Almost all electrical signals that display real messages contain an infinite spectrum of frequencies. For undistorted transmission of such signals, a channel with infinite bandwidth would be required. On the other hand, the loss of at least one spectrum component during reception leads to distortion of the temporal shape of the signal. Therefore, the task is to transmit a signal in a limited channel bandwidth in such a way that the signal distortion meets the requirements and quality of information transmission. Thus, a frequency band is a limited (based on technical and economic considerations and requirements for transmission quality) signal spectrum.

The frequency bandwidth ΔF is determined by the difference between the upper F B and lower F H frequencies in the message spectrum, taking into account its limitations. Thus, for a periodic sequence of rectangular pulses, the signal bandwidth can be approximately found from the expression:

where t n is the pulse duration.

1.Primary telephone signal (voice message), also called subscriber, is a non-stationary random process with a frequency band from 80 to 12,000 Hz. Speech intelligibility is determined by formants (amplified regions of the frequency spectrum), most of which are located in the band 300 ... 3400 Hz. Therefore, on the recommendation of the International Advisory Committee on Telephony and Telegraphy (ICITT), an efficiently transmitted frequency band of 300 ... 3400 Hz was adopted for telephone transmission. This signal is called a voice frequency (VF) signal. At the same time, the quality of the transmitted signals is quite high - syllable intelligibility is about 90%, and phrase intelligibility is 99%.

2.Audio broadcast signals . Sound sources when transmitting broadcast programs are musical instruments or human voice. The spectrum of the audio signal occupies the frequency band 20...20000 Hz.

For enough high quality(first class broadcast channels) the frequency band ∆F C should be 50...10000 Hz, for flawless reproduction of broadcast programs (highest class channels) - 30...15000 Hz, second class - 100...6800 Hz.

3. In broadcast television a method has been adopted for sequentially converting each image element into an electrical signal and then transmitting this signal through one communication channel. To implement this principle, special cathode ray tubes are used on the transmitting side, converting the optical image of the transmitted object into an electrical video signal unfolded in time.

Figure 2.6 – Design of the transmitting tube

As an example, Figure 2.6 shows a simplified version of one of the transmitting tube options. Inside the glass flask, which is under high vacuum, there is a translucent photocathode (target) and an electronic spotlight (EP). A deflection system (OS) is placed on the outside of the tube neck. The spotlight generates a thin electron beam, which, under the influence of an accelerating field, is directed towards the target. Using a deflection system, the beam moves from left to right (along the lines) and from top to bottom (along the frame), running around the entire surface of the target. The collection of all (N) rows is called a raster. An image is projected onto the tube target, coated with a photosensitive layer. As a result, each elementary section of the target acquires electric charge. A so-called potential relief is formed. The electron beam, interacting with each section (point) of the potential relief, seems to erase (neutralize) its potential. The current that flows through the load resistance R n will depend on the illumination of the target area that the electron beam hits, and a video signal U c will be released at the load (Figure 2.7). The video signal voltage will vary from a “black” level, corresponding to the darkest areas of the transmitted image, to a “white” level, corresponding to the lightest areas of the image.

Figure 2.7 – The shape of a television signal in a time interval where there are no frame pulses.

If the “white” level corresponds to the minimum signal value, and the “black” level corresponds to the maximum, then the video signal will be negative (negative polarity). The nature of the video signal depends on the design and operating principle of the transmitting tube.

The television signal is a pulsed unipolar (since it is a function of brightness, which cannot be multipolar) signal. It has a complex shape and can be represented as the sum of constant and harmonic components of oscillations of various frequencies.
The DC component level characterizes the average brightness of the transmitted image. When transmitting moving images, the value of the constant component will continuously change in accordance with the illumination. These changes are happening very quickly low frequencies(0-3 Hz). Using the lower frequencies of the video signal spectrum, large image details are reproduced.

Television, as well as light cinema, became possible thanks to the inertia of vision. The nerve endings of the retina continue to be excited for some time after the cessation of the light stimulus. At a frame rate F k ≥ 50 Hz, the eye does not notice the intermittency of the image change. In television, the time for reading all N lines (frame time - Tk) is chosen equal to Tk = s. To reduce image flickering, interlaced scanning is used. First, in a half-frame time equal to T p/c = s, all odd lines are read one by one, then, in the same time, all even lines are read. The frequency spectrum of the video signal will be obtained when transmitting an image that is a combination of the light and dark half of the raster (Figure 2.8). The signal represents pulses close in shape to rectangular. The minimum frequency of this signal during interlaced scanning is the frequency of the fields, i.e.

Figure 2.8 – To determine the minimum frequency of the television signal spectrum

With the help of high frequencies, the finest details of the image are transmitted. Such an image can be represented in the form of small black and white squares alternating in brightness with sides equal to the diameter of the beam (Figure 2.9, a), located along the line. Such an image will contain the maximum number of image elements.

Figure 2.9 – To determine the maximum frequency of the video signal

The standard provides for the decomposition of an image in a frame into N = 625 lines. The time to draw one line (Fig. 2.9, b) will be equal to . A signal that changes along the line is obtained when black and white squares alternate. The minimum signal period will be equal to the time it takes to read a pair of squares:

where n pairs is the number of pairs of squares in a line.

The number of squares (n) in the line will be equal to:

where is the frame format (see Figure 2.2.4, a),

b – width, h – height of the frame field.

Then ; (2.10)

The frame format is assumed to be k=4/3. Then the upper frequency of the signal F in will be equal to:

When transmitting 25 frames per second with 625 lines each, the nominal line frequency (line frequency) is 15.625 kHz. The upper frequency of the television signal will be 6.5 MHz.

According to the standard adopted in our country, the voltage of the complete video signal U TV, consisting of synchronization pulses U C, brightness signal and damping pulses U P, is U TV = U P + U C = 1V. In this case, U C = 0.3 U TV, and U P =0.7 U TV. As can be seen from Figure 2.10, the audio signal is located higher in the spectrum (fn 3V = 8 MHz) of the video signal. Typically, a video signal is transmitted using amplitude modulation (AM), and an audio signal using frequency modulation (FM).

Sometimes, in order to save channel bandwidth, the upper frequency of the video signal is limited to the value Fv = 6.0 MHz, and the audio carrier is transmitted at a frequency fн з = 6.5 MHz.

Figure 2.10 – Placement of spectra of image and sound signals in a television broadcast radio channel.

Workshop (similar tasks are included in the exam papers)

Task No. 1: Find the pulse repetition rate of the transmitted signal and the signal bandwidth if there are 5 pairs of black and white alternating vertical stripes on the TV screen

Task No. 2: Find the pulse repetition rate of the transmitted signal and the signal bandwidth if there are 10 pairs of black and white alternating horizontal stripes on the TV screen

When solving problem No. 1, it is necessary to use the known duration of one line of a standard TV signal. During this time, there will be a change of 5 pulses corresponding to the black level and 5 pulses corresponding to the white level (you can calculate their duration). In this way, the pulse frequency and signal bandwidth can be determined.

When solving problem No. 2, proceed from the total number of lines in the frame, determine how many lines are in one horizontal stripe, keep in mind that scanning is carried out interlaced. This way you will determine the duration of the pulse corresponding to the black or white level. Continue as in task No. 1

When preparing the final work, for convenience, use graphic image signals and spectra.

4. Fax signals. Facsimile (phototelegraph) communication is the transmission of still images (drawings, drawings, photographs, texts, newspaper strips, and so on). The fax message (image) conversion device converts the light flux reflected from the image into an electrical signal (Figure 2.2.6)

Figure 2.11 - Functional diagram of fax communication

Where 1 is the fax communication channel; 2 – drive, synchronizing and phasing devices; 3 – transmitting drum, on which the original of the transmitted image on paper is placed; FEP – photoelectronic converter of reflected light flux into an electrical signal; OS – optical system for forming a light beam.

When transmitting elements alternating in brightness, the signal takes the form of a pulse sequence. The frequency of repetition of pulses in a sequence is called the pattern frequency. The pattern frequency, Hz, reaches its maximum value when transmitting an image whose elements and the spaces separating them are equal to the dimensions of the scanning beam:

F rismax = 1/(2τ u) (2.12)

where τ u is the pulse duration equal to the transmission duration of the image element, which can be determined through the parameters of the scanning device.

So, if π·D is the length of the line, and S is the scan pitch (the diameter of the scanning beam), then there are π·D/S elements in the line. At N revolutions per minute of a drum having a diameter D, the image element transmission time, measured in seconds:

The minimum frequency of the picture (when changing along the line), Hz, will be when scanning an image containing black and white stripes along the length of the line, equal in width to half the length of the line. At the same time

F pус min = N/60, (2.14)

To perform phototelegraph communication of satisfactory quality, it is enough to transmit frequencies from F pic min to F pic max. The International Telegraph and Telephony Advisory Committee recommends N = 120, 90 and 60 rpm for fax machines; S = 0.15 mm; D = 70 mm. From (2.13) and (2.14) it follows that at N = 120 F rice max = 1466 Hz; F fig min = 2 Hz; at N =60 F fig max = 733 Hz; F fig min = 1 Hz; The dynamic range of the fax signal is 25 dB.

Telegraph and data signals. Messages and signals of telegraphy and data transmission are discrete.

Devices for converting telegraph messages and data represent each message character (letter, number) in the form of a certain combination of pulses and pauses of the same duration. A pulse corresponds to the presence of current at the output of the conversion device, a pause – to the absence of current.

For data transmission, more complex codes are used, which make it possible to detect and correct errors in the received combination of pulses that arise from interference.

Devices for converting telegraph signals and transmitting data into messages use the received combinations of pulses and pauses to restore message characters in accordance with the code table and output them to a printing device or display screen.

The shorter the duration of the pulses displaying messages, the more of them will be transmitted per unit of time. The reciprocal of the pulse duration is called the telegraphing speed: B = 1/τ and, where τ and is the pulse duration, s. The unit of telegraph speed was called the baud. With a pulse duration of τ and = 1 s, the speed is B = 1 Baud. Telegraphy uses pulses with a duration of 0.02 s, which corresponds to a standard telegraphy speed of 50 baud. Data transfer rates are significantly higher (200, 600, 1200 baud and more).

Telegraphy and data transmission signals usually take the form of sequences of rectangular pulses (Figure 2.4, a).

When transmitting binary signals, it is enough to fix only the sign of the pulse for a bipolar signal, or the presence or absence for a unipolar signal. Pulses can be reliably detected if they are transmitted using a bandwidth that is numerically equal to the baud rate. For a standard telegraph speed of 50 baud, the spectrum width of the telegraph signal will be 50 Hz. At a speed of 2400 Baud (medium-speed data transmission system), the signal spectrum width is approximately 2400 Hz.

5. Average message power P CP is determined by averaging the measurement results over a large period of time.

The average power that a random signal s(t) develops across a 1 Ohm resistor:

The power contained in a finite frequency band between ω 1 and ω 2 is determined by integrating the function G(ω) β within the corresponding limits:

The function G(ω) represents the spectral density of the average power of the process, that is, the power contained in an infinitesimal frequency band.

For convenience of calculations, power is usually given in relative units, expressed in logarithmic form (decibels, dB). In this case the power level is:

If the reference power R E = 1 mW, then p x is called the absolute level and is expressed in dBm. Taking this into account, the absolute level of average power is:

Peak power p peak (ε %) – this is the message power value that can be exceeded for ε % of the time.

The signal crest factor is determined by the ratio of the peak power to the average message power, dB,

From the last expression, dividing the numerator and denominator by RE, taking into account (2.17) and (2.19), we determine the peak factor as the difference between the absolute levels of peak and average powers:

The dynamic range D (ε%) is understood as the ratio of the peak power to the minimum message power P min . The dynamic range, like the crest factor, is usually estimated in dB:

The average power of the voice frequency signal, measured during busy hours (BHH), taking into account control signals - dialing, calling, etc. - is 32 μW, which corresponds to a level (compared to 1 mW) p av = -15 dBm

The maximum telephone signal power, the probability of exceeding which is negligibly small, is 2220 μW (which corresponds to a level of +3.5 dBm); The minimum signal power that can still be heard against the background noise is taken to be 220,000 pW (1 pW = 10 -12 mW), which corresponds to a level of 36.5 dBm.

The average power P CP of the broadcast signal (measured at a point with zero relative level) depends on the averaging interval and is equal to 923 μW when averaged over an hour, 2230 μW per minute and 4500 μW per second. The maximum broadcast signal power is 8000 μW.

The dynamic range of D C broadcast signals is 25...35 dB for announcer speech, 40...50 dB for an instrumental ensemble, and up to 65 dB for a symphony orchestra.

Primary discrete signals are usually in the form of rectangular pulses of direct or alternating current, usually with two resolved states (binary or on-off).

The modulation rate is determined by the number of units (chips) transmitted per unit of time, and is measured in baud:

B = 1/τ u, (2.23)

where τ and is the duration of an elementary message.

The speed of information transmission is determined by the amount of information transmitted per unit of time and is measured in bits/s:

where M is the number of signal positions.

IN binary systems(M=2) each element carries 1 bit of information, therefore, according to (2.23) and (2.24):

C max =B, bit/s (2.25)

Security questions

1. Define the concepts “information”, “message”, “signal”.

2. How to determine the amount of information in a single message?

3. What types of signals are there?

4. How does a discrete signal differ from a continuous one?

5. How does the spectrum of a periodic signal differ from the spectrum of a non-periodic signal?

6. Define signal bandwidth.

7. Explain the essence of fax transmission of messages.

8. How is a TV image scanned?

9. What is the frame rate in a TV system?

10. Explain the principle of operation of the TV transmitting tube.

11. Explain the composition of a complete TV signal.

12. Give the concept of dynamic range?

13. List the main telecommunication signals. What frequency ranges do their spectra occupy?

Stations are divided into analog and digital based on the type of switching. Telephone communication, operating on the basis of converting speech (voice) into an analog electrical signal and transmitting it over a switched communication channel (analog telephony), for a long time was the only means of transmitting voice messages over a distance. The ability to sample (by time) and quantize (by level) the parameters of an analog electrical signal (amplitude, frequency or phase) made it possible to convert an analog signal into a digital (discrete) one, process it using software methods and transmit it over digital telecommunication networks.

To transmit an analog voice signal between two subscribers in the PSTN (public telephone network) network, a so-called standard voice frequency (VoF) channel is provided, the bandwidth of which is 3100 Hz. In a digital telephony system, the operations of sampling (in time), quantization (in level), encoding and eliminating redundancy (compression) are performed on an analog electrical signal, after which the data stream thus generated is sent to the receiving subscriber and upon “arrival” at the destination is subjected to reverse procedures.

The speech signal is converted using the appropriate protocol, depending on the network through which it is transmitted. Currently, the most efficient transmission of the flow of any discrete (digital) signals, including those carrying speech (voice), is provided by digital electrical networks that implement packet technologies: IP (Internet Protocol), ATM (Asynchronous Transfer Mode) or FR ( Frame Relay).

The concept of digital voice transmission is said to have originated in 1993 at the University of Illinois (USA). During the next flight of the Endeavor shuttle in April 1994, NASA transmitted its image and sound to Earth using computer program. The received signal was sent to the Internet, and anyone could hear the voices of the astronauts. In February 1995, the Israeli company VocalTec offered the first version of the Internet Phone program, designed for owners of multimedia PCs running Windows. Then it was created private network Internet Phone servers. And thousands of people have already downloaded the Internet Phone program from home page VocalTec and started communicating.

Naturally, other companies very quickly appreciated the prospects offered by the ability to talk while in different hemispheres and without paying for it international calls. Such prospects could not go unnoticed, and already in 1995, a flood of products designed for voice transmission over the Network hit the market.

Today, there are several standardized methods of transmitting information that are most widespread in the digital telephony services market: these are ISDN, VoIP, DECT, GSM and some others. Let's try to briefly talk about the features of each of them.

So what is ISDN?

The abbreviation ISDN stands for Integrated Services Digital Network - a digital network with integration of services. This is the modern generation of the worldwide telephone network, which has the ability to transfer any type of information, including fast and correct data transmission (including voice) of high quality from user to user.

The main advantage of the ISDN network is that you can connect several digital or analog devices (telephone, modem, fax, etc.) to one network end, and each can have its own landline number.

A regular telephone is connected to a telephone exchange using a pair of conductors. In this case, one pair can only have one telephone conversation. At the same time, noise, interference, radio, and extraneous voices can be heard in the handset - the disadvantages of analog telephone communication, which “collects” all the interference in its path. When using ISDN, a network termination is installed for the subscriber, and the sound, converted by a special decoder into a digital format, is transmitted through a specially designated (also completely digital) channel to the receiving subscriber, while ensuring maximum audibility without interference and distortion.

The basis of ISDN is a network built on the basis of digital telephone channels (also providing the possibility of packet-switched data transmission) with a data transfer rate of 64 kbit/s. ISDN services are based on two standards:

Basic access (Basic Rate Interface (BRI)) - two B-channels 64 kbps and one D-channel 16 kbps

Primary Rate Interface (PRI) - 30 B-channels 64 kbps and one D-channel 64 kbps

Typically, BRI bandwidth is 144 Kbps. When working with PRI, the entire digital communication backbone (DS1) is fully used, which gives a throughput of 2 Mbit/s. The high speeds offered by ISDN make it ideal for large number modern communications services, including high-speed data transfer, screen sharing, video conferencing, large file transfer for multimedia, desktop video telephony and Internet access.

Strictly speaking, ISDN technology is nothing more than one of the varieties of “computer telephony”, or, as it is also called CTI telephony (Computer Telephony Integration).

One of the reasons for the emergence of CTI solutions was the emergence of requirements to provide company employees with additional telephone services that were either not supported by the existing corporate telephone exchange, or the cost of purchasing and implementing a solution from the manufacturer of this exchange was not comparable with the convenience achieved.

The first signs of CTI service applications were systems of electronic secretaries (autoattended) and automatic interactive voice greetings(menu), corporate voicemail, answering machine and conversation recording systems. To add the service of a particular CTI application, a computer was connected to the company’s existing telephone exchange. It contained a specialized board (first on the ISA bus, then on the PCI bus), which was connected to the telephone exchange via a standard telephone interface. Software computer running under a specific operating system(MS Windows, Linux or Unix), interacted with the telephone exchange through the program interface (API) of a specialized board and thereby ensured the implementation of an additional corporate telephony service. Almost simultaneously with this, a software interface standard for computer-telephony integration was developed - TAPI (Telephony API)

For traditional telephone systems, CTI integration is carried out as follows: some specialized computer board is connected to the telephone exchange and transmits (translates) telephone signals, state telephone line and its changes to a “program” form: messages, events, variables, constants. The telephone component is transmitted via the telephone network, and the software component is transmitted via a data network or IP network.

What does the integration process in IP telephony look like?

First of all, it should be noted that with the advent of IP telephony, the very perception of a telephone exchange (Private Branch eXchange - PBX) has changed. IP PBX is nothing more than another network service of the IP network, and, like most IP network services, it operates in accordance with the principles of client-server technology, i.e. it assumes the presence of service and client parts. So, for example, an email service on an IP network has a service part - mail server and the client part - the user program (for example Microsoft Outlook). The IP telephony service is structured similarly: the service part - the IP PBX server and the client part - the IP telephone (hardware or software) use a single communication medium - the IP network - to transmit voice.

What does this give the user?

The advantages of IP telephony are obvious. Among them are rich functionality, the ability to significantly improve employee interaction and at the same time simplify system maintenance.

In addition, IP communications are evolving in an open manner due to protocol standardization and global IP penetration. Thanks to the principle of openness in the IP telephony system, it is possible to expand the services provided and integrate with existing and planned services.

IP telephony allows you to build a single centralized system management for all subsystems with differentiation of access rights and operate subsystems in regional divisions using local personnel.

The modularity of the IP communications system, its openness, integration and independence of components (unlike traditional telephony) provide additional features for building truly fault-tolerant systems, as well as systems with a distributed territorial structure.

Wireless communication systems of the DECT standard:

Standard wireless access DECT (Digital Enhanced Cordless Telecommunications) is the most popular system mobile communications V corporate network, the cheapest and easiest option to install. It allows you to organize wireless communication throughout the entire territory of the enterprise, which is so necessary for “mobile” users (for example, enterprise security or heads of workshops and departments).

The main advantage of DECT systems is that with the purchase of such a phone you get a mini-PBX for several internal numbers almost free of charge. The fact is that you can purchase additional handsets for the DECT base once purchased, each of which receives its own internal number. From any handset you can easily call other handsets connected to the same base, transfer incoming and internal calls, and even carry out a kind of “roaming” - register your handset on another base. The reception radius of this type of communication is 50 meters indoors and 300 meters outdoors.

To organize mobile communications in public networks, networks are used cellular communication GSM and CDMA standards, the territorial effectiveness of which is practically unlimited. These are the standards of the second and third generation of cellular communications, respectively. What are the differences?

Every minute, several phones located in its vicinity try to contact any base station of a cellular network. Therefore, stations must provide “multiple access”, that is, simultaneous operation of several telephones without mutual interference.

In first generation cellular systems (standards NMT, AMPS, N-AMPS, etc.), multiple access is implemented by the frequency method - FDMA (Frequency Division Multiple Access): the base station has several receivers and transmitters, each of which operates at its own frequency, and the radiotelephone tunes to any frequency used in the cellular system. Having contacted the base station on a special service channel, the phone receives an indication of which frequencies it can occupy and tunes to them. This is no different from the way a particular radio wave is tuned.

However, the number of channels that can be allocated at the base station is not very large, especially since neighboring cellular network stations must have different sets of frequencies so as not to create mutual interference. In the majority cellular networks the second generation began to use the time-frequency method of channel division - TDMA (Time Division Multiple Access). In such systems (and these are networks of GSM, D-AMPS, etc. standards) various frequencies are also used, but each such channel is allocated to the phone not for the entire communication time, but only for short periods of time. The remaining same intervals are alternately used by other phones. Useful information in such systems (including speech signals) are transmitted in “compressed” form and in digital form.

Sharing Each frequency channel with several telephones makes it possible to provide service to a larger number of subscribers, but there are still not enough frequencies. CDMA technology, built on the principle of code division of signals, was able to significantly improve this situation.

The essence of the code division method used in CDMA is that all phones and base stations simultaneously use the same (and at the same time the entire) frequency range allocated for the cellular network. In order for these broadband signals to be distinguished from each other, each of them has a specific code “coloring”, which ensures that it stands out from the others.

Over the past five years, CDMA technology has been tested, standardized, licensed and launched by most wireless equipment vendors and is already in use around the world. Unlike other methods of subscriber access to the network, where signal energy is concentrated on selected frequencies or time intervals, CDMA signals are distributed in a continuous time-frequency space. In fact, this method manipulates frequency, time, and energy.

The question arises: can CDMA systems, with such capabilities, “peacefully” coexist with AMPS/D-AMPS and GSM networks?

It turns out they can. Russian regulatory authorities have allowed the operation of CDMA networks in the radio frequency band 828 - 831 MHz (signal reception) and 873-876 MHz (signal transmission), where two CDMA radio channels with a width of 1.23 MHz are located. In turn, the GSM standard in Russia is allocated frequencies above 900 MHz, so the operating ranges of CDMA and GSM networks do not overlap in any way.

What I want to say in conclusion:

As practice shows, modern users are increasingly gravitating towards broadband services (video conferencing, high-speed data transfer) and increasingly prefer a mobile terminal to a regular wired one. If we also take into account the fact that the number of such people in large companies can easily exceed a thousand, we get a set of requirements that only a powerful modern digital exchange (UPBX) can satisfy.

Today, the market offers many solutions from various manufacturers that have the capabilities of both traditional PBXs, switches or routers for data networks (including ISDN and VoIP technologies), and the properties of wireless base stations.

Digital PBXs today, to a greater extent than other systems, meet the specified criteria: they have the capabilities of switching broadband channels, packet switching, and are simply integrated with computer systems(CTI) and allow the organization of wireless microcells within corporations (DECT).

Which one specified types better connections? Decide for yourself.

Ensuring the transmission of electrical communication signals in an effectively transmitted frequency band (ETF) of 0.3 - 3.4 kHz. In telephony and communications, the abbreviation KTC is often used. The audio channel is a unit of measurement for the capacitance (density) of analog transmission systems (eg K-24, K-60, K-120). At the same time, for digital transmission systems (for example, PCM-30, PCM-480, PCM-1920), the unit of measurement of capacitance is the main digital channel.

Efficiently transmitted frequency band- frequency band, the residual attenuation at extreme frequencies of which differs from the residual attenuation at a frequency of 800 Hz by no more than 1 Np at the maximum communication range characteristic of a given system.

The width of the EPCH determines the quality of telephone transmission, and the possibility of using the telephone channel to transmit other types of communications. In accordance with the international standard for telephone channels of multi-channel equipment, the frequency range is set from 300 to 3400 Hz. With such a band, a high degree of speech intelligibility is ensured, its sound is well natural, and great opportunities are created for secondary multiplexing of telephone channels.

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Subtitles

PM channel operating modes

Purpose of modes

2 PR. OK - for open telephone communication in the absence of transit extenders on the telephone switch;
2 PR. TR - for temporary transit connections of open telephone channels, as well as for terminal communication if there are transit extenders on the telephone switch;
4 PR OK - for use in networks of multi-channel voice-frequency telegraph, closed telephone communication, data transmission, etc., as well as for transit connections with significant lengths of connecting lines;
4 PR TR - for long-term transit connections.

Remote meetings with poor quality audio are often annoying. Misunderstandings become more likely because important nuances and other subtleties are difficult to hear in a conversation. Therefore, it is necessary to strive to improve the sound quality during teleconferences. Below is a brief description of the various audio quality specifications.

Mobile solutions give you more flexibility and mobility, but sometimes the audio quality suffers. Many mobile operators now offer HD Voice technology on their networks, which provides HD audio if the phone supports it.
Traditional analog telephony provides acceptable sound quality, but with limitations in the frequency range. Sometimes this sound is called telephone or narrowband.
VoIP, i.e. digital telephony over a data network (Voice over IP), allows the use of an expanded frequency range, but with some compression. IP allows for superior audio quality, also called HD audio or wideband audio.
Remember that everything local networks and equipment such as Wi-Fi, DECT (cordless telephony) or Bluetooth® affect throughput and may have negative impact on sound quality.
All Konftel conference phones support HD audio.

Sound and its perception

Humans are able to perceive sounds between 20 and 20,000 Hz (20 Hz - 20 kHz). This range changes as a person ages and due to physical factors. An adult usually distinguishes sounds at frequencies in the range between 20 and 12 kHz.

Previously, the concept of “telephone quality” was used - an interval in which the frequency range, due to technical shortcomings, was limited between 200 Hz and 3.4 kHz. Today this is called narrowband communication. For analogue telephony, this means the loss of a significant part of the speech frequency range. This makes speech less natural and harder to understand than if the frequency range were larger. Compare this to FM radio, which has a frequency range of up to 15 kHz, allowing both voices and music to be heard much more naturally.

Analog telephony

Analogue telephony has an extremely limited frequency response (about 3.2 kHz). An analog signal is perceived by some as more natural, although a digital signal generally has a wider frequency range. This is because the human ear is very good at picking up artificial sound.

Data throughput and frequency range

The term "bandwidth" refers to the amount of information per second that is transmitted over a network. And the concept of “frequency range” refers to audio frequencies. Hertz (Hz) is the unit for both, so unfortunately this sometimes leads to misunderstandings since frequency range and data throughput are not the same thing. Moreover, throughput can be expressed as both Hertz and bits per second (you'll usually see the designation Mbps on the web). The sound is converted into digital networks. The audio signal is measured thousands of times per second and converted into a digital signal.

Mobile telephony

Depending on how much data the mobile networks of different operators have, the audio signal is always more or less limited in range to save bandwidth. Sound in 2G networks allows narrowband transmission (3.2 kHz), while 3G and 4G networks allow wideband transmission (7 kHz). More recently, a number of operators have begun to use broadband standards and launched so-called HD Voice technology. However, for this technology to work, the phone must also support this standard. Poor transmission and reception conditions can also affect sound quality. In this case, the system automatically reduces the network transmission speed. This has a negative impact on sound quality.

VOIP, wideband audio and codec

Telephony over a data network is called VoIP (Voice over IP). The sound in digital networks was initially of the same quality as in the old analogue technology, i.e. The audio bandwidth was 3.2 kHz (narrowband). This was necessary in early digital networks because data throughput was clearly limited.

On digital networks, audio quality is limited primarily by the codec that has been selected. A codec is a piece of software in a phone that compresses outgoing analog audio into data packets and converts incoming data packets into analog audio. Thus, modern phones that support wideband codecs are able to provide best sound. The last 10-15 years have seen fantastic advancements in VoIP.

Common designations for different codecs are: wideband codec (7 kHz), super wideband codec (14 kHz), and full bandwidth codec (20 kHz). There is also a wide range technical solutions and standards: G.718, G.722.2, G.729.1, etc.

Wireless solutions

Of course, the bandwidth of an office's broadband and/or mobile network is determined by how good the sound can be. It is also important to consider the internal structure of the office, since anything installed outside the telephone network can reduce audio throughput. These may be wireless systems such as DECT and Bluetooth®, or older networking products.

Bluetooth®

Bluetooth® is a standard that was originally developed to allow a variety of accessories to connect via wireless network To mobile phone or computer. Bluetooth® only works over short distances between the mobile phone and accessories. There is additional compression of the audio signal data, which may negatively affect the sound quality. The trend is increasingly spreading to modern technology Bluetooth® supporting HD audio.

DECT and CAT-IQ

DECT solutions for wireless telephony in offices and industrial premises were originally developed for use with analogue telephony. In a DECT network it is not possible to obtain sound quality better than standard telephone quality (3.2 kHz). This is unlikely to matter for regular phone calls, but if you want to host meetings where audio quality is particularly important, using direct connections (cables) to the VoIP network may be a good idea.

Simply put, CAT-iq is a digital optimization of DECT. The CAT-iq system has wideband codecs and thus allows the use of 7 kHz audio bandwidth.

Konftel solutions

Konftel products always provide optimal sound quality. If the network distributes HD audio, you will receive HD audio on Konftel conference phones.

This shows that there is reason to analyze the communications needs of your business and organization before selecting a network and upgrading your telephony and data infrastructure. For example, a VoIP network with wideband codecs (7 kHz) is better equipped to provide superior audio than an analogue or older mobile network. This may be obvious, but on the other hand, portability and simplicity can be a key factor in certain contexts.

Many Konftel products offer more than one connection option. HD Voice technology can give you both optimal sound quality and portability.

The wireless Konftel 300Wx is one example of how flexible our products are. Thanks to the analogue DECT connection, it can transmit a 3.2 kHz bandwidth, while the USB connection for the computer can use wideband codecs (7 kHz). You can also connect it to your mobile phone using a cable.

The same device also provides wireless HD audio (broadband) in IP telephony when a Konftel DECT 10 base station is connected via SIP. It can have up to 5 registered Konftel 300Wx calls. It is possible to configure the Konftel 300Wx with IP DECT base stations provided by third-party manufacturers supported by Konftel. However, Konftel IP DECT 10 offers unique benefits and simplifies your work.

Whatever your needs, Konftel has products that make teleconferencing at your desk and large meetings in conference rooms easier and faster.

KHOREV Anatoly Anatolyevich, Doctor of Technical Sciences, Professor

TECHNICAL CHANNELS FOR LEAKAGE OF INFORMATION TRANSMITTED THROUGH COMMUNICATION CHANNELS

Technical channels for leakage of information transmitted via wired communication channels

Until now, telephone communication prevails among many types of electrical and radio communications, so the telephone channel is the main one on the basis of which narrowband and broadband channels for other types of communication are built.

On the transmitting side of the telephone channel, a microphone is used as a transmitter, which converts acoustic signals in the frequency band DF = 0.3 ... 3.4 kHz into electrical signals of the same frequencies. On the receiving side, the telephone channel ends with a telephone capsule (telephone), which converts electrical energy into acoustic signals in the frequency band DF = 0.3 ... 3.4 kHz.

Analog and discrete (digital) channels are used to transmit information.

An analog channel is more often called a voice-frequency channel (TV channel). It is used to transmit speech, email, data, telegraphy, fax, etc. The capacity of the PM channel is C x = 25 kbit/s.

A standard digital channel (SDC) with a capacity of C x = 64 kbit/s is designed primarily for real-time speech transmission, i.e. for ordinary telephony for the purpose of transmitting signals of frequencies 0.3 - 3.4 kHz.

In order to convert the frequency band 0.3 - 3.4 kHz (analog signal - speech) into a digital stream at a speed of 64 kbit/s, three operations are performed: sampling, quantization and encoding.

In modern multi-channel equipment, it is possible to create channels with a higher throughput than those of the TC and SSC channels. An increase in throughput is achieved by expanding the effectively transmitted frequency band. All channels use the same transmission line, so the end equipment must perform channel separation.

Among possible methods Two types of channel divisions have become predominant: frequency and time. With the frequency method, each channel is allocated a certain section of the frequency range within the bandwidth of the communication line. Distinctive features of channels are the frequency bands they occupy within the overall bandwidth of the communication line. In the time division method, channels are connected to the communication line one by one, so that each channel is allocated a certain time interval during the total transmission time of the group signal. A distinctive feature of the channel in this case is the time it connects to the communication line.

Modern multichannel equipment is built on a group principle. When constructing terminal equipment, as a rule, multiple frequency conversion is used. The essence of multiple frequency conversion lies in the fact that in the transmitting part of the equipment the spectrum of each primary signal is converted several times before taking its place in the linear spectrum. The same multiple conversion, but in the reverse order, is carried out in the receiving part of the equipment.

Most types of multi-channel equipment are designed for a number of channels that are a multiple of twelve, and are completed from the corresponding number of standard 12-channel primary groups (PG). When forming the primary group, the spectrum of each of the twelve primary signals occupying the bands 0.3 - 3.4 kHz is transferred to the band 60 - 108 kHz using the corresponding carrier frequencies. The 12-channel group equipment is individual equipment for most types of multi-channel equipment. The total frequency band 60 - 108 kHz is fed further to the group transmission equipment.

Subsequent conversion stages are designed to create larger groups of channels: 60-channel (secondary) group (SG), 300-channel (tertiary) group (TG), etc. The frequency bands 60 - 108 kHz of each of the five primary groups are moved using group frequency converters to the band of the 60-channel group corresponding to this group. Bandpass filters form a common VG frequency band of 312 - 552 kHz.

By analogy with VG, a 300-channel group scheme is constructed, occupying a band from 812 to 2044 kHz.

The basic data of multi-channel equipment with frequency division of channels are given in table. 1.

The use of certain means to intercept information transmitted over telephone communication lines will be determined by the ability to access the communication line (Fig. 1).

To intercept information from various types of cables, they are used different types devices:

for symmetrical high-frequency cables - devices with induction sensors;
for coaxial high-frequency cables - direct (galvanic) connection devices;
for low-frequency cables - devices for direct (galvanic) connection, as well as devices with induction sensors connected to one of the wires.

For example, to “collect” information from underwater armored cable communication lines in the 80s of the last century, a technical reconnaissance device of the “Kambala” type was used. It's quite complicated electronic device with a nuclear (plutonium) power source designed for decades of operation.

It was made in the form of a steel cylinder 5 m long and 1.2 m in diameter. Several tons of electronic equipment were installed in a hermetically sealed pipe to receive, amplify and demodulate signals taken from the cable. The intercepted conversations were recorded by 60 automatically operating tape recorders, which turned on when there was a signal and turned off when there was no signal. Each tape recorder was designed for 150 hours of recording. And the total volume of recordings of intercepted conversations could be about three thousand hours.

Table 1. Basic data of multi-channel frequency division equipment

Equipment type, cable/line	Linear frequency band, kHz	Two-way communication system used	Average length of the reinforcement section, km	Basics appointment
K-3600, coaxial	812 - 17600		3	Trunk connection
K-1920P, coaxial	312 - 8500	Single-way four-wire, single-cable	6	Trunk connection
K-300, coaxial; K-300R, coaxial	60 - 1300	Single-way four-wire, single-cable	6	Intrazone or trunk communication
K-1020R, coaxial;	312 - 6400	Single-way four-wire, single-cable	3	Distribution system (intraarea communication)
K-120, coaxial	60 - 552,		10	Intrazone communication
K-1020R, symmetrical	312 - 4636		3,2	Trunk connection
K-60P, symmetrical	12 - 252	Single-band four-wire, two-cable	10	Intrazone communication.
KRR-M, KAMA, symmetrical	12 - 248 312 - 548	Two-way two-wire, single-cable	13 2 – 7	Local communications, connecting lines between telephone exchanges
V-12-3, overhead line with non-ferrous metal wires	36 - 84 92 - 143	Two-way two-wire.	54	Rural connection

Rice. 1. Diagram of a telephone information transmission channel

By the time the film was used up, the underwater swimmer found the device using a hydroacoustic beacon installed on the container, removed the induction sensor and pre-amplifier from the cable and delivered the device to a specially equipped submarine, where the tape recorders were replaced, after which the device was again installed on the communication line.

The device's special sensitive induction sensors were capable of reading information from an underwater cable protected not only by insulation, but also by double armor made of steel tape and steel wire tightly wrapped around the cable. Signals from the sensors were amplified by preliminary antenna amplifier, and then sent for demodulation, isolating individual conversations and recording them on a tape recorder. The system provided the ability to simultaneously record 60 conversations conducted over a cable communication line.

To intercept information from cable communication lines passing overland, American specialists developed the “Mole” device more than 20 years ago. It used the same principle as the “Kambala” device. Information from the cable was taken using a special sensor. To install it, wells were used through which the cable passes. The sensor in the well is mounted on a cable and, to make detection difficult, is pushed into the pipe leading the cable to the well. The information intercepted by the sensor was recorded on a magnetic disk of a special tape recorder. Once full, the disk is replaced with a new one. The device made it possible to record information transmitted simultaneously through 60 telephone channels. The duration of continuous recording of the conversation on a tape recorder was 115 hours.

Demodulation of intercepted conversations was carried out using special equipment in stationary conditions.

In order to simplify the task of finding a “Mole” device to replace disks, they were equipped with a radio beacon mounted in the device’s body. The agent, driving or passing in the area where the device was installed, asked him using his portable transmitter if everything was normal. If no one touched the device, the radio beacon transmitted the corresponding signal. In this case, the tape recorder disk was replaced.

One of the “Mole” devices was discovered on a cable communication line running along the highway approaching Moscow. More than ten similar devices, at the request of the Syrian side, were removed by Soviet specialists in Syria. All of them were camouflaged as local objects and mined to make them indestructible.

Interception of information from ordinary subscriber two-wire telephone lines can be carried out either by direct contact connection to the lines, or using simple small-sized inductive sensors connected to one of the wires of the subscriber line.

The fact of contact connection to the communication line is easy to detect. When connecting an induction sensor, the integrity of the cable braid is not damaged, the cable parameters do not change, and in this case it is almost impossible to detect the fact of connection to the line.

Information intercepted from a telephone line may be recorded on a tape recorder or transmitted over the air using microtransmitters, which are often called telephone bookmarks or telephone repeaters.

Phone bookmarks can be classified by type of design, installation location, power source, method of transmitting information and encoding it, control method, etc. (Fig. 2).

As a rule, they are made either in the form of a separate module, or are camouflaged as elements of a telephone set, for example, a capacitor, telephone or microphone capsules, telephone plug, socket, etc.

Phone bookmarks in the usual design have small sizes (volume from 1 cm 3 to 6 - 10 cm 3) and weight from 10 to 70 g. For example, the phone bookmark HKG-3122 has dimensions of 33x20x12 mm, and SIM-A64 - 8x6x20 mm.

Rice. 2. Classification of phone bookmarks

Phone bookmarks transmit intercepted information, as a rule, via a radio channel. Typically a telephone wire is used as an antenna.

To transmit information, the most commonly used are VHF (meter), UHF (decimeter) and GHz (GHz) wavelength ranges, wideband frequency (WFM) or narrowband (NFM) frequency modulation.

To increase secrecy, digital signals with phase or frequency shift keying are used, transmitted information can be encoded various methods.

The information transmission range at a radiation power of 10 - 20 mW, depending on the type of modulation and the type of receiver used, can range from 200 to 600 m.

The transmission of information (radiation work) begins the moment the subscriber picks up the handset. However, there are bookmarks that record information into a digital storage device and transmit it upon command.

Telephone bookmarks can be installed: in the telephone body, handset or telephone socket, as well as directly in the telephone line path.

The ability to install a telephone bookmark directly into the telephone line is important, since to intercept a telephone conversation there is no need to enter the room where one of the subscribers is located. Telephone bookmarks can be installed either in the telephone line path to the distribution box, located, as a rule, on the same floor as the room where the controlled device is installed, or in the telephone line path from the distribution box to the distribution panel of the building, usually located on the ground floor or in basement of the building.

Telephone bookmarks can be installed in series in the break of one of the telephone wires, in parallel or through an inductive sensor.

When switched on in series, the bookmark is powered from the telephone line, which ensures unlimited operating time. However, a serial connection is quite easy to detect by changing the line parameters and, in particular, the voltage drop. In some cases, a serial connection with voltage drop compensation is used, but the implementation of this requires an additional power source.

Telephone bookmarks with parallel connection to the line can be powered either from the telephone line or from autonomous power sources. The higher the input resistance of the bookmark, the more insignificant the change in the line parameters and the more difficult it is to detect. It is especially difficult to detect a plug connected to the line through a high-resistance adapter with a resistance of more than 18 - 20 MOhm. However, such a bookmark must have autonomous power supply.

Along with a contact connection, non-contact information retrieval from a telephone line is also possible. For these purposes, bookmarks with miniature induction sensors are used. Such bookmarks are powered from autonomous power sources and the fact of connecting them to the line can be established even by the most modern means almost impossible, since the line parameters do not change when connected.

When powered from a telephone line, the operating time of the bookmark is not limited. When using autonomous power sources, the operating time of the bookmark ranges from several tens of hours to several weeks. For example, the 4300-TTX-MR telephone radio bookmark, installed in a handset, with a radiation power of 15 mW and using a PX28L battery, provides operating time from 3 to 12 weeks.

Methods of using telephone bookmarks are determined by the ability to access the room where the controlled telephone is installed.

If it is possible to enter the premises even for a short time, the bookmark can be installed in the telephone body, handset, etc. Moreover, this requires from 10 - 15 seconds to several minutes. For example, replacing a regular microphone capsule with a similar one, but with a telephone bookmark installed in it, takes no more than 10 seconds. Moreover, it is impossible to distinguish them visually.

Phone bookmarks, made in the form of separate elements of a telephone circuit, are soldered into the circuit instead of similar elements or are disguised among them. The most commonly used bookmarks are made in the form of various types of capacitors. Installation of such devices takes several minutes and is usually carried out during troubleshooting or preventive maintenance of a telephone set.

It is possible that a bookmark can be installed on a telephone before it arrives at an institution or enterprise.

If access to the controlled premises is impossible, bookmarks are installed either directly in the telephone line path, or in distribution boxes and panels, usually in such a way that their visual detection is difficult.

The smaller the bookmark, the easier it is to disguise it. However, small-sized bookmarks in some cases do not provide the required information transmission range. Therefore, to increase the range of information transmission, special repeaters are used, installed, as a rule, in hard-to-reach places or in a car within the range of the bomb.

To intercept fax transmissions, special complexes such as 4600-FAX-INT, 4605-FAX-INT, etc. are used. .

A typical system for intercepting fax transmissions is located in a standard briefcase, can be powered either from AC power or from built-in batteries, is connected to the line via a high-resistance adapter, so it is almost impossible to determine the fact of connection, allows you to automatically recognize voice and fax messages, record transmitted messages, has high noise immunity and adapts to changes in line parameters and information transmission speed. The system allows you to continuously monitor the reception and transmission of several faxes.

Registration of intercepted messages can be carried out in several forms:

line-by-line registration in real time;
line-by-line printing with simultaneous recording to a storage device;
printing recorded information to output devices;
recording information into a storage device without printing.

In addition to recording intercepted messages, such a system records service information about the nature of transmitted messages, non-standard fax operating modes, searches and cryptography methods (techniques).

The system software allows you to simulate a fax machine receiver with advanced capabilities for visual analysis of recorded signals and setting demodulation parameters in cases where automatic demodulation is unsatisfactory.

Technical channels for leaking information transmitted via radio communication channels

One of the most common methods of transmitting large amounts of information over long distances is multi-channel radio communication using radio relay lines and space communication systems. Radio relay communication is communication using intermediate amplifiers-repeaters. The routes of multichannel radio relay lines, as a rule, are laid near highways to facilitate servicing of remote repeaters, which are located at dominant heights, masts, etc. IN space systems Communication information is transmitted through relay satellites located in geostationary and high elliptical orbits.

The global strategy for the modern development of radio communications is the creation of international and global public radio networks based on the widespread use of mobile (mobile) radio communications.

The dominant position in the mobile radio market today is occupied by:

departmental (local, autonomous) systems with communication channels strictly assigned to subscribers;
trunking radio communication systems with free access for subscribers to a common frequency resource;
cellular mobile radiotelephone communication systems with spatially separated frequency reuse;
personal radio call systems (PRC) - paging;
systems cordless phones(Cordless Telephony).

Fixed-channel communication systems have been used by government and commercial organizations, law enforcement, emergency services and other services for a long time. They can use both simplex and duplex communication channels, analog and digital ways masking messages, have a high efficiency of establishing communication.

The main frequency ranges for networks with assigned channels: 100 - 200, 340 - 375, 400 - 520 MHz.

The use of public mobile radio communication networks (trunking, cellular) is currently recognized as the most optimal, since they provide subscribers with a greater variety of services (from the formation of dispatch communications for individual services to automatic access to subscribers of city and long-distance telephone networks), and also allow for a sharp increase in network bandwidth. In these networks, any subscriber has the right to access any unoccupied network channel and is subject only to queuing discipline.

The term “trunking” is understood as a method of equal access of network subscribers to a common dedicated channel bundle, in which a specific channel is assigned individually for each communication session. Depending on the load distribution in the system, communication between individual subscribers in such a network is carried out mainly through a special transceiver base station. The range of a base station in urban conditions, depending on the frequency range of the network, the location and power of the base and subscriber stations, ranges from 8 to 50 km.

The most widely used trunked radio communication systems are presented in Table. 2.

The main consumers of trunking communication services are law enforcement agencies, emergency call services, armed forces, security services of private companies, customs, municipal authorities, security and escort services, banks and collection services, airports, energy substations, construction companies, hospitals, forestries, transport companies, railways, industrial enterprises.

Cellular radiotelephone communications occupy a special place among public communication networks. The cellular principle of network topology with frequency reuse has largely solved the problem of frequency resource shortage and is currently the main one in the created public mobile communication systems.

Table 2. Characteristics of trunked radio communication systems

System (standard)	Name of characteristics
System (standard)	Frequency bands, MHz	Channel bandwidth, kHz, (channel spacing)	Number of channels (including control channels)	Note
Altai	337 - 341 301- 305	25	180	Analog
Smartrunk	146 - 174 403 - 470	150/250	16	Single zone Analog
MRI 1327	146 - 174 300 - 380 400 - 520	12,5/25	24	Multi-zone Analog Digital control
EDACS	30 - 300 800-900	25/30 12,5	20	Analogue (speech) FM Digital (speech, data)
TETRA	380 - 400	25	200	Digital (TDMA) p/4 DQPSK

The structure of cellular networks is a collection of small service areas adjacent to each other and having different communication frequencies, which can cover vast territories. Since the radius of one such zone (cell, cell) does not, as a rule, exceed several kilometers, in cells that are not directly adjacent to each other, it is possible to reuse the same frequencies without mutual interference.

Each cell houses a stationary (base) transceiver radio station, which is connected by wire to the central station of the network. The number of frequency channels in the network usually does not exceed 7 - 10, and one of them is organizational. The transition of subscribers from one zone to another does not involve any changes in equipment. When a subscriber crosses the zone boundary, he is automatically given another free frequency belonging to the new cell.

Basic technical specifications cellular communication systems are presented in table. 3.

Table 3. Main technical characteristics of cellular communication systems

System (standard)	Name of characteristics
System (standard)	Frequency bands, MHz	Channel bandwidth, kHz	Maximum power, W	Number of channels	Signal class, modulation type
NMT-450	453 – 457.5 (PS) 463 – 467.5 (BS)	25	50 (BS) 15 (PS)	180	16KOF3EJN
AMPS	825 – 845 (PS) 870 – 890 (BS)	30	45(BS) 12 (PS)	666	30KOF3E
D-AMPS	825 – 845 (PS) 870 – 890 (BS)	30	-	832	30KOG7WDT p/4 DQPSK
GSM	890 – 915 (PS) 935 – 960 (BS)	200	300 (BS)	124	200KF7W GMSK
DCS-1800	1710 – 1785 (PS) 1805 –1880 (BS)	200	<1 Вт (ПС)	374	200KF7W GMSK
IS-95	825 – 850 (PS) 870 – 894 (BS)	1250	50 (BS) 6 (PS)	55 per carrier	1M25B1W QPSK (BS), OQPSK(PS)

Note: MS – mobile station, BS – base station.

The NMT-450 and GSM standards are adopted as federal standards, and AMPS/D-AMPS is aimed at regional use. The DCS-1800 standard is promising.

The NMT-450 standard uses a duplex frequency spacing of 10 MHz. Using a frequency grid of 25 kHz, the system provides 180 communication channels. Cell radius is 15 - 40 km.

All service signals in the NMT system are digital and are transmitted at 1200/1800 bps FFSK (Fast Frequency Shift Keying).

Cellular systems based on the NMT standard are used in Moscow, St. Petersburg and other regions of the country.

The AMPS cellular communication system operates in the range 825 - 890 MHz and has 666 duplex channels with a channel width of 30 kHz. The system uses antennas with a radiation pattern width of 120°, installed in the corners of the cells. Cell radii 2 - 13 km.

In Russia, systems according to the AMPS standard are installed in more than 40 cities (Arkhangelsk, Astrakhan, Vladivostok, Vladimir, Voronezh, Murmansk, Nizhny Novgorod, etc.). However, experts believe that in large cities AMPS will gradually be replaced by digital standards. For example, in Moscow, in the ranges above 450 MHz, only digital standards are now used.

The D-AMPS digital system using TDMA multiple access technology is currently the most widespread digital cellular system in the world. The digital standard has a frequency channel width of 30 kHz. The D-AMPS standard has been adopted as a regional standard. Systems have been created according to this standard in Moscow, Omsk, Irkutsk, and Orenburg.

The GSM standard is closely related to all modern digital network standards, primarily ISDN (Integrated Services Digital Network) and IN (Intelligent Network).

The GSM standard uses narrowband Time Division Multiple Access (TDMA). The TDMA frame structure contains 8 time positions on each of the 124 carriers.

To protect against errors in radio channels when transmitting information messages, block and convolutional coding with interleaving is used. Increasing the efficiency of coding and interleaving at low speeds of movement of mobile stations is achieved by slow switching of operating frequencies (SFH) during a communication session at a rate of 217 hops per second.

To combat interference fading of received signals caused by multipath propagation of radio waves in urban conditions, communication equipment uses equalizers that ensure equalization of pulse signals with a standard deviation of the delay time of up to 16 μs. The synchronization system is designed to compensate for the absolute signal delay time of up to 233 μs, which corresponds to a maximum communication range or maximum cell radius of 35 km.

The GSM standard selects Gaussian minimum shift keying (GMSK) with a normalized bandwidth of 0.3. Frequency Shift Keying Index - 0.5. With these parameters, the radiation level in the adjacent channel will not exceed -60 dB.

Speech processing is carried out within the framework of the adopted system of discontinuous transmission of speech (DTX), which ensures that the transmitter is turned on only when a speech signal is present and the transmitter is turned off during pauses and at the end of a conversation. A speech codec with regular pulse excitation/long-term prediction and linear predicative coding with prediction (RPE/LTP-LPC codec) was selected as a speech converting device. The overall speech signal conversion speed is 13 kbit/s.

The GSM standard achieves a high degree of security for message transmission; messages are encrypted using the public key encryption algorithm (RSA).

The DCS-1800 system operates in the 1800 MHz band. The core of the DCS-1800 standard consists of more than 60 GSM standard specifications. The standard is designed for cells with a radius of about 0.5 km in dense urban areas and up to 8 km in rural areas.

The IS-95 standard is a cellular communication system standard based on CDMA Code Division Multiple Access. Security of information transmission is a property of CDMA technology, so operators of these networks do not require special message encryption equipment. The CDMA system is built using the direct frequency spread method based on the use of 64 types of sequences formed according to the law of Walsh functions.

The standard uses separate processing of reflected signals arriving with different delays and their subsequent weighted summation, which significantly reduces the negative impact of the multipath phenomenon.

The IS-95 CDMA system in the 800 MHz band is the only operational cellular communication system with code division technology. It is planned to use its version for the 1900 MHz band.

Personal radio calling (paging) provides wireless one-way transmission of alphanumeric or audio information of a limited volume within the service area. The frequency range of paging systems is from 80 to 930 MHz.

Currently, in our country, the most widely used protocols for use in personal calling systems (paging systems) are POCSAG (Post Office Standardization Advisory Group), ERMES (European Radio Message System) and FLEX (Table 4). All these protocols are analog-to-digital. The main class of signals used is 16KOF1D.

Table 4. Main characteristics of paging systems

When transmitting POCSAG messages, two-level frequency modulation is used with a maximum frequency deviation of 4.5 kHz.

The FLEX protocol is characterized by high data transfer speed and, therefore, high throughput. At 1600 bps, two-level frequency modulation (FM) is used, at 6400 bps, four-level FM is used. The frequency deviation value in both cases is 4.8 kHz.

For the operation of paging systems using the ERMES protocol, a single frequency range (or part of it) 169.4 - 169.8 MHz is allocated, in which 16 operating channels are organized with a frequency spacing of 25 kHz. The data transfer rate is 6.25 kbit/s.

Cordless telephone systems (WPT) at the initial stage of their development were intended mainly to replace the handset cord with a wireless radio line in order to provide greater mobility to the subscriber. Further development of this type of communication, especially the transition to digital methods of information processing, significantly expanded the scope of application of BPT.

In analog-type BPT systems, most often used in residential premises and small institutions, personal-use BPTs are used, consisting of a base station (BS) connected to the city telephone network and a portable radiotelephone (PTA). When using BPT in large companies as an internal means of communication, branched networks of low-power radiotelephones are organized, the operating principle of which is similar to cellular networks. These systems mainly use digital signal processing methods to provide stronger encryption of transmitted messages.

Both analog and digital cordless phones operate in full-duplex mode on multiple channels, with channel selection performed automatically from unused channels. The range of certified radio transmitters (radiation power does not exceed 10 mW) of the BPT, depending on the type of equipment and operating conditions, is 25 - 200 m.

The power of uncertified BPT transmitters can be 0.35 - 1.2 W or more, while their range can range from several kilometers to several tens of kilometers.

List of frequency bands allocated for BPT subject to a maximum output power limitation of 10 mW and on a secondary basis, i.e. without any guarantee of ether purity are presented in Table 5.

Table 5. List of frequency bands allocated for wireless phones with power up to 10 mW

Standard	Frequency range, MHz
CT-0R	30 – 31/39 – 40
CT-1R	814 – 815/904 – 905
CT-2R	864 – 868,2
DECT	1880 – 1900

In fact, analog BPTs in Russia operate in the following main frequency ranges:

26.3125 - 26.4875 MHz/41.3125 - 41.4875 MHz;
30.075 - 30.300 MHz/39.775 - 40.000 MHz;
31.0125 - 31.3375 MHz/39.9125 - 40.2375 MHz;
31.025 - 31.250 MHz/39.925 - 40.150 MHz;
31.0375 - 31.2375 MHz/39.9375 - 40.1375 MHz;
31.075 - 30.300 MHz/39.775 - 39.975 MHz;
30.175 - 30.275 MHz/39.875 - 39.975 MHz;
30.175 - 30.300 MHz/39.875 - 40.000 MHz;
307.5 - 308.0 MHz/343.5 - 344.0 MHz;
46.610 - 46.930 MHz/49.670 - 49.990 MHz;
254 MHz/380 MHz; 263 – 267 MHz/393 – 397 MHz;
264 MHz/390 MHz; 268 MHz/394 MHz;
307.5 – 308.0 MHz/343.5 – 344.0 MHz;
380 – 400 MHz/250 – 270 MHz;
814 – 815 MHz/904 – 905 MHz;
885.0125 - 886.9875 MHz/930.0125 - 931.9875 MHz;
902 – 928 MHz/902 – 928 MHz;
959.0125 - 959.9875 MHz/914.0125 - 914.9875 MHz.

Digital BPTs use the following main frequency ranges: 804 - 868 MHz; 866 - 962 MHz; 1880 - 1990 MHz.

To intercept information transmitted using radio relay and space communication systems, radio reconnaissance means are used, and to intercept conversations conducted using cellular phones, special complexes for intercepting cellular communication systems are used.

Modern interception systems for cellular communication systems can provide (depending on the configuration) monitoring of control (calling) channels of up to 21 cells simultaneously, and allow monitoring and recording telephone conversations of 10 or more selected subscribers.

The complexes are produced in three types: “pocket” (in the form of a cell phone), mobile (in the form of a compact unit, a PC “Notebook” type and an antenna) and stationary (in the form of a desktop unit).

In addition to registering controlled conversations, the complexes can be equipped (depending on the standard) with some additional functions: monitoring conversations on a given number, “scanning” phones and intercepting incoming communications from a controlled subscriber.

For the “pocket” option, it is possible to control the conversations of one subscriber within the cell coverage area; for mobile - simultaneous monitoring and recording of conversations of one (several) subscribers in the coverage area of several cells and it is possible to maintain a database of monitored cells; for the stationary option - it is possible to simultaneously monitor and record conversations of more than ten subscribers throughout the entire cellular network and maintain an expanded database.

The phone “scanning” function is used to secretly determine the phone number and service parameters of a phone.

If you use the function of intercepting incoming communications of a controlled phone, it is possible to intercept all incoming calls from a specified subscriber.

Main functions of the complex:

decoding the service channel to identify the mobile phone number on which the conversation is taking place;
listening directly to a telephone conversation;
the ability to simultaneously control the frequency of the base station and the frequency of the mobile handset, that is, ensuring stable audibility of both interlocutors;
the ability to simultaneously control both incoming and outgoing calls;
monitoring frequency changes and conversation support when a subscriber moves from cell to cell;
control of several cells from one point;
recording telephone conversations using sound recording equipment in automatic mode;
recording on the hard drive of mobile phone numbers that carried out conversations throughout the entire cellular communication system, indicating the date and time.

During the operation of the complex, the monitor displays:

numbers of all telephones called on all cells of the system;
phone numbers that communicated in the cell to which the control channel is configured, as well as service information.

Hardware and software systems are also used to intercept paging messages. The standard complex includes:

modified scanning receiver;
PC with an input signal conversion device;
software.

The complex allows you to solve the following main tasks:

receive and decode text and digital messages transmitted in radio paging communication systems, save all received messages on the hard drive in an archive file;
filter the general flow of messages, select data addressed to one or a number of specific subscribers using a priori known or experimentally determined cap codes, promptly change the parameters of the list of observed subscribers;
carry out Russification of the entire input stream of messages or those addressed only to specific subscribers included in the list of monitored ones;
process the output data files in any text editor with the implementation of the standard search function for the entered string of characters and printing the necessary data on the printer.

While the program is running, the following is displayed on the monitor screen:

messages received via one of the active channels (the number of the displayed channel is entered by the operator from the keyboard without interrupting the program);
current time of day and date;
time and date of receipt of each selected message, its serial number, as well as the identifier of the corresponding selection attribute.

To decode intercepted messages hidden by encryption equipment, special devices are used (for example, 640-SCRD-INT). Such devices decode and restore with high quality in real time conversations closed by ZAS equipment.

Radio reconnaissance equipment and special systems for intercepting cellular communication systems are in service with special services of leading foreign countries and provide interception and decoding of messages transmitted using any communication systems, including the GSM standard.

To intercept telephone conversations conducted using analog UPTs, as well as cellular communication systems using analog signals, conventional scanning receivers can be used; the characteristics of some of them are given in Table. 6.

Table 6. Characteristics of scanning receivers

Name of characteristics	Index (type)
Name of characteristics	AR-5000	EB-200 “Miniport”	AR-8200 MK3
Manufacturer	A.O.R	ROHDE & SCHWARZ	A.O.R
Frequency range, MHz	0,01 – 3000	0,01 – 3000	0,10 – 3000
Types of modulation	AM, FM, LSB, USB, CW	AM, FM, LSB, USB, CW, Pulse	AM, FM, LSB, USB, CW
Sensitivity at signal-to-noise ratio 10 dB, µV	AM: 0.36 – 0.56 FM: 0.2 – 1.25 SSB: 0.14 – 0.25	AM: 1.0 – 1.5 FM: 0.3 – 0.5	AM: 0.70 – 2.50 FM: 0.35 – 2.50 SSB: 0.30 – 1.50
Selectivity at -6 dB, kHz	3; 6; 15; 40; 110; 220	0,15; 0,3; 0,6; 1,5; 2,5; 6; 9; 15; 30; 120; 150	SSB/NAM: 3 kHz AM/SFM: 9 kHz NFM: 12 kHz WFM: 150 kHz
Frequency tuning step, kHz	1 Hz to 1 MHz	10 Hz to 10 kHz
Number of memory channels	100 in 10 jars	1000	50 in 20 banks
Scanning speed, channel/s	50	Synth setup time 3 µs	37.42 with auto-tuning mode turned off, 10 kHz sampling step, 2 ms turn-off time
Receiver outputs	Headphones, IBM PC	Headphones. Built-in panoramic indicator from 150 kHz to 2 MHz. Digital IF output. IF 10.7 MHz. IBM PC	Headphones.
Power, V	DC 12 (external)	Battery (4h) DC (10 – 30 V external) power supply	4xAA batteries or 12V D.C. external source
Dimensions, mm	204x77x240	210x88x270	61x143x39
Weight, kg	3,5	5,5	0,340

Literature

1. Brusnitsin N.A. Openness and espionage. M.: Voenizdat, 1991, 56 p.
2. Loginov N.A. Current issues of radio monitoring in the Russian Federation. M.: Radio and communications, 200, 240 p.
3. Petrakov A.V., Lagutin V.S. Protection of subscriber teletraffic: Textbook. allowance. 3rd ed., corrected and expanded. M.: Radio and communication, 2004, 504 p.
4. Covert audio intercept. Volume ont: Catalog. – USA: Serveillance Technology Group (STG), 1993. – 32 p.
5. Discrete surveillance. Navelties: Catalog. – Germany: Helling, 1996. – 13 p.
6. Drahtlose Audioubertragungs – Systeme: Catalog. – Germany: Hildenbrand - Elektronic, 1996 – 25 p.