMA data transmission is one or more time slots that repeat periodically throughout each frame.

Rice. 1.11. Seal with temporary separation.

Code division multiple access (CDMA) is a practical application of spread spectrum techniques that can be divided into two main categories: direct sequence spread spectrum and frequency hopping spread spectrum.

Let's consider expanding the spectrum using the direct sequence method. Spread spectrum gets its name from the fact that the bandwidth used for signal transmission is much wider than the minimum required for data transmission. So, N users receive an individual code g i (t), where i = 1,2,…,N. The codes are approximately orthogonal.

Block diagram standard system CDMA is shown in Fig. 1.12.

Rice. 1.12. Code Division Multiple Access.

The first block of the circuit corresponds to data modulation of the carrier wave Acosω 0 t. The output of the modulator belonging to the user from group 1 can be written in the following form: s 1 (t)=A 1 (t)cos(ω 0 t+φ 1 (t)).

The type of signal received can be arbitrary. The modulated signal is multiplied by the spreading signal g 1 (t), assigned to group 1; the result g 1 (t)s 1 (t) is transmitted over the channel. Similarly, for users of groups from 2 to N, the product of the code function and the signal is taken. Quite often, access to code is limited to a clearly defined group of users. The resulting signal in a channel is a linear combination of all transmitted signals. Neglecting delays in signal transmission, the specified linear combination can be written as follows: g 1 (t)s 1 (t)+ g 2 (t)s 2 (t)+…+ g N (t)s N (t).

Multiplying s 1 (t) and g 1 (t) results in a function whose spectrum is the convolution of the spectra s 1 (t) and g 1 (t). Since the signal s 1 (t) can be considered narrowband (compared to g 1 (t)), the bands g 1 (t) s 1 (t) and g 1 (t) can be considered approximately equal. Consider a receiver configured to receive messages from user group 1. Assume that the received signal and the code g 1 (t) generated by the receiver are completely synchronized with each other. The first step of the receiver will be to multiply the received signal by g 1 (t). As a result, the function g 1 2 (t)s 1 (t) and a set of side signals g 1 (t)g 2 (t)s 2 (t)+ g 1 (t)g 3 (t)s 3 (t) will be obtained )+…+ g 1 (t)g N (t)s N (t). If the code functions g i (t) are mutually orthogonal, the resulting signal can be ideally extracted in the absence of noise, because

.

Side signals are easily filtered out by the system, since

.

The main advantages of CDMA are privacy and noise immunity.

1. Confidentiality. If a user group code is known only to authorized members of that group, CDMA ensures communication confidentiality because unauthorized persons who do not have the code cannot access the information being transmitted.

2. Noise immunity. Modulating a signal with a sequence when transmitting requires it to be re-modulated with the same sequence when receiving (which is equivalent to demodulating the signal), resulting in the restoration of the original narrowband signal. If the interference is narrowband, then the demodulating direct sequence upon reception acts on it as a modulating one, i.e. “spreads” its spectrum over a wide band W ss, as a result of which only 1/G part of the interference power falls into the narrow band of the signal W s, so that the narrow-band interference will be weakened by G times, where G=W ss /W s (W ss – extended spectrum band, W s – original spectrum). If the interference is broadband - with a band of the order of W ss or wider, then demodulation will not change the width of its spectrum, and the interference will enter the signal band weakened as many times as its band is wider than the W s band of the original signal.

1.4 Standard DECT for communication

DECT systems and devices are distributed in more than 30 countries on all continents of the planet. In fact, DECT is a set of specifications that define radio interfaces for various types of communication networks and equipment. DECT combines the requirements, protocols and messages that enable communication networks and terminal equipment to interact. The organization of the networks themselves and the design of the equipment are not included in the standard. The most important task of DECT is to ensure compatibility between equipment from different manufacturers.

Initially, DECT was focused on telephony - radio extenders, wireless private branch exchanges, providing radio access to public telephone networks. But the standard turned out to be so successful that it began to be used in data transmission systems and wireless subscriber access to public communication networks. DECT has found use in multimedia applications and home radio networks, Internet access and fax communications.

What is a DECT radio interface? In the 20 MHz wide range (1880 – 1900 MHz), 10 carrier frequencies are allocated with an interval of 1.728 MHz. DECT uses time division access technology – TDMA. The time spectrum is divided into separate frames of 10ms (Fig. 1.13.). Each frame is divided into 24 time slots: 12 slots for reception (from the point of view of the wearable terminal) and 12 for transmission. Thus, on each of the 10 carrier frequencies, 12 duplex channels are formed - 120 in total. Duplex is ensured by time division (with an interval of 5 ms) of reception/transmission. For synchronization, the 32-bit sequence “101010...” is used. DECT provides speech compression in accordance with adaptive differential pulse code modulation technology at a speed of 32 Kbit/s. Therefore, the information part of each slot is 320 bits. When transmitting data, it is possible to combine time slots. The radio path uses Gaussian frequency modulation.

Base stations (BS) and subscriber terminals (AT) DECT constantly scan everything available channels(up to 120). In this case, the signal strength on each channel is measured and entered into the RSSI list. If a channel is busy or very noisy, its RSSI is high. The BS selects the channel with the lowest RSSI value for constant transmission of service information about subscriber calls, station identifier, system capabilities, etc. This information plays the role of reference signals for the AT - using them, subscriber devices determine whether they have the right to access a particular BS, whether it provides the services required by the subscriber, whether there is free capacity in the system, and select the BS with the highest quality signal.

In DECT, the communication channel always determines the AT. When a connection is requested from the BS (incoming connection), the AT receives a notification and selects a radio channel. Service information is transmitted by the base station and analyzed by the subscriber terminal constantly, therefore, the AT is always synchronized with the closest available BS. When establishing a new connection, the AT selects the channel with the lowest RSSI value - this ensures that the new connection occurs on the “cleanest” channel available. This dynamic channel allocation procedure allows you to get rid of frequency planning - the most important property of DECT.

Rice. 1.13. DECT spectrum.

Since the AT constantly, even when a connection is established, analyzes the available channels, they can be dynamically switched during a communication session. Such switching is possible both to another channel of the same BS, and to another BS. This procedure is called "handover". During handover, the AT establishes a new connection, and communication is maintained for some time on both channels. Then the best one is selected. Automatic switching between channels of different BS occurs almost unnoticed by the user and is completely initiated by the AT.

It is important that in the radio path of DECT equipment the signal power is very low - from 10 to 250 mW. Moreover, 10 mW is practically the nominal power for microcellular systems with a cell radius of 30–50 m inside a building and up to 300–400 m in open space. Transmitters with a power of up to 250 mW are used for radio coverage of large areas (up to 5 km).

With a power of 10 mW, it is possible to locate base stations at a distance of 25 m. As a result, a record density of simultaneous connections is achieved (about 100 thousand subscribers), provided that the BS is located according to a hexagonal pattern in the same plane (on the same floor).

To protect against unauthorized access, DECT systems use the BS and AT authentication procedure. The AT is registered in the system or at individual base stations to which it has access. With each connection, authentication occurs: the BS sends a “request” to the AT - a random number (64 bits). Based on this number and the authentication key, the AT and BS, using a given algorithm, calculate an authentication response (32 bits), which the AT transmits to the BS. The BS compares the calculated response with the received one and, if they match, allows the AT to connect. DECT has a standard authentication algorithm, DSAA.

As a rule, the authentication key is calculated based on the subscriber authentication key UAK with a length of 128 bits or the AC authentication code (16 - 32 bits). UAK is stored in the AT ROM or in a DAM card - an analogue of a SIM card. AC can be manually written to the AT ROM or entered during authentication. Together with UAK, a personal UPI user identifier of 16-32 bits in length is used, entered only manually. In addition, unauthorized retrieval of information in systems with TDMA is extremely difficult and is only accessible to specialists.

1.5 Standards Bluetooth , Wi - Fi (802.11, 802.16)

The Bluetooth specification describes a packet method for transmitting information with time multiplexing. Radio exchange occurs in the frequency band 2400-2483.5 MHz. The radio path uses the method of spectrum expansion through frequency jumps and two-level Gaussian frequency modulation.

The frequency hopping method implies that the entire frequency band allocated for transmission is divided into a certain number of subchannels, each 1 MHz wide. The channel is a pseudo-random sequence of hops across 79 or 23 RF subchannels. Each channel is divided into time segments of 625 μs duration, with each segment corresponding to a specific subchannel. The transmitter uses only one subchannel at a time. Jumps occur synchronously in the transmitter and receiver in a pre-fixed pseudo-random sequence. Up to 1600 frequency jumps can occur per second. This method ensures confidentiality and some noise immunity of transmissions. Noise immunity is ensured by the fact that if a transmitted packet could not be received on any subchannel, the receiver reports this and the transmission of the packet is repeated on one of the following subchannels, at a different frequency.

The Bluetooth protocol supports both point-to-point and point-to-multipoint connections. Two or more devices using the same channel form a piconet. One of the devices operates as the main one, and the rest - as slaves. A single piconet can have up to seven active slaves, with the remaining slaves in a "parked" state, remaining synchronized with the master. Interconnected piconets form a "distributed network".

There is only one master device in each piconet, but slave devices can be part of different piconets. In addition, the main device of one piconet can be a slave device in another (Fig. 1.14.). Piconets are not synchronized with each other in time and frequency - each of them uses its own sequence of frequency hops. In one piconet, all devices are synchronized in time and frequencies. The hop sequence is unique for each piconet and is determined by the address of its host device. The cycle length of the pseudo-random sequence is 2 27 elements.

Rice. 1. 14. A piconet with one slave device a), several b) and a distributed network c).

The Bluetooth standard provides duplex transmission based on time sharing. The main device transmits packets in odd time segments, and the slave device - in even ones (Fig. 1.15.). Packets, depending on their length, can occupy up to five time segments. In this case, the channel frequency does not change until the end of the packet transmission (Fig. 1.16.).

Rice. 1. 15. Time diagram of channel operation.

The Bluetooth protocol can support an asynchronous data channel, up to three synchronous (constant-rate) voice channels, or a channel with simultaneous asynchronous data and synchronous voice.

In a synchronous connection, the host device reserves time segments that follow at so-called synchronous intervals. Even if a packet is received with an error, it is not retransmitted during a synchronous connection. Asynchronous communication uses time segments that are not reserved for synchronous communication. If no address is specified in the address field of an asynchronous packet, the packet is considered “broadcast” - it can be read by all devices. An asynchronous connection allows packets received with errors to be retransmitted.

Rice. 1. 16. Transmission of packets of various lengths.

A standard Bluetooth packet contains a 72-bit passcode, a 54-bit header, and an information field of no more than 2745 bits. The access code identifies packets belonging to the same piconet and is also used for synchronization and query procedures. It includes a preamble (4 bits), a sync word (64 bits) and a trailer - 4 bits checksum.

The header contains information for communication control and consists of six fields: AM_ADDR – 3-bit address of the active element; TYPE – 4-bit data type code; FLOW – 1 bit of data flow control, indicating the device’s readiness to receive; ARQN – 1 bit confirming correct reception; SEQN – 1 bit used to determine the sequence of packets; HEC – 8-bit checksum.

The information field, depending on the type of packets, may contain either voice fields, or data fields, or both types of fields simultaneously.

Let's consider the IEEE 802.11 standard used in local data networks - i.e. in Ethernet-like wireless networks, which are fundamentally asynchronous in nature.

IEEE 802.11 considers the lower two layers of the interworking model open systems– physical (the method of working with the transmission medium, speed and modulation methods are determined) and the data link level, and at the last level the lower sublayer is considered - MAC, i.e. control access to the channel (transmission medium). IEEE 802.11 uses the range 2.400 - 2.4835 GHz with a bandwidth of 83.5 MHz and provides packet transmission with 48-bit address packets.

The standard provides two main ways to organize a local network - according to the principle of “everyone with everyone” (communication is established directly between two stations, all devices must be within radio visibility, no administration occurs) and in the form of a structured network (an additional device appears - an access point, as a rule, stationary and operating on a fixed channel; communication between devices occurs only through access points, through which access to external wired networks is possible).

As a rule, control functions are distributed among all devices of the IEEE 802.11 network - DCF mode. However, for structured networks, PCF mode is possible, when control is transferred to one specific access point. The need for PCF mode arises when transmitting delay-sensitive information. After all, IEEE 802.11 networks operate on the principle of competitive access to the channel - there are no priorities. To set them if necessary, the PCF mode was introduced. However, operation in this mode can only occur at certain periodically repeated intervals.

To ensure the security of data transmission, station authentication and encryption of transmitted data are provided at the MAC level.

IEEE 802.11 provides multiple channel access with carrier sense and collision detection. The station can start transmitting only if the channel is free. If stations detect that multiple stations are trying to operate on the same channel, they all stop transmitting and try to resume it after a random period of time. Thus, even when transmitting, the device must monitor the channel, i.e. work at reception.

Before the first attempt to access a channel, the device loads the duration of a random wait interval into a special counter. Its value is decremented at a given frequency while the channel is idle. Once the counter is reset to zero, the device can occupy the channel. If another device occupies the channel before the counter is reset, the counting stops, maintaining the achieved value. On the next attempt, the countdown starts from the saved value. As a result, those who didn’t make it last time get a better chance to occupy the channel next time. This is not the case with wired Ethernet networks.

The packets through which transmission occurs are actually formed at the MAC layer; at the physical layer, a physical layer header (PLCP) is added to them, consisting of a preamble and the PLCP header itself. MAC layer packets can be of three types - data packets, control packets and control packets. Their structure is the same. Each packet includes a MAC header, an information field, and a checksum.

Fixed-access broadband metropolitan wireless data networks use the IEEE 802.16 standard.

The IEEE 802.16 standard describes operation in the 10 - 66 GHz range of systems with a point-to-multipoint architecture (from the center to many). This is a bidirectional system, i.e. downstream (from the base station to subscribers) and upstream (to the base station) flows are provided. In this case, the channels are broadband (about 25 MHz), and the transmission speeds are high (for example, 120 Mbit/s).

The IEEE 802.16 standard provides a single-carrier modulation scheme (per frequency channel) and allows three types of quadrature amplitude modulation: four-position QPSK and 16-position 16-QAM (required for all devices), as well as 64-QAM (optional).

Data at the physical layer is transmitted as a continuous sequence of frames. Each frame has a fixed duration – 0.5; 1 and 2 ms. The frame consists of a preamble (sync sequence 32 QPSK symbols long), a control section, and a sequence of data packets. Since the system defined by the IEEE 802.16 standard is bidirectional, a duplex mechanism is required. It provides both frequency and time separation of upstream and downstream channels. With time channel duplexing, the frame is divided into downlink and uplink subframes, separated by a special interval. With frequency duplexing, the uplink and downlink channels are each broadcast on their own carrier.

The IEEE 802.16 MAC layer is divided into three sublayers - the service transformation sublayer (services are different applications), the main sublayer and the security sublayer. At the security sublevel, authentication mechanisms and data encryption are implemented. At the service transformation sublayer, data streams of upper-level protocols are transformed to transmit data through IEEE 802.16 networks. For each type of upper-level application, the standard provides its own conversion mechanism. At the main MAC sublayer, data packets are generated, which are then transmitted to the physical layer and broadcast through the communication channel. The MAC packet includes a header and a data field, which may be followed by a checksum.

A key point in the IEEE 802.16 standard is the concept of service flow and the associated concepts of “connection” and “connection identifier” (CID). A service stream in the IEEE 802.16 standard is a data stream associated with a specific application. In this context, a connection is the establishment of a logical connection at the MAC layers on the transmitting and receiving sides for the transmission of a service stream. Each connection is assigned a 16-bit CID, which is uniquely associated with the type and characteristics of the connection. The service flow is characterized by a set of requirements for the information transmission channel (symbol delay time, level of delay fluctuations and guaranteed throughput). Each service flow is assigned an SFID, based on which the BS determines the necessary parameters of the specific connection associated with this service flow.

The basic principle of providing channel access in the IEEE 802.16 standard is access on demand. No AS (subscriber station) can transmit anything except requests for registration and provision of a channel until the BS allows it to do so, i.e. will allocate a time slot in the uplink and indicate its location. The speaker can either request a certain size of bandwidth in the channel, or ask to change the channel resource already provided to it. The IEEE 802.16 standard provides two modes of granting access - for each individual connection and for all connections of a specific AS. Obviously, the first mechanism provides greater flexibility, but the second significantly reduces the volume of overhead messages and requires less performance from the hardware.

2. Complex signal systems for telecommunication systems

2.1 Signal spectra

The signal spectrum s(t) is determined by the Fourier transform

In general, the spectrum is a complex function of frequency ω. The spectrum can be represented in the form

,

where |S(ω)| is the amplitude, and φ(ω) is the phase spectrum of the signal s(t).

The signal spectrum has the following properties:

1. Linearity: if there is a set of signals s 1 (t), s 2 (t), ..., and s 1 (t)S 1 (ω), s 2 (t)S 2 (ω), ..., then the sum of the signals is Fourier transformed as follows:

where a i are arbitrary numerical coefficients.

2. If the signal s(t) corresponds to the spectrum S(ω), then the same signal, shifted by t 0, corresponds to the spectrum S(ω) multiplied by e - jωt 0 s(t-t 0)S(ω)e - jωt 0 .

3. If s(t)S(ω), then

4. If s(t)S(ω) and f(t)=ds/dt, then f(t)F(ω)=jωS(ω).

5. If s(t)S(ω) and g(t)=∫s(t)dt, then g(t)G(ω)=S(ω)/jω.

6. If u(t)U(ω), v(t)V(ω) and s(t)=u(t)v(t), then

.

The signal is located across the spectrum using the inverse Fourier transform

.

Let's consider the spectra of some signals.

1. Rectangular pulse.

Fig.2.1. Spectrum of a rectangular pulse.

2. Gaussian impulse.

s(t)=Uexp(-βt 2)

Fig.2.2. Spectrum of a Gaussian pulse.

3. Smoothed impulse

Using numerical integration we find the spectrum S(ω).

S(0)=2.052 S(6)=-0.056

S(1)=1.66 S(7)=0.057

S(2)=0.803 S(8)=0.072

S(3)= 0.06 S(9)=0.033

S(4)=-0.259 S(10)=-0.0072

S(5)=-0.221 S(ω)=S(-ω)

Rice. 2.3. Spectrum of a smoothed pulse.

2.2 Correlation properties of signals

To compare time-shifted signals, an autocorrelation function (ACF) of the signal is introduced. It quantitatively determines the degree of difference between the signal u(t) and its time-shifted copy u(t - τ) and is equal to the scalar product of the signal and the copy:

It is immediately clear that at τ=0 the autocorrelation function becomes equal to the signal energy: B u (0)=E u .

The autocorrelation function is even: B u (τ)=B u (-τ).

For any value of the time shift τ, the ACF modulus does not exceed the signal energy |B u (τ)|≤B u (0)=E u .

The ACF is related to the signal spectrum as follows:

.

The opposite is also true:

.

For a discrete signal, the ACF is determined as follows:

and has the following properties.

The discrete ACF is even: B u (n)=B u (-n).

At zero shift, the ACF determines the energy of the discrete signal:

.

Sometimes a cross-correlation function (MCF) of signals is introduced, which describes not only the time shift of the signals relative to each other, but also the difference in the shape of the signals.

VKF is determined as follows

for continuous signals and

for discrete signals.

Let's consider the ACF of some signals.

1. Sequence of rectangular pulses

Rice. 2.4. ACF of a sequence of rectangular pulses.

2. 7-position Barker signal

B u (0)=7, B u (1)= B u (-1)=0, B u (2)= B u (-2)=-1, B u (3)= B u (-3 )=0, B u (4)= B u (-4)=-1, B u (5)= B u (-5)=0, B u (6)= B u (-6)=-1 , B u (7)= B u (-7)=0.

Rice. 2.5. ACF of 7-position Barker signal.

3. 8-position Walsh functions

Walsh function of 2nd order

B u (0)=8, B u (1)= B u (-1)=3, B u (2)= B u (-2)=-2, B u (3)= B u (-3 )=-3, B u (4)= B u (-4)=-4, B u (5)= B u (-5)=-1, B u (6)= B u (-6)= 2, B u (7)= B u (-7)=1, B u (8)= B u (-8)=0.

Rice. 2.6. ACF of the 2nd order Walsh function.

7th order Walsh function

B u (0)=8, B u (1)= B u (-1)=-7, B u (2)= B u (-2)=6, B u (3)= B u (-3 )=-5, B u (4)= B u (-4)=4, B u (5)= B u (-5)=-3, B u (6)= B u (-6)=2 , B u (7)= B u (-7)=-1, B u (8)= B u (-8)=0.

Rice. 2.7. ACF of the 7th order Walsh function.

2.3 Types of complex signals

A signal is a physical process that can carry useful information and propagate along a communication line. By signal s(t) we mean a function of time that reflects a physical process with a finite duration T.

Signals for which the base B, equal to the product of the signal duration T and the width of its spectrum, is close to unity, are called “simple” or “ordinary”. Such signals can be distinguished by frequency, time (delay) and phase.

Complex, multidimensional, noise-like signals are formed according to a complex law. During the duration of the signal T, it undergoes additional manipulation (or modulation) in frequency or phase. Additional amplitude modulation is rarely used. Due to additional modulation, the spectrum of the signal Δf (while maintaining its duration T) expands. Therefore, for such a signal B=T Δf>>1.

Under certain laws of formation of a complex signal, its spectrum turns out to be continuous and almost uniform, i.e. close to the noise spectrum with limited bandwidth. In this case, the signal autocorrelation function has one main spike, the width of which is determined not by the duration of the signal, but by the width of its spectrum, i.e. has a form similar to the band-limited noise autocorrelation function. In this regard, such complex signals are called noise-like.

Noise-like signals are used in broadband communication systems because: they provide high noise immunity of communication systems; allow you to organize the simultaneous operation of many subscribers in a common frequency band; allow you to successfully combat multipath propagation of radio waves by splitting the beams; provide better use of the frequency spectrum in a limited area compared to narrowband communication systems.

A large number of different noise-like signals (NLS) are known. However, the following main NPS are distinguished: frequency-modulated signals; multi-frequency signals; phase-shift keyed signals; discrete frequency signals; discrete composite frequency signals.

Frequency-modulated signals (FM) are continuous signals, the frequency of which varies according to a given law (Fig. 2.8.).

Rice. 2.8. FM signal.

In communication systems, it is necessary to have many signals. At the same time, the need to quickly change signals and switch generation and processing equipment leads to the fact that the law of frequency change becomes discrete. In this case, they move from FM signals to DF signals.

Multi-frequency (MF) signals are the sum of N harmonics u 1 (t)…u N (t), the amplitudes and phases of which are determined in accordance with the laws of signal formation (Fig. 2.9.).

Rice. 2.9. MF signal.

MF signals are continuous and it is difficult to adapt digital techniques for their formation and processing.

Phase-manipulated (PM) signals represent a sequence of radio pulses, the phases of which change according to a given law (Fig. 2.10., a). Typically, the phase takes two values ​​(0 or π). In this case, the radio frequency FM signal corresponds to the video FM signal (Fig. 2.10., b).

Rice. 2.10. FM signal.

FM signals are very common because... they allow extensive use of digital methods in generation and processing, and such signals can be realized with relatively large bases.

Discrete frequency (DF) signals represent a sequence of radio pulses (Fig. 2.11.), the carrier frequencies of which vary according to a given law.

Rice. 2.11. HF signal.

Discrete composite frequency (DCF) signals are CD signals in which each pulse is replaced by a noise-like signal.

In Fig. 2.12. depicts an FM video frequency signal, individual parts of which are transmitted at different carrier frequencies.

Rice. 2.12. DHF signal.

2.4 Derived signal systems

A derivative signal is a signal that is obtained by multiplying two signals. In the case of PM signals, multiplication must be carried out element-by-element or, as is more often called, character-by-character. A system made up of derivative signals is called a derivative. Among derivative systems, systems constructed as follows are of particular importance. As a basis, a certain system of signals is used, the correlation properties of which do not fully satisfy the requirements for CF, but which has certain advantages in terms of ease of generation and processing. Such a system is called the original one. Then a signal is selected that has certain properties. Such a signal is called producing. By multiplying the producing signal by each signal of the original system, we obtain a derivative system. The producing signal should be chosen so that the derived system is truly better than the original one, i.e. so that it has good correlation properties. The complex envelope of the derivative signal S μ m (t) is equal to the product of the complex envelopes of the original signals U m (t) and the producing signal V μ (t), i.e. S μ m (t)= U m (t)V μ (t). If the indices vary within m=1..M, μ=1..H, then the volume of the derivative signal system is L=MH.

The choice of generating signals is determined by a number of factors, including the source system. If the signals of the original system are broadband, then the producing signal can be broadband and have small levels of side peaks of the uncertainty function, close to the root mean square value. If the signals of the original system are narrow-band, then it is sufficient to fulfill the inequality F V >>F U (F V is the width of the spectrum of the producing signals, F U is the width of the spectrum of the original signals) and the requirement that the side peaks of the ACF are small.

Let's take the Walsh system as a starting point. In this case, the producing signals must be broadband and have good ACF. In addition, the producing signal must have the same number of elements as the original signals, i.e. N=2 k elements, where k is an integer. These conditions are generally satisfied by nonlinear sequences. Since the main requirement is the smallness of the ACF side peaks, the best signals with the number of elements N = 16, 32, 64 were selected in the class of nonlinear sequences. These signals are shown in Fig. 2.13. In Fig. 2.13. The number of blocks μ for each generating signal is also indicated. They are close to the optimal value μ 0 =(N+1)/2. This is a necessary condition for obtaining a good ACF with small side peaks.

Rice. 2.13. Producing FM signals.

The volume of the derivative system is equal to the volume of the Walsh system N. Derivative systems have better correlation properties than Walsh systems.

3. Modulation of complex signals

3.1 Geometric representation of signals

Let's consider a geometric or vector representation of signals. Let us define an N-dimensional orthogonal space as the space defined by a set of N linearly independent functions (ψ j (t)), called basis ones. Any function of this space can be expressed through a linear combination of basis functions, which must satisfy the condition

,

where the operator is called the Kronecker symbol. For non-zero constants K j the space is called orthogonal. If the basis functions are normalized so that all K j =1, the space is called orthonormal. The basic orthogonality condition can be formulated as follows: each function ψ j (t) of a set of basis functions must be independent of the other functions of the set. Each function ψ j (t) should not interfere with other functions during the detection process. From a geometric point of view, all functions ψ j (t) are mutually perpendicular.

In orthogonal signal space, the Euclidean distance measure used in the detection process is most easily defined. If the waves carrying the signals do not form such a space, they can be converted into a linear combination of orthogonal signals. It can be shown that an arbitrary finite set of signals (s i (t)) (i=1...M), where each element of the set is physically realizable and has a duration T, can be expressed as a linear combination of N orthogonal signals ψ 1 (t), ψ 2 ( t), …, ψ N (t), where NM, so

Where

The type of basis (ψ j (t)) is not specified; these signals are selected for convenience and depend on the waveform of the signals being transmitted. A set of such waves (s i (t)) can be considered as a set of vectors (s i )=(a i 1, a i 2, …,a iN). The relative orientation of the signal vectors describes the relationship between the signals (with respect to their phases or frequencies), and the amplitude of each set vector (s i ) is a measure of the signal energy carried over the symbol transmission time. In general, after selecting a set of N orthogonal functions, each of the transmitted signals s i (t) is completely determined by the vector of its coefficients s i =(a i 1, a i 2, …,a iN) i=1…M.

3.2 Methods of phase shift keying of signals (PM2, PM4, OFM)

Phase shift keying (PSK) was developed early in the development of the deep space exploration program; PSK is now widely used in commercial and military communications systems. The signal in PSK modulation has the following form:

Here the phase φ i (t) can take M discrete values, usually defined as follows:

The simplest example of phase shift keying is binary phase shift keying (PSK). Parameter E is the symbol energy, T is the symbol transmission time. The operation of the modulation circuit is to shift the phase of the modulated signal s i (t) to one of two values, zero or π (180 0). A typical form of the PM2 signal is shown in Fig. 3.1.a), where characteristic sharp phase changes during the transition between symbols are clearly visible; if the modulated data stream consists of alternating zeros and ones, such abrupt changes will occur at each transition. The modulated signal can be represented as a vector on a graph in a polar coordinate system; the length of the vector corresponds to the amplitude of the signal, and its orientation in the general M-ary case corresponds to the phase of the signal relative to other M – 1 signals in the set. When modulating PM2 (Fig. 3.1.b)), the vector representation gives two antiphase (180 0) vectors. Sets of signals that can be represented by similar antiphase vectors are called antipodal.

Rice. 3.1. Binary phase shift keying.

Another example of phase shift keying is PM4 modulation (M=4). When modulating PM4, parameter E is the energy of two symbols, time T is the transmission time of two symbols. The phase of the modulated signal takes one of four possible values: 0, π/2, π, 3π/2. In vector representation, the PM4 signal has the form shown in Fig. 3.2.

Rice. 3.2. PM4 signal in vector representation.

Let's consider another type of phase shift keying - relative phase shift keying (RPK) or differential phase shift keying (DPSK). The name differential phase shift keying requires some explanation, since the word "differential" refers to two different aspects of the modulation/demodulation process: the encoding procedure and the detection procedure. The term "differential encoding" is used when the encoding of binary characters is determined not by their value (i.e., zero or one), but by whether the character is the same or different from the previous one. The term "differential coherent detection" of signals in differential PSK modulation (this is the meaning in which the name DPSK is usually used) is associated with a detection circuit that is often classified as a non-coherent circuit because it does not require phase matching with the received carrier.

In non-coherent systems, no attempt is made to determine the actual phase value of the incoming signal. Therefore, if the transmitted signal has the form

then the received signal can be described as follows.

Here α is an arbitrary constant, usually assumed to be a random variable uniformly distributed between zero and 2π, and n(t) is noise.

For coherent detection, matched filters are used; For incoherent detection, this is not possible, since in this case the output of the matched filter will depend on the unknown angle α. But if we assume that α changes slowly relative to an interval of two periods (2T), then the phase difference between two successive signals will not depend on α.

The basis of differential coherent signal detection in DPSK modulation is as follows. During the demodulation process, the carrier phase of the previous symbol transmission interval may be used as a reference phase. Its use requires differential encoding of the message sequence in the transmitter, since the information is encoded by the phase difference between two successive pulses. To transmit the i-th message (i=1,2,…,M), the phase of the current signal must be shifted by φ i =2πi/M radians relative to the phase of the previous signal. In general, the detector calculates the coordinates of the incoming signal by determining its correlation with the locally generated signals cosω 0 t and sinω 0 t. Then, as shown in Fig. 3.3., the detector measures the angle between the vector of the current received signal and the vector of the previous signal.

Rice. 3.3. Signal space for DPSK scheme.

DPSK is less efficient than PSK because in the first case, due to correlation between signals, errors tend to spread (to adjacent symbol times). It is worth remembering that the PSK and DPSK schemes differ in that in the first case the received signal is compared with an ideal reference signal, and in the second case two noisy signals are compared. Note that DPSK modulation produces twice as much noise as PSK modulation. Therefore, you should expect twice the error rate with DPSK than with PSK. The advantage of the DPSK scheme is the reduced complexity of the system.

3.3 Modulation with minimum frequency shift.

One modulation scheme without phase discontinuity is minimum frequency shift keying (MSK). MSK can be considered a special case of frequency shift keying without phase breaking. The MSK signal can be represented as follows.

Here f 0 is the carrier frequency, d k =±1 represents bipolar data, which is transmitted at a rate of R=1/T, and x k is the phase constant for the kth binary data transmission interval. Note that when d k =1 the transmitted frequency is f 0 +1/4T, and when d k =-1 it is f 0 -1/4T. During each T-second data transmission interval, the value of x k is constant, i.e. x k =0 or π, which is dictated by the requirement of signal phase continuity at moments t=kT. This requirement imposes a constraint on the phase, which can be represented by the following recursive relationship for x k.

The equation for s(t) can be rewritten in quadrature representation.

The common-mode component is denoted as a k cos(πt/2T)cos2πf 0 t, where cos2πf 0 t is the carrier, cos(πt/2T) is the sinusoidal symbol weighting, and k is the information-dependent term. Similarly, the quadrature component is b k sin(πt/2T)sin2πf 0 t, where sin2πf 0 t is the quadrature term of the carrier, sin(πt/2T) is the same sinusoidal symbol weighting, b k is the information-dependent term. It may seem that the quantities a k and b k can change their value every T seconds. However, due to the requirement of phase continuity, the value of a k can change only when the function cos(πt/2T) passes through zero, and b k - only when sin(πt/2T) passes through zero. Therefore, symbol weighting in an in-phase or quadrature channel is a sinusoidal pulse with a period of 2T and a variable sign. The in-phase and quadrature components are shifted relative to each other by T seconds.

The expression for s(t) can be rewritten in another form.

Here d I (t) and d Q (t) have the same meaning of in-phase and quadrature data streams. An MSK scheme written in this form is sometimes called pre-coded MSK. A graphical representation of s(t) is given in Fig. 3.4. In Fig. 3.4. a) and c) sinusoidal weighing of pulses of in-phase and quadrature channels is shown, here multiplication by a sinusoid gives more smooth transitions phases than in the original data representation. In Fig. 3.4. b) and d) the modulation of the orthogonal components cos2πf 0 t and sin2πf 0 t by sinusoidal data streams is shown. In Fig. 3.4. e) the summation of the orthogonal components shown in Fig. is presented. 3.4. b) and d). From the expression for s(t) and Fig. 3.4. we can conclude the following: 1) the signal s(t) has a constant envelope; 2) the phase of the RF carrier is continuous during bit transitions; 3) the signal s(t) can be considered as an FSK modulated signal with transmission frequencies f 0 +1/4T and f 0 -1/4T. Thus, the minimum tone spacing required in MSK modulation can be written as follows:

which is equal to half the bit rate. Note that the tone spacing required for MSK is half (1/T) of the spacing required for non-coherent detection of FSK modulated signals. This is because the carrier phase is known and continuous, allowing coherent demodulation of the signal.

Rice. 3.4. Minimal shift manipulation: a) modified in-phase bit stream; b) the product of an in-phase bit stream and a carrier; c) modified quadrature bit stream; d) the product of the quadrature bit stream and the carrier; e) MSK signal.

3.4 Quadrature modulation and its characteristics ( Q PSK , QAM )

Consider quadrature phase shift keying (QPSK). The original data stream d k (t)=d 0 , d 1 , d 2 ,… consists of bipolar pulses, i.e. d k take the values ​​+1 or -1 (Fig. 3.5.a)), representing a binary one and a binary zero. This pulse flow is divided into an in-phase flow d I (t) and a quadrature flow - d Q (t), as shown in Fig. 3.5.b).

d I (t)=d 0 , d 2 , d 4 ,… (even bits)

d Q (t)=d 1 , d 3 , d 5 ,… (odd bits)

A convenient orthogonal implementation of a QPSK signal can be obtained using amplitude modulation of in-phase and quadrature streams on sine and cosine functions of the carrier.

Using trigonometric identities, s(t) can be represented in the following form: s(t)=cos(2πf 0 t+θ(t)). The QPSK modulator shown in Fig. 3.5.c), uses the sum of the sine and cosine terms. The pulse stream d I (t) is used to amplitude modulate (with an amplitude of +1 or -1) a cosine wave. This is equivalent to shifting the phase of the cosine wave by 0 or π; therefore, the result is a BPSK signal. Similarly, a stream of pulses d Q (t) modulates a sinusoid, which gives a BPSK signal orthogonal to the previous one. By summing these two orthogonal carrier components, a QPSK signal is obtained. The value θ(t) will correspond to one of four possible combinations of d I (t) and d Q (t) in the expression for s(t): θ(t)=0 0, ±90 0 or 180 0; the resulting signal vectors are shown in the signal space in Fig. 3.6. Since cos(2πf 0 t) and sin(2πf 0 t) are orthogonal, the two BPSK signals can be detected separately. QPSK has a number of advantages over BPSK: because with QPSK modulation, one pulse transmits two bits, then the data transfer rate is doubled, or at the same data transfer rate as in the BPSK scheme, half the frequency band is used; and also increases noise immunity, because The pulses are twice as long and therefore more powerful than BPSK pulses.

Rice. 3.5. QPSK modulation.

Rice. 3.6. Signal space for QPSK scheme.

Quadrature amplitude modulation (KAM, QAM) can be considered a logical continuation of QPSK, since the QAM signal also consists of two independent amplitude-modulated carriers.

With quadrature amplitude modulation, both the phase and amplitude of the signal change, which allows you to increase the number of encoded bits and at the same time significantly improve noise immunity. The quadrature representation of signals is a convenient and fairly universal means of describing them. The quadrature representation is to express the oscillation as a linear combination of two orthogonal components - sine and cosine (in-phase and quadrature):

s(t)=A(t)cos(ωt + φ(t))=x(t)sinωt + y(t)cosωt, where

x(t)=A(t)(-sinφ(t)),y(t)=A(t)cosφ(t)

Such discrete modulation (manipulation) is carried out over two channels, on carriers shifted by 90 0 relative to each other, i.e. located in quadrature (hence the name).

Let us explain the operation of the quadrature circuit using the example of generating four-phase PM (PM-4) signals (Fig. 3.7).

Rice. 3.7. Quadrature modulator circuit.

Rice. 3.8. Hexadecimal signal space (QAM-16).

Using a shift register, the original sequence of binary symbols of duration T is divided into odd pulses y, which are supplied to the quadrature channel (cosωt), and even pulses – x, supplied to the in-phase channel (sinωt). Both sequences of pulses are supplied to the inputs of the corresponding manipulated pulse shapers, at the outputs of which sequences of bipolar pulses x(t) and y(t) with an amplitude ±U m and a duration of 2T are formed. Pulses x(t) and y(t) arrive at the inputs of channel multipliers, at the outputs of which two-phase (0, π) PM oscillations are formed. After summation, they form an FM-4 signal.

In Fig. 3.8. shows a two-dimensional signal space and a set of signal vectors modulated by hex QAM and represented by points that are arranged in a rectangular array.

From Fig. 3.8. it can be seen that the distance between signal vectors in the signal space with QAM is greater than with QPSK, therefore, QAM is more noise-resistant compared to QPSK,

3.5 Implementation of quadrature modems

The modem is designed to transmit/receive information over regular telephone wires. In this sense, the modem acts as an interface between the computer and the telephone network. Its main task is to convert the transmitted information into a form acceptable for transmission via telephone communication channels, and to convert the received information into a form acceptable for a computer. As you know, a computer is capable of processing and transmitting information in binary code, that is, in the form of a sequence of logical zeros and ones, called bits. A logical one can be associated with a high voltage level, and a logical zero with a low voltage level. When transmitting information over telephone wires, it is necessary that the characteristics of the transmitted electrical signals (power, spectral composition, etc.) meet the requirements of the receiving equipment of the telephone exchange. One of the main requirements is that the signal spectrum should lie in the range from 300 to 3400 Hz, that is, have a width of no more than 3100 Hz. In order to satisfy this and many other requirements, the data undergoes appropriate encoding, which, in fact, is done by the modem. There are several possible encoding methods in which data can be transmitted over subscriber switched channels. These methods differ from each other in both transmission speed and noise immunity. At the same time, regardless of the encoding method, data is transmitted over subscriber channels only in analog form. This means that a sinusoidal carrier signal is used to transmit information, which is subjected to analog modulation. The use of analog modulation results in a spectrum of much smaller width at a constant information transfer rate. Analog modulation is a physical encoding method in which information is encoded by changing the amplitude, frequency and phase of a sinusoidal carrier signal. There are several basic methods of analog modulation: amplitude, frequency and relative phase. Modems use listed methods modulation, but not separately, but all together. For example, amplitude modulation can be used in conjunction with phase modulation (amplitude-phase modulation). The main problem that arises when transmitting information over subscriber channels is increasing speed. The speed is limited by the spectral bandwidth of the communication channel. However, there is a method that can significantly increase the speed of information transfer without increasing the signal spectrum width. The main idea of ​​this method is to use multi-position coding. The sequence of data bits is divided into groups (symbols), each of which is associated with a discrete signal state. For example, using 16 different signal states (they can differ from each other in both amplitude and phase), it is possible to encode all possible combinations for sequences of 4 bits. Accordingly, 32 discrete states will allow a group of five bits to be encoded in one state. In practice, to increase the speed of information transmission, multi-position amplitude-phase modulation with several possible values ​​of amplitude levels and signal phase shift is used. This type of modulation is called quadrature amplitude modulation (QAM). In the case of QAM, it is convenient to depict signal states on the signal plane. Each point on the signal plane has two coordinates: the amplitude and phase of the signal and is an encoded combination of a sequence of bits. To increase the noise immunity of quadrature amplitude modulation, the so-called Trellis Code Modulation (TCM) or, in other words, trellis coding can be used. In Trellis modulation, one extra Trellis bit is added to each group of bits transmitted during one discrete signal state. If, for example, the information bits are divided into groups of 4 bits (a total of 16 different combinations are possible), then 16 signal points are placed in the signal plane. Adding a fifth Trellis bit will result in 32 possible combinations, doubling the number of signal points. However, not all bit combinations are legal, that is, meaningful. This is the idea behind trellis coding. The value of the added trellis bit is determined using a special algorithm. A special encoder is responsible for calculating the added trellis bit. On the receiving modem, a special decoder is designed to analyze incoming bit sequences - the so-called Viterbi decoder. If the received sequences are allowed, then the transmission is considered to occur without errors and the trellis bit is simply removed. If among the received sequences there are forbidden sequences, then using a special algorithm the Viterbi decoder finds the most suitable allowed sequence, thus correcting transmission errors. So, the meaning of trellis coding is to increase the noise immunity of transmission at the cost of relatively little redundancy. The use of Trellis coding makes it possible, mainly, to protect from confusion precisely the neighboring points in the signal space, which are most susceptible to the possibility of being “confused” under the influence of interference.

4. Characteristics of signal reception in telecommunication systems

4.1 Discrimination error probabilities M known signals

Signal detection in radio electronics refers to the analysis of a received oscillation y(t), which ends with a decision about the presence or absence of some useful component in it, which is called a signal. Discrimination of M signals is defined as an analysis of the received oscillation y(t), ending with a decision on which of the M signals belonging to the pre-specified set S(s 0 (t), s 1 (t), ..., s M -1 ( t)) is present in y(t). Signal detection is a special case of distinguishing two signals, one of which is equal to zero over the entire observation interval.

Let the observed fluctuation y(t) be the implementation of a random process that has a distribution W y, i.e. n-dimensional probability density (PD) W(y) [or PD functional W(y(t))], belonging to one of M disjoint classes W i (W i ∩W k =Ø, i≠k, i, k= 0, 1, …, M-1). It is necessary, after observing the implementation of y(t), to decide which of the classes W y belongs to. The assumption that W y Wi is called the hypothesis H i: W y Wi . Decisions that are the result of testing hypotheses will be denoted by , where i(0, 1, ..., M-1) is the number of the hypothesis, the truth of which is declared by the decision made. The analyzed oscillation y(t) is the result of the interaction of the signal s i (t) present in it with the interfering random process (interference, noise) x(t): y(t)=F. From which of M possible signals present in y(t), the PV of the ensemble to which y(t) belongs depends, so that each s i (t) corresponds to a certain class Wi of distributions of the ensemble represented by y(t). Thus, hypotheses H i are interpreted as assumptions about the presence of the i-th (and only the i-th) signal in y(t). In this case, solutions, one of which serves as the result of the discrimination procedure, are statements that the received oscillation contains precisely the i-th signal. Hypotheses H i correspond to classes W i . A hypothesis H i is said to be simple if the class W i contains one and only one distribution. Any other hypothesis is called complex. M complex hypotheses are called parametric if the corresponding classes differ from each other only in the values ​​of a finite number of parameters of the same distribution, described by a known law. Otherwise, the hypotheses are called parametric.

Let us consider the discrimination of M deterministic non-zero signals of the same energy. In this case, the maximum likelihood rule (ML) will be taken as a basis.

optimal in the case when the quality criterion is the sum of conditional error probabilities, or the total error probability with equal posterior probabilities of all signals p i =1/M.

For an arbitrary M, a discriminator adhering to the MP rule considers present in y(t) the signal least distant from y(t) in the sense of Euclidean distance or, which is equivalent for the same signal energies, having the maximum correlation with y(t) . If we consider signals s 0 (t), s 1 (t), ..., s M -1 (t) as a bundle of vectors located in M-dimensional space, then in order to reduce as much as possible the probability of confusing the i-th signal with k -th, you should “spread” the i-th as far as possible and k-th vectors. Thus, the optimal choice of M deterministic signals comes down to searching for such a configuration of a bundle of M vectors in which the minimum Euclidean distance between a pair of vectors would be maximum: mind ik =max (i≠k). Since when the energies are equal, i.e. vector lengths

where ρ ik is the correlation coefficient of the i-th and k-th signals, E is the signal energy, then the requirement for a maximum minimum distance is identical to the condition for a minimum maximum correlation coefficient in the set of signals S(s 0 (t), s 1 (t), ..., s M -1 (t)). The maximum achievable minimum of the maximum correlation coefficient is established quite easily. Summing ρ ik over all i and k, we obtain

where the inequality follows from the non-negativity of the square under the integral. In addition, in the sum on the left, M terms for i=k are equal to one, and the remaining M(M-1) are not more than ρ max =max ρ ik (i≠k). Therefore M+M(M-1)ρ max ≥0 and ρ max ≥-1/(M-1).

A configuration of M vectors in which the cosine of the angle between any pair of vectors is equal to -1/(M-1) is called a regular simplex. If these vectors are taken as M signals, then the resulting deterministic ensemble, with equal probability of all s i (t), will provide a minimum of the total error probability P osh, which solves the issue of the optimal choice of M signals. When М>>1 the relation -1/(М-1)≈0 holds, and therefore, with a large number of distinguishable signals, the orthogonal ensemble is practically no worse than the simplex ensemble in terms of P error.

The sequence of deriving the exact expression for the probability of error in distinguishing M signals with arbitrary ρ ik is as follows. The probability density (PD) of a system of random variables z 0 , z 1 , …, z M -1 is an M-dimensional normal law, to specify which it is enough to know the averages of all z i and their correlation matrix. For averages, if the hypothesis H l is true, we have . The correlation moment of the i-th and k-th correlations is equal to N 0 Eρ ik /2. After the M-dimensional PV is found, its M-fold integral over the region z l ≥z i , i=0, 1, …, M-1, allows us to obtain the probability the right decision provided that H l is true. The sum of such probabilities divided by M (taking into account the equiprobability of the signals) will be the total probability of the correct solution P ex, associated with P osh by the obvious equality P osh =1-P ex. The M-fold integral obtained in this way in a number of important cases can be reduced to one-time So, for any equally correlated (equidistant) signals (ρ ik =ρ, i≠k)

In practical calculations, this expression is rarely used due to the need for numerical integration. Its upper estimate is useful; to derive it we will assume that the hypothesis H l is true. In this case, an error always occurs when at least one of the events z i >z l , i≠l is true. Its probability P osh l , equal to the probability of combining events z i >z l , i≠l, according to the theorem of addition of probabilities,

and by Boole's inequality is not greater than the first sum on the right. Since each term of this sum is the probability of mixing up two signals, then for equidistant signals

Here is the signal-to-noise ratio at the output of the filter matched with s i (t) under the hypothesis Hi, - the probability of mixing up two signals. For equally probable signals (p i =1/M) we arrive at the so-called additive bound on the total error probability

The use of this expression is justified, on the one hand, by the asymptotic convergence of its right side and P osh as the requirements for the quality of discrimination increase (P osh →0), and on the other hand, by the fact that, when choosing the required signal energy (minimum value q) based on on the right side of the expression, the developer always acts with a certain amount of reinsurance, ensuring that the actual probability of error is kept below the figure he accepted in the calculation.

4.2 Discrimination error probabilities M fluctuating signals

The observer is not always aware in detail a priori of the distinguishable signals. More often, he does not know in advance not only the number of the signal present in the analyzed implementation, but also the values ​​of any parameters (amplitude, frequency, phase, etc.) of each of the M possible signals. In this case, the signals themselves are no longer deterministic, since their parameters are not specified; the corresponding discrimination task is called discrimination of signals with unknown parameters.

Let us consider the solution to this problem using the example of distinguishing signals with random initial phases. Such signals are described by the model

s i (t; φ)=Re( i (t)exp),

where f 0 is the known central frequency; φ – random initial phase with a priori PV W 0 (φ); (t) =S(t)e jγ (t) – complex envelope of the signal s(t), which is a realization of s(t; φ) at φ=0: s(t)=s(t; 0); S(t) and γ(t) are the known laws of amplitude and angular modulation. The application of the MP rule must be preceded by the calculation of the likelihood function (functional) W(y(t)|H i), i.e. averaging of the PT W(y(t)|H i, φ), constructed for deterministic signals with a fixed phase φ over all its possible values, taking into account the a priori PT W 0 (φ). With a uniform SW phase W 0 (φ)=1/(2π), |φ|≤π, taking into account the equality of the energies of all distinguished signals, W(y(t)|H i) is a modified Bessel function of zero order:

where c is a coefficient containing factors independent of i, and - correlation module of the complex envelopes of the received oscillation y(t) and the i-th signal. The monotonicity of the function I 0 (·) on the positive semi-axis allows us to move to sufficient statistics Z i and write the MP rule in the form

Thus, the optimal discriminator of M signals of equal energy with random initial phases must calculate all M values ​​of Z i and, if the maximum of them is Z k , decide on the presence of the kth signal in y(t). This means that the signal whose complex envelope has the greatest correlation in magnitude with the complex envelope y(t) is considered to be contained in the observed oscillation y(t).

The exact formulas for the probabilities of errors in distinguishing M arbitrary signals are quite cumbersome even for M = 2, however, in applications, ensembles of signals that are orthogonal in a strengthened sense are more often encountered. The latter means that any two divergent signals s i (t; φ i), s k (t; φ k) are orthogonal for any values ​​of the initial phases:

∫s i (t; φ i)s k (t; φ k)dt=0 for any φ i , φ k and i≠k,

or, equivalently, the deterministic complex envelopes of these signals are orthogonal:

.

The orthogonality condition in a stronger sense is stricter than the usual orthogonality requirement that appeared earlier in application to deterministic signals. Thus, two segments of a cosine wave, shifted by an angle ±π/2, being orthogonal in the usual sense, are not orthogonal when the phase shift changes, i.e. in a stronger sense. At the same time, signals that do not overlap in time or spectrum are orthogonal in a stronger sense.

If we first turn to the distinction between two signals, it is not difficult to understand that the opposite pair, which minimizes P osh in the class of deterministic signals, is unacceptable in problems where the initial phases of the signals are random. Indeed, the only feature by which opposite signals are distinguished is the sign, i.e. the presence or absence of the term π in the initial phase. However, when each signal acquires a random phase shift before entering the discriminator, attempts to use the initial phase as a characteristic feature of the signal are meaningless, and the discriminator has to get rid of the uninformative value φ. Thus, we can come to the conclusion that in the class M≥2 of signals with random phases, simplex ensembles do not have optimal properties. It is the ensembles of signals that are orthogonal in the amplified sense that are optimal: each of such signals causes a response at the output of only one of the filters of the receiving circuit, and therefore the confusion of the i-th signal with the k-th will occur only in the case when the noise envelope at the output k The th matched filter (MF) will have a value exceeding the value of the envelope of the sum of the signal with noise at the output of the i-th MF. Violation of the orthogonality condition in a stronger sense will lead to the appearance of a reaction to the i-th signal at the output of not only the i-th, but also other SF, for example the k-th, resulting in an envelope surge at the output of the k-th SF greater than the value of Z i , will become more likely.

To find the probability of confusion p 01 s 0 (t; φ) with s 1 (t; φ) when distinguishing two signals, it is necessary to integrate the joint PV Z 0, Z 1 under the hypothesis H 0 W(Z 0, Z 1 |H 0) over the region Z 1 >Z 0 . For signals that are orthogonal in the amplified sense, the values ​​Z 0 and Z 1 are independent, therefore W(Z 0 , Z 1 |H 0)=W(Z 0 |H 0)W(Z 1 |H 0). The one-dimensional PVs Z 0 and Z 1 are known: if H 0 is true, Z 0 as the envelope of the sum of the signal with noise has a generalized Rayleigh PV; Z 1 as an envelope of noise only is a Rayleigh random variable. Multiplying these PVs, after integrating the resulting PV W(Z 0 , Z 1 |H 0) and taking into account the obvious equality p 01 =p 10 for the total probability of error in distinguishing two equally probable orthogonal in the amplified sense signals with random phases, we obtain

Repetition of the reasoning of paragraph 4.2. (for deterministic signals) leads to an additive bound

which, as a rule, is used to estimate the probability of error if the number of equally probable orthogonal signals in the amplified sense is M≥2.

4.3 Calculation of discrimination errors M signals with unknown non-energy parameters

Let us consider the problem of distinguishing “M” orthogonal signals with an unknown time position in asynchronous communication systems with code division of channels. The decision about the presence of a signal in the channel is made using the maximum likelihood method. Let us find the probability of discrimination error taking into account noise emissions in the interval of possible time delays of signals.

Let us assume that there are “M” subscribers of a communication system, each of which uses its own signal. Simplex signals provide the greatest noise immunity when transmitting information under such conditions. When M>>1, the noise immunity of such a signal system practically coincides with the noise immunity of a system of orthogonal signals, for which

Here E kf is the energy of the signal f k . The orthogonality condition, which can be called “orthogonality at a point,” in practice requires a uniform time system to organize synchronous communication. In asynchronous systems, signals that are orthogonal in the amplified sense are used, for which, for all values ​​of τ k and τ m

If R km (τ k , τ m)<0.25 – 0.3, то можно считать ансамбль сигналов практически удовлетворяющим условию ортогональности.

We will consider a system of complex signals (f k (t)), k=1...M orthogonal with an arbitrary shift. Among complex signals, phase-shift keyed (PM) signals with a complex envelope of the form

where a i is the sequence code, u 0 (t) is the shape of the envelope of the elementary parcel, Δ is its duration. In the case of a rectangular shape of the envelope of the elementary parcel, the autocorrelation function (ACF) has the form:

Here R 0 (τ)=(1-|τ|/Δ). In the vicinity of the ACF maximum R(τ)= R 0 (τ)=(1-|τ|/Δ). At the receiver input, after passing through a multipath channel, the useful signal can be written as

δ n is the relative delay of the signal along the beam with number n, τ is the unknown arrival time, which is inside the interval. ε n =A n /A 0 – relative amplitude of the “n” beam, parameter ν has the meaning of the number of additional propagation rays. Relative delays δ n >Δ, i.e. the beams are separated when processing a complex signal. When ν=0, the signal has the form s(t)=A 0 f(t-τ 0).

Let's consider the processing algorithm. A mixture is supplied to the receiver input

x(t)=s k (t-τ 0k)+η(t), (t),

where s k (t) is one of the possible signals, k=1...M, τ 0 k is the time delay of the signal, η(t) is white Gaussian noise with zero average value and power spectral density N 0 /2. It is necessary to make a decision which of M possible signals is present at the receiver input. Let's consider a receiver without multipath compensation. The linear part of such a receiver contains M channels in which statistics of the form

The expression for L k (τ k) can be rewritten in a form more convenient for analysis

Here and in subsequent formulas, the index k is omitted for brevity if the characteristics of one channel are studied, z 0 2 =2A 0 2 E f /N 0 – signal-to-noise energy ratio, S(τ-τ 0)=∫f(t-τ ) f(t-τ 0)dt/E f – normalized signal function, N(τ)=∫n(t)f(t-τ)dt – normalized noise function with zero mean, unit variance and correlation function =S(τ"-τ""). The envelope of the signal function S(τ-τ 0) is the ACF.

According to the maximum likelihood algorithm, a decision in favor of signal number m is made if supL m (τ m)≥supL k (τ k). To find the probabilities of correct and incorrect decisions using this rule, it is necessary to calculate the distribution of absolute maxima of the processes L(τ) on the interval [T 1, T 2].

Let us consider a technique for calculating the probability of error in distinguishing M signals with unknown parameters during single-path signal propagation (or in an optimal signal combining scheme). Let us denote by H k =supL k (τ k) the value of the absolute maximum of statistics at the output of the kth channel of the receiver. We write the joint distribution of random variables (H 1 ,H 2 ,..H M ) as w(u 1 ,u 2 ,..u M). The orthogonality condition for signals f k (t) in a statistical sense means the independence of random variables H k , k=1..M. Then the probability of a correct decision using the maximum likelihood algorithm can be written

If we take into account the condition of orthogonality of the signal system (s k (t)), then

Let us assume that the system of signals (s k (t)) has the same energy, that is, z 0 m =z 0 k =z 0 . Then the formulas for H m and H k can be rewritten in the form

The distribution function of the absolute maximum h k of the implementation of a Gaussian process with the correlation function R(τ) can be approximated by the formula

ξ=(T 2 -T 1)/Δ is the reduced length of the a priori interval [T 1,T 2], which has the meaning of the resolution number of PM signals in this interval. The approximation is asymptotically exact for ξ→∞, u→∞. For finite values ​​of ξ and u, a more accurate approximation can be used

Probability integral. For ξ>>1 and z 0 >>1, the distribution function of the absolute maximum h m can be written as F m (u)=F s (u)F N (u)≈Φ(u-z 0)F N (u). Substituting the expressions F N (u) and F m (u) into the relation for P rights, we obtain after appropriate transformations

The first term corresponds to the a priori probability of the correct solution for M equally possible events. The second term determines changes in probability due to decision making. As z 0 →∞, the integral in the expression for P rights tends to 1 and, accordingly, P rights →1.

The total probability of error in distinguishing M signals with unknown parameters is equal to

It is clear from the formulas that with an increase in the number of distinguished signals, the probability of decision error P e (z 0) increases. With an increase in the a priori interval of signal time delays ξ, the probability of discrimination error P e (z 0) increases significantly.

4.4 Comparison of synchronous and asynchronous communication systems

Typically, when considering receiver or demodulator performance, some level of signal synchronization is assumed. For example, coherent phase demodulation (PSK) assumes that the receiver can generate reference signals whose phase is identical (perhaps up to a constant offset) to the phase of the transmitter signal alphabet elements. Then, in the process of making a decision regarding the value of the received symbol (using the maximum likelihood principle), the reference signals are compared with the incoming ones.

When generating such reference signals, the receiver must be synchronized with the receiving carrier. This means that the phase of the incoming carrier and its copy at the receiver must be consistent. In other words, if there is no information encoded in the incoming carrier, the incoming carrier and its copy at the receiver will pass through zero at the same time. This process is called phase-locked loop (this is a condition that must be met as closely as possible if we want to accurately demodulate coherently modulated signals at the receiver). As a result of phase-locked loop, the local oscillator of the receiver is synchronized in frequency and phase with the received signal. If the carrier signal directly modulates a subcarrier rather than a carrier, both the carrier phase and the subcarrier phase need to be determined. If the transmitter does not phase-lock the carrier and subcarrier (which is usually the case), the receiver will be required to generate a copy of the subcarrier, with the phase control of the subcarrier copy being controlled separately from the phase control of the copy carrier. This allows the receiver to obtain phase locking on both the carrier and subcarrier.

In addition, the receiver is assumed to know exactly where the incoming symbol begins and where it ends. This information is needed to know the appropriate symbol integration interval—the energy integration interval—before making a decision regarding the meaning of the symbol. Obviously, if the receiver integrates over an interval of inappropriate length or over an interval spanning two symbols, the ability to make an accurate decision will be reduced.

It can be seen that symbol and phase synchronization have in common that both involve creating a copy of part of the transmitted signal at the receiver. For phase locking, this will be an exact copy of the carrier. For symbolic, this is a meander with a transition through zero simultaneously with the transition of the incoming signal between symbols. A receiver capable of doing this is said to have symbol synchronization. Since there are usually a very large number of carrier periods per symbol period, this second level of synchronization is much coarser than phase synchronization and is usually accomplished using a different circuit than that used in phase synchronization.

Many communications systems require an even higher level of synchronization, commonly called frame synchronization. Frame synchronization is required when information is delivered in blocks, or messages containing a fixed number of characters. This occurs, for example, when block code is used to implement a forward error protection scheme or when the communication channel is time-shared and used by several users (TDMA technology). With block coding, the decoder must know the location of the boundaries between codewords, which is necessary for correct decoding of the message. When using a time division channel, you need to know the location of the boundaries between users of the channel, which is necessary for the correct direction of information. Like symbol synchronization, frame synchronization is equivalent to the ability to generate a square wave at frame rate with zero transitions coinciding with the transitions from one frame to the next.

Most digital communications systems using coherent modulation require all three levels of synchronization: phase, symbol, and frame. Non-coherent modulation systems typically require only symbol and frame synchronization; since the modulation is non-coherent, precise phase locking is not required. In addition, incoherent systems require frequency synchronization. Frequency synchronization differs from phase synchronization in that the copy of the carrier generated by the receiver can have arbitrary phase shifts from the received carrier. The receiver structure can be simplified if there is no requirement to determine the exact phase value of the incoming carrier. Unfortunately, this simplification entails a deterioration in the dependence of transmission reliability on the signal-to-noise ratio.

Until now, the focus of the discussion has been on the receiving end of the communication channel. However, sometimes the transmitter takes a more active role in synchronization - it changes the timing and frequency of its transmissions to match the receiver's expectations. An example of this is a satellite communications network, where many ground terminals send signals to a single satellite receiver. In most of these cases, the transmitter uses the reverse communication channel from the receiver to determine timing accuracy. Therefore, successful transmitter synchronization often requires two-way communication or networking. For this reason, transmitter synchronization is often called network synchronization.

There is a cost associated with the need to synchronize the receiver. Each additional level of synchronization implies greater system cost. The most obvious investment is the need for additional software or hardware for the receiver to acquire and maintain synchronization. Also, and less obviously, we sometimes pay in the time spent synchronizing before communication begins, or in the energy required to transmit the signals that will be used at the receiver to obtain and maintain synchronization. At this point, one may wonder why a communications system designer should even consider a system design that requires a high degree of synchronization. The answer: improved performance and versatility.

Consider a typical commercial analog AM radio, which can be an important part of a broadcast communications system that includes a central transmitter and multiple receivers. This communication system is not synchronized. At the same time, the receiver bandwidth must be wide enough to include not only the information signal, but also any carrier fluctuations resulting from the Doppler effect or drift in the transmitter reference frequency. This transmitter bandwidth requirement means that additional noise energy is delivered to the detector in excess of the energy theoretically required to transmit information. Slightly more sophisticated receivers containing a carrier frequency tracking system may include a narrow bandpass filter centered on the carrier, which will significantly reduce noise energy and increase the received signal-to-noise ratio. Therefore, although conventional radio receivers are quite suitable for receiving signals from large transmitters over distances of several tens of kilometers, they may not work under lesser conditions.

For digital communications, trade-offs between receiver performance and complexity are often considered when choosing modulation. The simplest digital receivers include those designed for use with binary FSK with non-coherent detection. The only requirement is bit synchronization and frequency tracking. However, if you choose a coherent BPSK scheme as modulation, you can get the same bit error probability, but with a lower signal-to-noise ratio (by about 4 dB). A disadvantage of BPSK modulation is that the receiver requires precise phase tracking, which can present a design challenge if the signals have high Doppler rates or are prone to fading.

Another trade-off between price and performance involves error correction coding. Significant performance improvements are possible when appropriate error-proofing techniques are used. At the same time, the price, expressed in the complexity of the receiver, can be high. Proper operation of a block decoder requires the receiver to achieve block, frame, or message synchronization. This procedure is in addition to the normal decoding procedure, although there are certain error correction codes that have built-in block timing. Convolutional codes also require some additional synchronization to obtain optimal performance. Although performance analysis of convolutional codes often makes the assumption that the length of the input sequence is infinite, in practice this is not the case. Therefore, to ensure a minimum probability of error, the decoder must know the initial state (usually all zeros) from which the information sequence begins, the final state, and the time to reach the final state. Knowing when the initial state ends and the final state is reached is equivalent to having frame synchronization. In addition, the decoder must know how to group the channel symbols to make a branching decision. This requirement also applies to synchronization.

The above discussion of trade-offs was done in terms of the tradeoff between the performance and complexity of individual channels and receivers. It is worth noting that the ability to synchronize also has significant potential implications related to system efficiency and versatility. Frame synchronization allows the use of advanced, universal multiple access techniques such as demand-based multiple access (DAMA) schemes. In addition, the use of spread spectrum techniques, both multiple access and interference suppression schemes, requires a high level of system synchronization. These technologies offer the ability to create highly versatile systems, which is a very important feature when the system changes or is subject to intentional or unintentional interference from various external sources.

Conclusion

The first section of my work describes the principles of constructing wireless telecommunications communication systems: a diagram of the construction of a cellular communication system is given, methods for separating subscribers in cellular communications are indicated and the advantages (confidentiality and noise immunity) of code separation are noted compared to time and frequency, and also common wireless standards are considered DECT, Bluetooth and Wi-Fi communications (802.11, 802.16).

Next, the correlation and spectral properties of signals are considered and, for example, calculations of the spectra of some signals (rectangular pulse, Gaussian bell, smoothed pulse) and autocorrelation functions of Barker signals and Walsh functions common in digital communications are given, as well as types of complex signals for telecommunication systems are indicated.

The third chapter presents methods for modulating complex signals: phase shift keying methods, modulation with minimum frequency shift (one of the modulation methods with continuous phase), quadrature amplitude modulation; and their advantages and disadvantages are indicated.

The last part of the work contains a consideration of the probabilities of errors in distinguishing M known and M fluctuating signals against a background of interference, as well as an algorithm for calculating errors in distinguishing M orthogonal signals with an unknown time position in asynchronous communication systems with code division.

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8. Radchenko Yu.S., Radchenko T.A. Efficiency of code separation of signals with unknown arrival time. Proceedings of the 5th international. conf. “Radar, navigation, communications” - RLNC-99, Voronezh, 1999, vol. 1, p. 507-514.

9. Radio engineering systems: Textbook. for universities for special purposes “Radio Engineering” / Yu.P. Grishin, V.P. Ipatov, Yu.M. Kazarinov and others; Ed. Yu.M. Kazarinova. – M.: Higher. school, 1990. – 469 p.

Timely transmission of information is the basis for the stable functioning of many industries and agriculture.

The modern information society actively uses various telecommunication systems to exchange large amounts of information in a short time.

Modern telecommunication systems and networks

Telecommunication systems are technical means, designed to transmit large volumes of information via fiber optic communication lines. As a rule, telecommunication systems are designed to serve a large number of users: from several tens of thousands to millions. The use of such a system involves the regular transfer of information in digital form between all participants in the telecommunications network.

main feature modern equipment for networks - ensuring uninterrupted connections so that information is constantly transmitted. In this case, periodic deterioration in the quality of communication at the time of connection establishment is allowed, as well as periodic technical problems caused by external factors.

Types and classification of telecommunication communication systems

Modern telecommunication systems are combined according to several main characteristics.

Depending on the purpose, television broadcasting systems, personal communications systems, and computer networks differ.

Depending on the technical support, which is used to transmit information, there are traditional cable communication systems, more advanced ones - fiber optic, as well as terrestrial and satellite.

Depending on the method of encoding an array of information, analogue and digital communication channels are distinguished. The latter type has become widespread, while analog communication channels are becoming less and less popular today.

Computer systems

Computer systems are a collection of several PCs combined into a single information field through cables and specialized programs.

The totality of installed equipment and software is an autonomous self-regulating system that serves the enterprise as a whole.

Depending on its functions, computer system equipment is divided into:

• service (for intermediate and backup storage of information);


• active (to ensure timely and high-quality delivery of signals;


• personal devices.

To ensure the operation of the entire system, appropriate software is required, properly configured based on the needs of the users.

Radio and television systems

Radio engineering message transmission systems are based on electromagnetic oscillations, which are broadcast over a special radio channel. The unit of system functioning is a signal that is converted in the transmitting device and then transformed into an information message in the receiving device.

The basis for the uninterrupted functioning of radio systems is the communication line - the physical environment and hardware that ensure timely and complete transmission of information.

Television systems operate on a similar principle of receiver and transmitter. Most of them use a digital signal, which allows the message to be transmitted in higher quality.

Global telecommunication systems

Global telecommunications systems include those hardware and software that connect users regardless of their physical location on the planet. The main feature of global networks is intelligentization, which makes it easy to use network capacity with optimal efficiency, while minimizing equipment maintenance costs. Among global networks, there are several main types.

Digital networks with integrated modules use continuous circuit switching, while data arrays are processed in digital form. Network users have access only to some functions; the interface does not allow them to independently change technical parameters.

X25 networks are the oldest, most reliable and proven technologies for transmitting information between an unlimited number of users. The main difference between such networks is the presence of a device for “assembling” individual blocks of transmitted information into “packets” for the fastest transmission.

Asynchronous data transfer mode is a modern technology used for broadband networks that are based on fiber optic cables.

Optical telecommunication systems

The basis of optical telecommunication systems is a fiber optic cable that connects individual devices into a single global network.

Signals are transmitted using infrared radiation, and the throughput of a fiber optic cable is many times greater than that of other types of equipment.

The technical characteristics of the material provide a low level of signal attenuation over long distances, which allows the cable to be used for communication between continents. Layed along the ocean floor, the fiber optic cable is protected from unauthorized access, since intercepting transmitted signals is quite difficult from a technical standpoint.

Multichannel telecommunication systems

A distinctive feature of such communication systems is the use of several channels for transmitting information signals.

Modern telecommunication systems use cable, waveguide, radio relay, and space communication lines. The encrypted signal is transmitted at speeds of several gigabits per second over vast distances.

The main advantage of multi-channel systems is to ensure stable operation. When one communication channel fails, the next one is automatically connected.

Users are protected from sudden connection interruption and loss important information. Such systems are based on structured cable structures.

Multiservice telecommunication systems

Multiservice telecommunication systems are a hardware and software environment designed to transmit data using packet switching technology - connecting individual blocks of information into large messages.

A feature of multiservice systems is the need to ensure stable operation of all elements of the transport environment. As a rule, different technologies are used to transmit data, as well as voice and video information, but the infrastructure is the same. Therefore, the main principle of building multiservice networks is the universality of the technological solution, with the help of which heterogeneous equipment designed to perform various operations is serviced.

A multiservice system uses a single channel to transmit various types of data. This saves money on system maintenance and hardware: a single design requires less personnel and costs.

Structure, equipment and components of telecommunication systems

At the heart of any telecommunications system are servers on which the information needed by users is stored and processed.

Server rooms are small rooms with industrial ventilation, ensuring the functioning of many hard drives large volume.

User computers are a means of communication between the database and specific users of information carrying out search queries.

The technical basis of telecommunication networks is communication lines, that is, data transmission media, which use fiber optic, coaxial or wireless communication channels.

Network equipment providing data transmission and reception:

• modems;
• adapters;
• routers;
• hubs.

Such devices complement the telecommunications system and are necessary for stable operation.

The software allows you to effectively control the operation of installed equipment, which ensures timely transfer of information in the required volumes.

Methods and measuring instruments in telecommunication systems

Depending on the stage of implementation, three types of measurements are distinguished:

1. Installation measurements are carried out after installation of the equipment to ensure the functionality of all components of the telecommunications system.


2. During the work, it is necessary to carry out adjustment measurements that allow you to adapt the functionality of the equipment to changing environmental conditions. For example, if hardware or software changes are made to a telecommunications system, you need to ensure that it continues to function fully.


3. Control or preventive measurements are carried out regularly in order to prevent sudden breakdowns of the telecommunications network.

Basics of construction and installation of telecommunication systems and networks

The main principle of building a telecommunications system of any size and purpose is its division into separate functional sections. The maintenance time for each of them is reduced, and the procedure for finding the location of a breakdown in the event of any technical malfunction is simplified.

In addition, when installing systems, it is necessary to take care of the insulation of the cable itself so that data transmission is as little dependent as possible on external factors. Modern fiber optic cables are located underground, on the ocean floor, or in special corrugations, which protects them as much as possible from harmful influences.

Ensuring information security of telecommunication systems

The main task when building a security system in telecommunications is to prevent information leakage through individual channels. The cause of such phenomena may be hardware damage to the transmitting channel (fiber optic cable), or an attack by intruders using software.

In the first case, information security consists of providing high-quality cables that can withstand heavy loads and regular use.

The second requires the development, implementation and maintenance of software that limits access to the resources of the telecommunication system.

Hotel telecommunication systems

The hotel business represents a whole range of services that provide comfortable accommodation for guests on the hotel premises. That is why timely provision of complete and reliable information about everything that may interest guests is a guarantee of customer retention.

As a rule, telecommunication systems in hotel complexes consist of:

• video communications;
• computer systems;
• software.

Thus, each guest receives the convenience of staying in the room and all the necessary information.

Telecommunication systems and railway transport networks

Unlike the hospitality industry, the main priority of telecommunications in the railway sector is the reliability of information. Therefore, telecommunication networks in railway transport are designed in such a way that all transmitted information can be quickly tracked, while minimal attention is paid to possible leaks.

Companies serving telecommunication systems

Servicing of telecommunication systems is carried out by suppliers of equipment for conducting data communications and service companies.

Among the enterprises we can note:

• "Telecommunication Systems" is one of the oldest specialized companies in St. Petersburg, providing clients with services current repairs, setting up and maintaining information transmission systems;


• "Stroykom-A" is a small company providing services for maintaining and improving dilapidated telecommunication systems;


• "Cryptocom" is a narrow-profile company engaged in ensuring security in telecommunication systems of defense industry enterprises.

Manufacturers and suppliers of equipment for telecommunication systems

The following companies produce and supply equipment for telecommunication systems:

• Montair is a provider of turnkey solutions for telecommunication systems, offering customers big choice server equipment.


• Rdcam is a full-cycle company that offers customers not only ready-made equipment, but also the development of engineering solutions for telecommunication systems.


• "LAN-ART" is a supplier of network switching equipment and a manufacturer of communication cables.

Modern telecommunication systems and specialized communication equipment are demonstrated at the annual exhibition “Svyaz”.

Part 1

TELECOMMUNICATION AND INFORMATION NETWORKS

Chapter 1 ______

TELECOMMUNICATION NETWORKS AND SYSTEMS. GENERAL PROVISIONS

List of abbreviations

BASIC CONCEPTS OF TELECOMMUNICATION NETWORKS AND SYSTEMS

The modern development of communication technology is characterized by two features: the digital form of representation of all signals - regardless of what type of information is represented by these signals - speech, text, data or image; service integration, which can only be fully realized by transferring communications to digital technology. Information transmission and switching systems are being integrated, and the tasks of terminal devices and communication networks are being redistributed in a new way. Multifunctional terminal devices are being created that differ from telephone and telegraph devices, terminal devices for visual display of data, suitable for more than one type of information. Finally, the communication network allows voice, text, data and images to be transmitted over the same connection: the user will access this network regardless of the type of service through the “communications plug”.

With the help of these “revolutionary” means, the productivity and economic efficiency of both entire organizations and individuals were significantly increased. The conclusion suggests itself that the combined efforts of three industries - the computer industry (information technology), consumer radio electronics (entertainment industry) and telecommunications - brought closer the achievement of the main goal - the creation of a global information infrastructure(GII, GII).

The ultimate goal of the GII is to guarantee every consumer access to the information community.

There are some fundamental characteristics that GII must have in order to meet the requirements of information consumers. These characteristics are called attributes. Proposed

For each type of information messages it is traditionally used specific way transmission in a network, characterized by the principle of converting a message into a telecommunication signal and the type of communication (form of communication). Thus, for the transmission of audio information, the accepted form of communication is telephone, for the transmission of still images, facsimiles are used, and for moving images, television is used. Data refers to a type of encoded messages, the method of transmission of which is based on the representation of each information element (letter, sign, number) in the form of a code combination transmitted in the form of a signal over the network. For coded messages, the telegraph method of transmitting information and data transmission is used. Recently, so-called “multi-medium” forms of communication have been used - multimedia (translated from English. milty- a lot of, media- medium) for simultaneous transmission of sound, image and data.

Depending on the form of communication, telecommunication systems can be divided into telephone communication systems, fax communication, television broadcasting, telegraph communication, data transmission, etc.; depending on the signal transmission medium (copper, ether, optical fiber) - into telecommunication and optical communication systems, as well as wired communications using guide media (copper and optical cables), and wireless communications, where ether is used to transmit signals. It is necessary to emphasize what unites all these systems into the general concept of a telecommunications system:

1. The general purpose of all communication systems is to provide services to users.

2. All communication systems belong to the type of distributed systems, the main component of which is a telecommunications network, which allows the use of general principles of structural optimization of such systems.

3. Communication systems, like any complex systems, cannot be considered in isolation from the external environment. The external environment is understood as a set of elements of any nature that exist outside the system and have certain impacts on it. Such elements in relation to any communication system include users who determine the requirements for the volume of services consumed, their list, quality, and thereby impact the communication system.

It should be noted that the very concept of “system” is abstract in relation to the real object, which is associated with it and can be interpreted as a model of the object. The model allows you to reflect the most important components of an object and omit details that are insignificant from the point of view of the purpose of its consideration. In this regard, the same object can be characterized differently by different systems depending on the aspects of its consideration.

When considering the models of most networks and telecommunications systems, the concepts of protocol and interface are widely used. A protocol is a set of rules and formats that define the interaction of objects of the same network levels, for example, “person - person”, “terminal - terminal”, “computer - computer”, “process - process”, i.e. protocols that describe the order of interaction between users, terminals, network nodes or separate networks. In this case, the same language, the same syntactic rules and information formats must be used. The level structure of the model allows for independent development of protocols. Each layer of the model can have multiple protocols. Interaction between adjacent levels is provided by interfaces. An interface is a set of technical and software tools used to interface devices, systems or programs. A set of means for interaction between two adjacent levels (inter-level interface) contains rules for logical and electrical coordination, as well as a detailed description of message formats.

Information networks are designed to provide users with services related to the exchange of information, its consumption, processing, storage and accumulation. A consumer of information who has gained access to an information network becomes a user. Users can be both individuals and legal entities(firms, organizations, enterprises). Using the network provides the opportunity to obtain information when it is needed. An information network is understood as a set of geographically dispersed end systems that are united into telecommunication networks and provide access to any of these systems to all network resources and their collective use. It is advisable to divide telecommunication networks according to the type of communications (telecommunication networks, optical communications, telephone communications, data transmission, rail or air communications, etc.).

Terminal systems of an information network can be classified as: - terminal (terminal system), providing access to the network and its resources;

Workers (server, host system), representing information and computing resources;

Administrative (management system), implementing management of the network and its individual parts.

Information network resources are divided into information, data processing and storage, software and communication.

Informational resources- this is information and knowledge accumulated in all areas of science, culture and society, as well as products of the entertainment industry. It's all a system

is organized in network databases with which network users interact. These resources determine the consumer value of the information network and must not only be constantly created and expanded, but also update outdated data in a timely manner.

Processing and storage resources data are determined by processor performance network computers and the volume of their storage devices (memory), as well as the time during which they are used.

Software Resources represent software (software) involved in the provision of services to users, as well as programs of related functions. The latter include: issuing invoices, accounting for payment for services, navigation (ensuring information search on the network), maintaining network electronic mailboxes, organizing a bridge for teleconferences, converting the formats of transmitted messages, cryptoprotection of information (encoding and encryption), authentication ( electronic signature documents certifying their authenticity).

Communication Resources participate in the transportation of information and redistribution of flows in the switching node. These include the capacity of communication lines, the switching capabilities of nodes, as well as the time they are occupied during user interaction with the network. Communication resources are classified according to the type of vehicle: public switched telephone network, packet data network, network mobile communications, television and radio broadcasting networks, digital integrated service network, etc.

Telecommunication networks are usually assessed by a number of indicators that reflect the ability to efficiently transport information. The ability to transmit information to a vehicle is related to the degree of its operability, i.e., the performance of specified functions in a specified volume at the required level of quality during a certain period of network operation or at an arbitrary point in time. ->operability of a communication network is determined by the concepts of reliability and survivability. The difference between these concepts is due to the reasons and factors that disrupt the normal operation of the network, and the nature of the violations.

Reliability A communication network is characterized by its ability to provide communication while maintaining over time the values ​​of “established quality indicators under given operating conditions. It reflects the ability to maintain the functionality of a communication network when exposed mainly to internal factors - random failures of technical equipment caused by aging processes, manufacturing technology defects or maintenance personnel errors.

Vitality a communication network characterizes its ability to maintain full or partial operability when exposed to causes outside the network and leading to destruction or significant damage to some of its elements (points and communication lines). Such reasons can be divided into two classes: spontaneous And deliberate. Natural factors include:

such as earthquakes, landslides, river floods, etc., and deliberate ones - nuclear missile strikes, sabotage actions, etc.

When analyzing vehicle capacity, the concepts of call and message are very important. A call is a connection between two network users to transmit a message. Message- user formation converted into telecommunication signals. Considering the difference between a call and a message, we can say that a flow of calls arrives at a network node or some part of it, and a flow of messages circulates in communication networks to transmit information to the user. The need to deliver messages from one point in the network to another can be expressed by the gravity between these points. Gravity characterizes the assessment of the need for various types of communication between two points of the network and is determined by the volume of messages that need to be delivered over a certain period of time from one point to another. From the gravity expressed by the volume of messages or the volume of information, you can move * to the gravity expressed by the time of occupation of the communication line (LC), and from it - to the number of necessary 1C. Gravity, determined by the volume of information, is convenient for a data transmission network, and determined by the occupancy of channels - for a telephone network and various types of broadcasting networks. The channel occupation time is divided into hourly occupations per year, day or hour. Gravity depends on the type of information, the territorial location of users, their characteristics, economic, cultural and other relationships. It is impossible to unambiguously determine gravity, since it is influenced by many factors, therefore the accuracy of gravity estimates is usually low.

Amount of information, transmitted between two points over a period of time, is determined by the sum of the volumes of all messages (including repeated ones) or the product of the number of transmitted messages - and the average volume of one message. Time of occupation of lines or devices, expressed in hours of occupation, With"-divides the load on these lines or devices as the product of the total number of incoming calls * g average duration of classes . Load intensity- this is the number of hours of use over a certain period of time, for example, the busiest hour (BHH) is a 60-minute interval during which the load on the network is greater than in any other similar period. Usually the concept of load intensity is used, although for simplicity it is often called load. The dimensionless unit of load intensity is called Erlang. One Erlang is the load intensity sinogo device continuously occupied for an hour.

In the case when the network cannot service the incoming load, it makes sense to talk about the volume of realized load in the network. The amount of realized load is determined by the capacity of the communication network. In some cases, throughput is quantified. For example, by the maximum flow of information that can be skipped between a certain pair of points. In this way, the throughput of the network section is determined, which is the bottleneck when dividing the network between the source and the recipient into two parts.

A point-to-point message flow is a sequence of messages sent from one point to another. In addition to useful information, control and signaling messages that are of no value to the user are transmitted on the network. Significantly load communication networks (without giving any useful effect) and repeated calls, arising in case of failure during the initial call. A message flow is characterized by a sequence of moments in time when each next message arrives. The flow can also be expressed in terms of time intervals between these moments. The type of message flow can also be described by the distribution of the duration of device occupations for each incoming message. All flows circulating in communication networks are divided into deterministic, random and mixed. Deterministic flows are those whose arrival times and message volumes are known in advance. Such streams include almost all broadcast streams (both audio and television), regular transmissions of various reports, etc. For random streams, the moments of arrival, the volumes of individual messages and their addresses are not predetermined and are random variables described using probabilistic distributions. These streams include streams of telephone messages. Depending on the specific conditions, random flows can be very diverse, however, for most practical cases, it is possible to approximate (describe) the durations of the intervals between the arrival of two neighboring messages using known probabilistic distribution laws, which make it possible to obtain a mathematical model of the flow. A mixed flow has both deterministic and random components.

1.2. BOARDS OF DEVELOPMENT OF TELECOMMUNICATION TECHNOLOGIES AND COMMUNICATION SERVICES

In order to find out the prospects for the development of the National Information Infrastructure of Ukraine (SRI) within the framework of the Global Information Infrastructure, it is necessary to understand how this process will proceed in the world, in industrialized countries and in Ukraine, what new infocommunication technologies and services will be offered in the coming years and decades.

The information revolution has become the engine of progress for the entire society. It has long been known that scientific and technological revolutions (STR) radically changed the way of life of mankind and the appearance of the world as a whole. The result of scientific and technological revolution was a sharp increase in population, which should be expected in the next two centuries. Many scientists working in the field of forecasting believe that in the 21st-22nd centuries three scientific and technological revolutions should occur: 1 - informational, 2 - biotechnical, 3 - quantum.

Each of these revolutions will lead to drastic changes in the world. The information revolution will create IT, which will become the technical basis of the global information society. The biotechnological revolution will solve the problem of food supply for the world population, and the quantum revolution will create new efficient and safe sources of energy.

The information revolution (late 20th - early 21st centuries) significantly changed the face of information communications. The main factors in the development of infocommunications of the 21st century are economics, technology and services.

Information and communication technologies and services are derivatives of the economy. In turn, the level of development of technologies and services depends on the level of scientific and technological progress, and their implementation depends on the level of the economy and, first of all, on the effective demand of the population for certain infocommunication services.

In the historical development of communication networks and services, five main milestones can be distinguished (Fig. 1.3). Each milestone has its own development logic and relationship with previous and subsequent stages.

In addition, each milestone depends on the level of economic development and the national characteristics of the individual state.

First milestone- construction of a public telephone network (PSTN, PSTN – public Switched Telephone Network). For a long time, each state created its own national analogue public telephone network. Telephone communication was recommended to the population, institutions, and enterprises and was compared with a single service - the transmission of language messages. Later, data transmission began to be carried out over telephone networks using modems. However, even today, the telephone remains the main telecommunications service that generates more than 80% of telecom operators' profits.

Second milestone- digitalization of the telephone network. To improve the quality of communication services, increase their number, increase the level of automation of management and technological equipment in industrialized countries in the 1970s, work was carried out on the digitalization of primary and secondary networks communications. Integrated digital networks were created IDN (Integral Digital Network), which provide primarily telephone services based on digital switching and transmission systems. By now, in many countries the digitalization of telephone networks has almost been completed.

Third milestone- integration of services. Digitalization of communication networks has made it possible not only to improve the quality of services, but also to increase their number based on integration. This is how the concept of a narrowband digital network with the integration of N-ISDN services emerged (Narrowband Integrated Srsice Digital Network). The user (subscriber) of this network is provided with basic access (2B + D), through which information is transmitted over three digital channels: two channels IN with a transmission rate of 64 kbit/s and channel D with a transmission rate of 16 kbit/s. Two channels IN used to transmit language messages and data, channel th- for signaling and data transmission in packet switching mode. For a user with greater needs, primary access can be provided, which contains (30 B + D) channels. The N-ISDN concept has existed for about 20 years, but has not become widespread in the world for several reasons. Firstly, N-ISDN equipment is quite expensive to become widespread; secondly, the user constantly pays for three digital channels; thirdly, the list of services /U-/50L/ exceeds the needs of the mass user. This is why service integration is beginning to be replaced by the concept of the smart grid.

During the same period, networks with mobile PLMN systems also developed ( Public land Mobil Network) and data network service technologies based on circuit and packet switching: X.25, IP (Internet Protocol) , GR (Frame relay), 1Р-telephony, email, etc.

Fourth milestone- smart network /N (Intelligent Network). The history of this network is usually calculated from 1980, when the Bell System (USA) company carried out work to improve the service called “service-800”. This service was mainly intended for charging long-distance calls to the caller and was widely used in the service sector and trade. Since 1993, IN has been developing within the framework of the concept TINA (Telecommunication Information Networking Architecture) to support the client-server architecture. This network is designed to quickly, efficiently and economically provide information services to the mass user. The required service is provided to the user when and at the time when he needs it. Accordingly, he is obliged to pay for the service provided during this time. Thus, the speed and efficiency of service provision ensure its cost-effectiveness, since if the user uses the communication channel for a significantly shorter period of time, this will allow him to reduce costs. This is the fundamental difference between an intelligent network and previous networks, namely the flexibility and cost-effectiveness of service provision.

Fifth milestone- broadband B-ISND (Droadband Integrated Service Digital Network) pioneered the development of technology-based multimedia services after 1980 ATM (- switching of fixed-length packets (53 bytes): interactive, information and distribution search. Conversational services provide services for transmitting information (telephone service, speech service, video conferencing, etc.). Information retrieval services (query services) provide the user with the opportunity to obtain information from a variety of data banks. Distribution services, with or without control over the provision of information on the part of the user, can send information from one common source to an unlimited number of subscribers who have the right to access (data, text, moving and still images, sound, graphics, etc.). The practice of business communication is beginning to include not only conference calls, but also video conferences, which make it possible to exchange information without wasting time and money on travel.

In turn, reducing individual user costs for new services should increase demand for them, that is, lead to increased profits for service providers. A corresponding increase in demand for services will lead to an increase in the supply of necessary equipment, which will lead to an increase in the profits of equipment suppliers. Thus, the flexibility of providing services using modern technologies leads to the unification of the economic interests of three parties: users, service providers and equipment suppliers.

Control questions

1. Indicate the features of the development of communication technology at the present stage.

2. What is communication integration?

3. Describe multifunctional terminal devices.

4. Define Global Information Infrastructure.

5. What is needed to implement the concept of Global Information Infrastructure?

6. What attributes (characteristics) must be taken into account when creating a Global Information Infrastructure standard?

7. Explain the principles and purpose of the Global Information Infrastructure.

8. Indicate the main characteristics of the Global Information Infrastructure.

9. List the features of building an information network.

10. Explain the structure of the information network.

11. Describe the resources of the information network.

12. How are telecommunication systems divided depending on the type of communication?

13. What indicators of a telecommunications network characterize its effectiveness in transmitting information?

14. Define the concepts of protocol and interface in information networks.

15. What is the reliability of a communication network?

16. Explain the concept of communication survivability; list the factors on which it depends.

17. Describe the capacity of a telecommunications network.

18. What is a challenge?

19. What is meant by the concept of message in a telecommunications network?

20. What parameters determine the volume of information?

21. Name the units of measurement of telephone load and its intensity.

22. What is message flow? Give an example.

23. What information is called useful? Name its other types.

24. What is the message flow characterized by?

25. Name and characterize the flows circulating in communication networks.

26. What are information flows called if the moment of receipt and volume of messages are known in advance? Give an example.

27. What does the concept of “gravity” mean in a communication network?

28. Give a description of the ENSSU, the Ukrainian Research Institute, and the Global Information Infrastructure.

29. Explain the main milestones in the development of communication networks and services.

30. What are the features of the B-ISDN broadband network?

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BRANCH OF THE FEDERAL STATE BUDGET EDUCATIONAL INSTITUTION OF HIGHER PROFESSIONAL EDUCATION

"TYUMEN STATE UNIVERSITY"

IN TOBOLSK

Tobolsk Pedagogical Institute named after. DI. Mendeleev

Department of Physics, Mathematics, Computer Science and Teaching Methods

Course work

Telecommunication systems

5th year student of correspondence course

Faculty of Natural Sciences,

directions “Vocational training

(electronics, radio engineering and communications)"

Sorochenko Alexander Nikolaevich

Teacher: Candidate of Pedagogical Sciences,

Associate Professor Kutumova A. A.

Tobolsk 2016

Introduction

1. Characteristics and classification of information networks

2. Multi-level architecture of information networks

3. Types of communication channels

4. Organization of access to information networks

4.1 Structure of territorial networks

4.2 Main types of access

4.2.1 Telecommunications technology service

4.2.2 Email

4.2.3 File sharing

4.2.4 Teleconferences and bulletin boards

4.2.5 Access to distributed databases

4.2.6 WWW information system

Conclusion

Bibliography

Introduction

Without exaggeration, the 21st century can be called the century of information technology. The concept of information technology includes many aspects. One of the most important parts of this direction is the direct transmission of information through information networks.

Telecommunications technologies are the principles of organizing modern analog and digital communication systems and networks, including computer and INTERNET networks.

Telecommunications means are a collection of technical devices, algorithms and software that allow the transmission and reception of speech, information data, multimedia information using electrical and electromagnetic waves via cable, fiber-optic and radio channels in various wavelengths. These are devices for converting information, encoding and decoding, modulating and demodulating, these are modern Computer techologies processing.

1. Characteristics and classification of information networks

Modern telecommunication technologies are based on the use of information networks.

Communication network is a system consisting of objects that perform the functions of generating, converting, storing and consuming a product, called points (nodes) of the network and transmission lines (communications, communications, connections) that transfer the product between points.

A distinctive feature of the communication network is the large distances between points compared to the geometric dimensions of the areas of space occupied by the points.

An information network is a communication network in which the product of generation, processing, storage and use is information.

A computer network is an information network that includes computing equipment. Components of a computer network can be computers and peripheral devices that are sources and receivers of data transmitted over the network. These components make up the data terminal equipment (DTE or Data Terminal Equipment). Computers, printers, plotters and other computing, measuring and execution equipment of automatic and automated systems can act as DTEs. The actual transfer of data occurs using media and means collectively called a data transfer medium.

The preparation of data transmitted or received by the DTE from the data transmission medium is carried out by a functional block called data channel termination equipment (DCE or DCE - Data Circuit-Terminating Equipment). The AKD can be a structurally separate unit or a unit built into the OOD. The DTE and DCE together constitute a data station, often called a network node. An example of a DCE is a modem.

Computer networks are classified according to a number of criteria.

Depending on the distances between connected nodes, computer networks are distinguished:

Territorial, covering a significant geographical area; Among territorial networks, one can distinguish regional and global networks, having regional or global scales, respectively; regional networks are sometimes called MAN (Metropolitan Area Network) networks, and the common English name for territorial networks is WAN (Wide Area Network);

Local (LAN)? covering a limited area (usually within the distance of stations no more than a few tens or hundreds of meters from each other, less often 1...2 km); local area networks stand for LAN (Local Area Network);

Corporate (enterprise scale)? a set of interconnected LANs covering the territory in which one enterprise or institution is located in one or more closely located buildings. Local and corporate computer networks are the main type of computer networks used in computer-aided design (CAD) systems.

Particularly noteworthy is the unique global Internet network (the World Wide Web (WWW) information service implemented in it is translated into Russian as the World Wide Web); it is a network of networks with its own technology. On the Internet there is the concept of intranets - corporate networks within the Internet.

There are integrated networks, non-integrated networks and subnets. An integrated computer network (Internet) is an interconnected collection of many computer networks, which are called subnets on the Internet.

In automated systems of large enterprises, subnets include computing facilities of individual design departments. Internets are needed to combine such subnets, as well as to combine technical means of automated design and production systems into a single system complex automation(CIM - Computer Integrated Manufacturing).

Typically, Internet networks are adapted for various types of communications: telephony, e-mail, transmission of video information, digital data, etc., and in this case they are called integrated service networks. The development of internetworks consists in the development of means for interfacing heterogeneous subnetworks and standards for building subnetworks that are initially adapted for interfacing. Subnets in internetworks are combined in accordance with the selected topology using interaction blocks.

2. Multi-level architecture of information networks

In general, for the functioning of computer networks, two problems must be solved:

Transfer data to its destination in the correct form and in a timely manner;

The data received by the user for its intended purpose must be recognizable and in the proper form for its correct use.

The first problem is related to routing tasks and is provided by network protocols (low-level protocols).

The second problem is caused by use in networks different types Computers with different codes and language syntax. This part of the problem is solved by introducing high-level protocols.

Thus, a complete end-user-centric architecture includes both protocols.

The developed Open Systems Interconnection (OSI) reference model supports the concept that each layer provides services to the layer above and builds on and uses the services of the layer below. Each level performs a specific data transfer function. Although they must work in strict order, each level allows for several options. Let's consider the reference model. It consists of 7 layers and is a multi-level architecture, which is described by standard protocols and procedures.

The three lower layers provide network services. Protocols that implement these layers must be provided in each network node.

The top four layers provide services to the end users themselves and are thus associated with them rather than with the network.

Physical level. This part of the model defines the physical, mechanical and electrical characteristics of the communication lines that make up the LAN (cables, connectors, fiber optic lines, etc.).

We can think of this layer as being responsible for the hardware. Although the functions of other levels can be implemented in the corresponding chips, they still belong to software. The function of the physical layer is to ensure that symbols entering the physical medium at one end of the channel reach the other end. When using this downstream symbol transport service, the purpose of the channel protocol is to ensure reliable (error-free) transmission of blocks of data over the channel. Such blocks are often called cycles, or frames. The procedure typically requires: synchronization on the first character in the frame, detection of the end of the frame, detection of erroneous characters if any occur, and correction of such characters in some way (usually this is done by requesting retransmission of the frame in which one or more erroneous characters are detected ).

Level channel. The data link layer and the underlying physical layer provide an error-free transmission channel between two nodes in the network. At this level, the rules for using the physical layer by network nodes are determined. The electrical representation of data on the LAN (data bits, data encoding methods, and tokens) are recognized at this level and only at this level. Here errors are detected (recognized) and corrected by requiring data retransmission.

Network level. The function of the network layer is to establish a route for transmitting data over a network, or optionally across multiple networks, from the transmission node to the destination node. This layer also provides flow or congestion control to prevent network resources (node ​​storage and transmission paths) from becoming overwhelmed, which could lead to service failure. When performing these functions at the network level, a service from a lower level is used - a data transmission channel, which ensures error-free arrival of a data block entered into the channel at the opposite end along the network route.

The main task of the lower levels is to transmit data blocks along the route from the source to the recipient, delivering them in a timely manner to the desired end.

Then the task of the upper layers is to actually deliver the data in the correct form and in a recognizable form. These upper layers are unaware of the existence of the network. They provide only the service required of them.

Transport level. Provides reliable, consistent data exchange between two end users. For this purpose, the transport layer uses a network layer service. It also controls the flow to ensure that blocks of data are received correctly. Due to the differences in terminal devices, data in the system can be transmitted from at different speeds, so if thread control is not in effect, slower systems can be overwhelmed by faster ones. When more than one package is being processed, transport layer controls the order of transmission of message components. If a duplicate of a previously received message arrives, this layer recognizes this and ignores the message.

Level session. The functions of this level are to coordinate communication between two application programs running on different workstations. It also provides services to the higher presentation layer. This occurs in the form of a well-structured dialogue. These functions include creating a session, managing the sending and receiving of message packets during a session, and terminating a session. This layer also manages negotiations, if necessary, to ensure correct data exchange. The dialogue between the session service user (ie the presentation layer parties and the higher layer) may consist of normal or accelerated data exchange. It can be duplex, i.e. simultaneous two-way transmission, when each side has the ability to transmit independently, or half-duplex, i.e. with simultaneous transmission in only one direction. In the latter case, special labels are used to transfer control from one side to the other. The session layer provides a synchronization service to overcome any detected errors. With this service, synchronization marks must be inserted into the data stream by users of the session service. If an error is detected, the session connection must be returned to a certain state, users must return to the established point in the dialog flow, reset some of the transferred data, and then re-establish the transfer starting from that point. computer teleconference communication network

Level representation. Manages and transforms the syntax of data blocks exchanged between end users. This situation may occur in PCs of different types (IBM PC, Macintosh, DEC, Next, Burrogh) that need to exchange data. Purpose - transformation of syntactic data blocks.

Applied level. Application layer protocols impart appropriate semantics or meaning to the information exchanged. This level is the boundary between the software and the processes of the OSI model. A message destined for transmission across a computer network enters the OSI model at a given point, passes through Layer 1 (physical), is forwarded to another PC, and travels from Layer 1 in the reverse order to reach the PC on the other PC through its application layer. Thus, the application layer ensures mutual understanding between two application programs on different computers.

3. Types of communication channels

Data transmission medium - a set of data transmission lines and interaction blocks (i.e. network equipment, not included in the data stations) intended for the transfer of data between data stations. Data transmission media can be public or dedicated to a specific user.

Data transmission line is a means that is used in information networks to distribute signals in the desired direction.

Channel (communication channel) - a means of one-way data transmission. An example of a channel could be a frequency band allocated to one transmitter during radio communications.

Data transmission channel - a means of two-way data exchange, including data channel termination equipment and a data transmission line. Based on the nature of the physical data transmission medium (TD), data transmission channels are distinguished on optical communication lines, wired (copper) communication lines and wireless.

Wired communication lines: Wired telecommunication lines are divided into cable, overhead and fiber optic.

Facsimile: Facsimile (or phototelegraph) communication is an electrical method of transmitting graphic information - still image text or tables, drawings, diagrams, graphs, photographs, etc. It is carried out using fax machines: telefaxes and telecommunication channels (mainly telephone).

Fiber optic communication lines: Telephone lines and television cables are mainly used as wired communication lines. The most developed is telephone wire communication. But it has serious disadvantages: susceptibility to interference, attenuation of signals when transmitting them over long distances and low throughput. Fiber optic lines do not have all these disadvantages - a type of communication in which information is transmitted via optical dielectric waveguides ("optical fiber").

Optical fiber is considered the most perfect medium for transmitting large flows of information over long distances. It is made of quartz, which is based on silicon dioxide - a widespread and inexpensive material, unlike copper. The optical fiber is very compact and lightweight, with a diameter of only about 100 microns.

Fiber optic lines differ from traditional wire lines:

Very high speed of information transmission (over a distance of more than 100 km without repeaters);

Security of transmitted information from unauthorized access;

High resistance to electromagnetic interference;

Resistance to aggressive environments;

The ability to simultaneously transmit up to 10 million telephone conversations and one million video signals over one fiber;

Flexibility of fibers;

Small size and weight;

Spark, explosion and fire safety;

Easy installation and installation;

Low cost;

High durability of optical fibers - up to 25 years.

Currently, information exchange between continents occurs primarily through undersea fiber optic cables rather than satellite communications. At the same time, the main driving force behind the development of underwater fiber optic communication lines is the Internet.

Wireless communication systems: Wireless communication systems are carried out over radio channels.

In the 1930s meter, and in the 40s - decimeter and centimeter waves were mastered, propagating rectilinearly, without bending around the earth's surface (i.e. within line of sight), which limits direct communication on these waves to a distance of 40-50 km in flat terrain, and in mountainous areas - several hundred kilometers. Since the width of the frequency ranges corresponding to these wavelengths - from 30 MHz to 30 GHz - is 1000 times greater than the width of all frequency ranges below 30 MHz (waves longer than 10 m), they can transmit huge flows of information and carry out multi-channel communications. At the same time, the limited propagation range and the possibility of obtaining sharp directionality with an antenna of simple design make it possible to use the same wavelengths at many points without mutual interference. Transmission over significant distances is achieved by using multiple relays in radio relay communication lines or using communication satellites located at a high altitude (about 40 thousand km) above the Earth (see "Space communications"). Allowing tens of thousands of telephone conversations to be carried out simultaneously over long distances and dozens of television programs to be transmitted, radio relay and satellite communications are significantly more effective in their capabilities than conventional long-distance radio communications on meter waves.

Radio relay communication lines: Radio relay communications were originally used to organize multi-channel telephone lines in which messages were transmitted using an analog electrical signal. The first such line, 200 km long with 5 telephone channels, appeared in the USA in 1935. It connected New York and Philadelphia.

Over the past decades, the need to transmit data - information represented in digital form - has led to the creation of digital transmission systems. Digital radio relay data transmission systems capable of exchanging digital information have emerged.

Satellite communications and navigation: Space or satellite communication is essentially a type of radio relay communication and is distinguished by the fact that its repeaters are not on the surface of the Earth, but on satellites in outer space.

In the 1980s, the development of personal satellite communications began. At the beginning of the 21st century, the number of its subscribers is several million people, and after another 10 years - much more. There will be a unification of satellite and terrestrial communication systems into a single global personal communication system. Reachability of any subscriber will be ensured by dialing his telephone number, regardless of his location. This is an advantage of satellite communications over cellular communications (discussed later in this chapter), since it is not localized. Indeed, at the beginning of the 21st century, the coverage area of ​​cellular communications is only 15% of the earth's surface. Therefore, the demand for personal mobile communications in many regions of the world can only be met using satellite communication systems. In addition to voice (radiotelephone) communication, they make it possible to determine the location (coordinates) of consumers.

A satellite phone connects directly to a satellite in low Earth orbit. From the satellite, the signal arrives at a ground station, from where it is transmitted to the regular telephone network. The number of satellites required for stable communication anywhere on the planet depends on the orbital radius of a particular satellite system.

The first global communications system, Iridium, is currently in operation. It allows the client to stay in touch wherever he is and use the same phone number.

The system consists of 66 low-orbit satellites located at a distance of 780 km from the Earth's surface. It provides signal reception and transmission from a mobile phone located anywhere in the world. The signal received by the satellite is transmitted along the chain to the next satellite until it reaches the system ground station closest to the called subscriber. This ensures high signal quality.

The main disadvantage of personal satellite communications is its relative high cost compared to cellular communications. In addition, high power transmitters are built into satellite phones. Therefore, they are considered unsafe for the health of users.

The most reliable satellite phones operate on the Inmarsat network, created more than 20 years ago. Inmarsat satellite phones are a flip-top case the size of early laptop computers. The lid of the satellite phone doubles as an antenna, which must be rotated towards the satellite (the signal level is displayed on the phone display). These phones are mainly used on ships, trains or heavy vehicles. Every time you need to make or answer someone's call, you will need to place the satellite phone on some flat surface, open the lid and twist it, determining the direction of the maximum signal. Such satellite phones cost more than $2,500 and weigh from 2.2 kg. A minute of conversation on such a satellite phone costs 2.5 US dollars and more.

Paging: Paging is radiotelephone communication, sending messages dictated by the sending subscriber by telephone and receiving them over the radio channel by the receiving subscriber using a pager - a radio receiver with a liquid crystal display on which received alphanumeric texts are displayed. A pager is a one-way communication device: you can only receive messages on it, but you cannot send messages from it.

The history of paging as a means of personal radio calling began in the mid-1950s in England. The first such device was developed in 1956. The number of subscribers could be no more than 57. When the subscriber received a tone signal, he had to bring the device to his ear and listen in speech form to the message that the dispatcher was transmitting. The users of the first network in England were doctors. The networks that existed at that time were local in nature and served the needs of specific services. The largest of them were airport services. Some similar networks still exist today. Paging began to become widespread in the late 1970s in the United States.

Since then, paging systems have become quite widespread in cities in Europe and the USA. At the same time, paging came to Russia.

The first pagers were simple frequency modulated signal receivers. They contained several tuned circuits that track a characteristic sequence of low-frequency signals (tones). When these tones were received, the device beeped. That's why such pagers are called tonal pagers.

The transition to digital systems was inevitable. Tone coding was not suitable for transmitting alphanumeric messages.

Mobile cellular communication: Communication is called mobile if the source of information or its recipient (or both) move in space. Radio communication has been mobile since its inception. The first radio stations were intended for communication with moving objects - ships. After all, one of the first radio communication devices A.S. Popov was installed on the battleship Admiral Apraksin. And it was thanks to radio communication with him that in the winter of 1899/1900 it was possible to save this ship, lost in ice in the Baltic Sea.

For many years, individual radio communications between two subscribers required their own separate radio communication channel operating on the same frequency. Simultaneous radio communication over many channels could be ensured by allocating a specific frequency band to each channel. But frequencies are also needed for radio broadcasting, television, radar, radio navigation, and military needs. Therefore, the number of radio communication channels was very limited. It was used for military purposes and government communications. Thus, in the cars used by members of the Politburo of the CPSU Central Committee, mobile phones were installed. They were installed in police cars and radio taxis. In order for mobile communications to become widespread, a new idea for its organization was needed. This idea was expressed in 1947 by D. Ring, an employee of the American company Bell Laboratories. It consisted of dividing space into small areas - cells (or cells) with a radius of 1-5 kilometers and separating radio communications within one cell from communications between cells. This made it possible to use the same frequencies in different cells. In the center of each cell it was proposed to locate a base - receiving and transmitting - radio station to ensure radio communication within the cell with all subscribers. Each subscriber has his own micro-radio station - a “mobile phone” - a combination of a telephone, a transceiver and a mini-computer. Subscribers communicate with each other through base stations connected to each other and to the city telephone network.

Each cell must be served by a core radio transmitter with a limited range and a fixed frequency. This makes it possible to reuse the same frequency in other cells. During a conversation, the cellular radiotelephone is connected to the base station by a radio channel through which the phone conversation. The cell size is determined by the maximum communication range between the radiotelephone and the base station. This maximum range is the cell radius.

The idea of ​​mobile cellular communication is that, without yet leaving the coverage area of ​​one base station, the mobile phone falls into the coverage area of ​​any neighboring one, up to the outer border of the entire network zone.

For this purpose, systems of antenna-relays have been created that cover their “cell” - an area of ​​the Earth’s surface. For reliable communication, the distance between two adjacent antennas must be less than their range. In cities it is about 500 meters, and in rural areas - 2-3 km. A mobile phone can receive signals from several repeater antennas at once, but it is always tuned to the strongest signal.

The idea of ​​mobile cellular communications was also to use computer control over the telephone signal from the subscriber when he moves from one cell to another. It was computer control that made it possible to switch a mobile phone from one intermediate transmitter to another within just a thousandth of a second. Everything happens so quickly that the subscriber simply does not notice it.

Computers are the central part of a mobile communication system. They find a subscriber located in any of the cells and connect him to the telephone network. When a subscriber moves from one cell to another, they transfer the subscriber from one base station to another, and also connect the subscriber from a “foreign” cellular network to “their own” when he is within its coverage area - they perform roaming (which in English means "wandering" or "wandering").

The principles of modern mobile communications were an achievement already at the end of the 40s. However, in those days computer technology was still at such a level that its commercial use in telephone communication systems was difficult. Therefore, the practical use of cellular communications became possible only after the invention of microprocessors and integrated semiconductor chips.

An important advantage of mobile cellular communications is the ability to use it outside the general area of ​​your operator - roaming. To do this, various operators agree among themselves on the mutual possibility of using their zones for users. A subscriber, leaving the general area of ​​his operator, automatically switches to the areas of other operators, even when moving from one country to another, for example, from Russia to Germany or France. Or, while in Russia, the user can make cellular calls to any country. Thus, cellular communications provide the user with the opportunity to communicate by telephone with any country, no matter where he is.

Leading cell phone manufacturing companies focus on a single European standard - GSM. That is why their equipment is technically advanced, but relatively inexpensive. After all, they can afford to produce huge quantities of phones that are sold.

A convenient addition to a cell phone is the SMS (Short Message Service) system. It is used to transfer short messages directly to the phone of a modern digital GSM system without the use of additional equipment, only using a numeric keypad and a cell phone display screen. SMS messages are also received on the digital display that any cell phone is equipped with. SMS can be used in cases where a regular telephone conversation is not the most convenient form of communication (for example, on a noisy, crowded train). You can send your phone number to a friend via SMS. Due to its low cost, SMS is an alternative to telephone conversation. The maximum length of an SMS message is 160 characters. You can send it in several ways: by calling a special service, using your GSM phone with a sending function, or using the Internet. The SMS system can provide additional services: send to your GSM phone exchange rates, weather forecast, etc. Essentially, a GSM phone with SMS is an alternative to a pager.

But the SMS system is not the last word in cellular communications. In the most modern cell phones (for example, from Nokia), the Chat function has appeared (in the Russian version - “dialogue”). With its help, you can communicate in real time with other cell phone owners, as is done on the Internet. Essentially, this is a new type of SMS messaging. To do this, you compose a message to your interlocutor and send it. The text of your message appears on the displays of both cell phones - yours and your interlocutor's. Then he answers you and his message is displayed on the displays. So you are having an electronic dialogue. But if the cell phone of your interlocutor does not support this function, then he will receive regular SMS messages.

Cell phones have also appeared that support high-speed Internet access via GPRS (General Packet Radio Service) - a standard for packet data transmission over radio channels, in which the phone does not need to “dial up”: the device constantly maintains a connection, sends and receives data packets. Cellular telephones with a built-in digital camera are also produced.

According to the research company Informal Telecoms & Media (ITM), the number of mobile communications users in the world in 2007 was 3.3 billion people.

Finally, the most complex and expensive devices are smartphones and communicators that combine the capabilities of a cell phone and a pocket computer.

Internet telephony: Internet telephony has become one of the most modern and economical types of communication. Her birthday can be considered February 15, 1995, when VocalTec released its first soft-phone - a program used for voice exchange over an IP network. Microsoft then released the first version of NetMeeting in October 1996. And already in 1997, connections via the Internet between two ordinary telephone subscribers located in completely different places on the planet became quite common.

Why is regular long-distance and international telephone communication so expensive? This is explained by the fact that during a conversation you occupy an entire communication channel, not only when you speak or listen to your interlocutor, but also when you are silent or distracted from the conversation. This happens when voice is transmitted over the telephone using the usual analog method.

With the digital method, information can be transmitted not continuously, but in separate “packets”. Then, information can be sent simultaneously from many subscribers via one communication channel. This principle of packet transmission of information is similar to transporting many letters with different addresses in one mail car. After all, they don’t “drive” one mail car to transport each letter separately! This temporary “packet compaction” makes it possible to use existing communication channels much more efficiently and “compress” them. At one end of the communication channel, information is divided into packets, each of which, like a letter, is equipped with its own individual address. Over a communication channel, packets from many subscribers are transmitted “mixed up”. At the other end of the communication channel, packets with the same address are again combined and sent to their destination. This packet principle is widely used on the Internet.

Through a personal computer you can send and receive letters, texts, documents, drawings, photographs over the Internet. But Internet telephony (IP telephony) works in exactly the same way - a telephone conversation between two personal computer users.

To do this, both users must have microphones connected to the computer and headphones or speakers, and their computers must have sound cards (preferably for two-way communication). In this case, the computer converts the analog “voice” signal (the electrical analogue of sound) into a digital one (a combination of pulses and pauses), which is then transmitted over the Internet.

At the other end of the line, your interlocutor’s computer performs the reverse conversion (digital signal to analog), and the voice is reproduced as in a regular telephone. Internet telephony is much cheaper than long-distance and international calls on a regular telephone. After all, with IP telephony you only need to pay for using the Internet.

Having a personal computer, sound card, a compatible microphone and headphones (or speakers), you can use Internet telephony to call any subscriber who has a regular landline phone. With this conversation, you will also only pay for using the Internet.

Before using Internet telephony, the subscriber who owns a personal computer must install a special program on it.

To use Internet telephony services it is not necessary to have a personal computer at all. To do this, it is enough to have a regular telephone with tone dialing. In this case, each dialed digit goes into the line not in the form of a different number of electrical impulses, as when the disk rotates, but in the form of alternating currents of different frequencies. Such tone mode found in most modern telephones.

To use Internet telephony using a telephone, you need to buy a credit card and call a powerful central computer server at the number indicated on the card. Then the server machine voice (optional in Russian or English language) tells the commands: use the telephone buttons to dial the serial number and card key, dial the country code and the number of your future interlocutor.

Next, the server converts the analog signal into a digital one, sends it to another city, country or continent to a server located there, which again converts the digital signal into analog and sends it to the desired subscriber. The interlocutors talk as if on a regular telephone, however, sometimes there is a slight (a fraction of a second) delay in the response. Let us recall once again that to save communication channels, voice information is transmitted in “packets” of digital data: your voice information is divided into segments, packets, called Internet protocols (IP).

TCP/IP (Transmission Control Protocol / Internet Protocol) is the main Internet protocol, or data transmission format on the Internet. At the same time, IP ensures the promotion of the packet through the network, and TCP guarantees the reliability of its delivery. They ensure that the transmitted data is broken down into packets, each of them is transmitted to the recipient along an arbitrary route, and then assembled in the correct order and without loss.

Not only your packets, but also the packets of several other subscribers are sequentially transmitted over the communication channel. At the other end of the communication line, all your packets are combined again, and your interlocutor hears your entire speech. In order not to feel a delay in the conversation, this process should not exceed 0.3 seconds. This is how information is compressed, thanks to which Internet telephony is several times cheaper than conventional long-distance and, especially, international calls.

In 2003, the Skype program was created (www.skype.com), which is completely free and does not require virtually any knowledge from the user either to install it or to use it. It allows you to talk with video accompaniment to your interlocutors sitting at their computers in different parts of the world. In order for the interlocutors to see each other, the computer of each of them must be equipped with a web camera.

Humanity has come such a long way in the development of communications: from signal fires and drums to a cellular mobile phone, which allows two people located anywhere on our planet to communicate almost instantly.

4. Organization of access to information networks

4.1 Structureterritorialnetworks

The Internet is the largest and only network of its kind in the world. Among global networks, it occupies a unique position. It is more correct to consider it as a union of many networks that retain their independent significance.

Indeed, the Internet has neither a clearly defined owner nor a national identity. Any network can have a connection to the Internet and, therefore, be considered part of it, if it uses the TCP/IP protocols adopted for the Internet or has converters to TCP/IP protocols. Almost all national and regional networks have access to the Internet.

A typical territorial (national) network has a hierarchical structure.

The top level is federal nodes connected to each other by trunk communication channels. Trunk channels are physically organized on fiber-optic lines or satellite communication channels.

Middle level - regional nodes forming regional networks. They are connected to federal nodes and, possibly, to each other by dedicated high- or medium-speed channels, such as T1, E1, B-ISDN channels or radio relay lines.

The lower level is local nodes (access servers) connected to regional nodes, mainly dial-up or dedicated telephone communication channels, although there is a noticeable tendency to move to high- and medium-speed channels.

Local networks of small and medium-sized enterprises, as well as computers of individual users, are connected to local nodes. Corporate networks large enterprises are connected to regional nodes using dedicated high- or medium-speed channels.

4.2 Basickindsaccess

4.2. 1 Telecommunications technology service

The main services provided by telecommunication technologies are:

Email;

File transfer;

Teleconferences;

Help desks (bulletin boards);

Video conferencing;

Access to information resources (information databases) of network servers;

Mobile cellular communications;

Computer telephony.

The specificity of telecommunications is manifested primarily in application protocols. Among them, the best known are Internet-related protocols and ISO-IP protocols (ISO 8473), which belong to the seven-layer open systems model. TO application protocols Internet include the following:

Telnet is a terminal emulation protocol, or, in other words, a remote control implementation protocol used to connect a client to a server when they are located on different computers; the user has access to the server computer through his terminal;

FTP is a file transfer protocol (remote host mode is implemented), the client can request and receive files from the server whose address is specified in the request;

HTTP (Hypertext Transmission Protocol) - a protocol for communication between WWW servers and WWW clients;

NFS is a network file system that provides access to files on all UNIX machines on a local network, i.e. node file systems appear to the user as a single file system;

SMTP, IMAP, POP3 - email protocols.

These protocols are implemented using appropriate software. For Telnet, FTP, SMTP, fixed protocol port numbers are allocated on the server side.

4.2. 2 Email

Electronic mail (E-mail) is a means of exchanging messages via electronic communications (off-line). You can forward text messages and archived files. The latter may contain data (for example, program texts, graphic data) in various formats.

4.2. 3 File sharing

File sharing - access to files distributed across different computers. The Internet uses the FTP protocol at the application level. Access is possible in off-line and on-line modes.

In off-line mode, a request is sent to the FTP server, the server generates and sends a response to the request. In on-line mode, you can interactively view FTP server directories, select and transfer necessary files. An FTP client is needed on the user's computer.

4.2. 4 Teleconferences and bulletin boards

Teleconferences - access to information allocated for group use in separate conferences (newsgroups). Global and local teleconferences are possible. Including materials in newsgroups, sending out notifications about new materials received, and fulfilling orders are the main functions of teleconferencing software. E-mail and on-line modes are possible.

The largest teleconferencing system is USENET. In USENET, information is organized hierarchically. Messages are sent out either in an avalanche or through mailing lists.

Teleconferences can be with or without a moderator. Example: a team of authors working on a book using mailing lists.

There are also audio conferencing (voice teleconferencing) facilities. A call, connection, conversation occurs for the user as in a regular telephone, but the connection is via the Internet.

Electronic "bulletin board" BBS (Bulletin Board System) - a technology similar in functionality to a teleconference, allows you to centrally and quickly send messages to many users.

BBS software combines email, teleconferencing, and file sharing capabilities. Examples of programs that contain BBS tools are Lotus Notes, World-group.

4.2. 5 Access to distributed databases

In client/server systems, the request must be generated in the user's computer, and the organization of data search, their processing and generation of a response to the request belongs to the computer server.

In this case, the necessary information can be distributed across various servers. There are special database servers on the Internet, called WAIS (Wide Area Information Server), which can contain collections of databases managed by various DBMSs.

Typical scenario for working with a WAIS server:

Selecting the required database;

Formation of a query consisting of keywords;

Sending a request to the WAIS server;

Receiving document titles corresponding to specified keywords from the server;

Selecting the desired header and sending it to the server;

Retrieving the document text.

Unfortunately, WAIS is not currently being developed, so it is little used, although indexing and searching by indexes in large arrays of unstructured information, which was one of the main functions of WAIS, is an urgent task.

4.2. 6 WWW Information System

WWW (World Wide Web) is a hypertext information system on the Internet. Its other short name is Web. This more modern system provides users with greater opportunities.

Firstly, this is hypertext - a structured text with the introduction of cross-references that reflect the semantic connections of parts of the text. Link words are highlighted with color and/or underlining. Selecting a link brings up the text or picture associated with the link word on the screen. You can search for the material you need using keywords.

Secondly, it makes it easier to submit and receive graphic images. Information accessible via Web technology is stored in Web servers.

The server has a program that constantly monitors requests from clients arriving at a specific port (usually port 80). The server satisfies requests by sending the client the contents of the requested Web pages or the results of the requested procedures. WWW client programs are called browsers.

Text and graphic browsers are available. Browsers have commands for paging, going to the previous or next document, printing, following a hypertext link, etc.

To prepare materials and include them in the WWW database, special HTML language(Hypertext Markup Language) and software editors that implement it, for example Internet Assistant as part of the Word or Site Edit editor, document preparation is also provided as part of most browsers.

For communication between Web servers and clients, the HTTP protocol based on TCP/IP has been developed. The Web server receives a request from the browser, finds the file that matches the request, and sends it to the browser for viewing.

Conclusion

Intranet and Internet technologies continue to evolve. New protocols are being developed; old ones are being reviewed. NSF added significant complexity to the system by introducing its backbone network, several regional networks, and hundreds of university networks.

Other groups also continue to join the Internet. The most significant change was not due to the addition of additional networks, but due to additional traffic.

Physicists, chemists, and astronomers work and exchange volumes of data larger than the computer science researchers who made up most of the traffic users of the early Internet.

These new scientists caused a significant increase in Internet downloads when they began to use it, and downloads continually increased as they increasingly used it.

To accommodate the growth in traffic, the NSFNET backbone network capacity was doubled, resulting in current capacity approximately 28 times greater than the original; Another increase is planned to bring this ratio to 30.

At present, it is difficult to predict when the need for additional capacity increases will cease. The increase in network exchange needs was not unexpected. The computer industry has taken great pleasure in the constant demands for more computing power and more data storage over the years.

Users are just beginning to understand how to use networks. In the future, we can expect an ever-increasing need for interaction.

Therefore, higher capacity interoperability technologies will be required to accommodate this growth.

The expansion of the Internet lies in the complexity created by the fact that several autonomous groups are part of a unified Internet. The original designs for many subsystems assumed centralized control. It took a lot of effort to refine these projects to work under decentralized control.

So, for the further development of information networks, higher-speed communication technologies will be required.

Bibliography

1. Lazarev V.G. Intelligent digital networks: Handbook/Ed. Academician N.A. Kuznetsova. - M.: Finance and Statistics, 1996.

2. New technologies for information transmission. - URL: http://kiberfix.ucoz.ru. - (Date of access: 12/18/2015).

3. Pushnin A.V., Yanushko V.V.. Information networks and telecommunications. - Taganrog: TRTU Publishing House, 2005. 128 p.

4. Semenov Yu.A. Internet protocols and resources. - M.: Radio and communication, 1996.

5. Telecommunication systems. - URL: http://otherreferats.allbest.ru/radio. - (Date of access: 12/18/2015).

6. Finaev V.I. Information exchanges in complex systems: Textbook. - Taganrog: TRTU Publishing House, 2001.

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