What is the standard bandwidth in traditional telephony? Standard telephone channel Band of frequencies in telephony.

Term frequency band regarding the signal associated with the concept of effective signal spectrum width, in which 90% of the signal energy is concentrated (by agreement), as well as the lower and upper limits of the signal bandwidth. These the most important characteristics signal source directly related to physics given source signal. For example, for an inductive vibration sensor, the frequency band of the output signal is actually limited from above by units of kilohertz due to the inertia of the mass of the metal magnetized core inside the sensor inductance coil, and from below - by the value associated with the coil inductance. The upper bandwidth limit of a signal is typically associated with the physical limitations of the slew rate, while the lower bandwidth limit is associated with the presence of the low frequency component of the signal, including the DC component.

Term frequency band transmission used in relation to converters and paths (interfaces) of signal transmission. We are talking about amplitude-frequency characteristic (AFC) of these devices and the bandwidth characteristics of that frequency response, which are traditionally measured in terms of -3 dB as shown in the figure above. The maximum (or average, by agreement) value of the signal amplitude in the passband is taken as zero decibel. In the figure, the frequencies F 1 and F 2 are the lower and upper frequencies of the passband, respectively. The lower limit F 1 = 0, if this converter or path passes the DC component of the signal. The more width frequency bands transmission∆F= F 2 - F 1 converter or data path, the higher resolution (detail) of the signal in time , the higher the information transfer rate in the corresponding interface, But at the same time the more interference and noise falls within the passband.

If the signal bandwidth partially or completely does not fall within the transducer or path bandwidth, then this leads to distortion or complete signal suppression in the path.

On the other hand, if the effective bandwidth of the signal is many times narrower than the bandwidth of the converter or path, then this case cannot be considered optimal, since in this physically implemented system there is always noise and interference of various nature, which are generally dispersed over the entire bandwidth of the bandwidth . Passage frequency regions that do not contain useful signal components will add noise, degrading the signal-to-noise ratio in a given signal conversion or transmission channel. Based on these assumptions, we have come close to term: optimal signal bandwidth is the bandwidth whose boundaries are consistent with effective signal bandwidth.

In the case of an ADC, the upper end of the passband can be provided by an anti-aliasing filter, and the lower end can be provided by a high pass filter.

As you can see, the general term frequency band, used in any context, is strongly related to the choice of equipment according to its frequency characteristics, and is also related to the issue of optimal matching of converters and transmission paths with signal sources.

with the term frequency band related articles:

Usually we don't care how the phone line works (but not when we have to shout at the top of our lungs: "Please repeat, I can't hear anything!").

Telephone companies provide a wide variety of customer services. It is not so easy to understand the price lists of these services - what, in fact, is offered, and how much you should pay for which service. In this article, we will not say a word about prices, but we will try to find out what is the difference between the most commonly offered products and services in the field of telephone communications.

ANALOGUE LINES, DIGITAL LINES

First, the lines are analog and digital. The analog signal changes continuously; it always has a certain value, representing, for example, the volume and pitch of the transmitted voice, or the color and brightness of a certain area of ​​the image. Digital signals have only discrete values. As a rule, the signal is either on or off, or it is, or it is not. In other words, its value is either 1 or 0.

Analog phone lines have been used in telephony since time immemorial. Even fifty-year-old phones are likely to be connected to a local loop, the line between a home telephone jack and the central telephone exchange. (The central office is not a glittering skyscraper in the center of the city; the length of the local loop is on average no more than 2.5 miles (four kilometers), so the "central office" is usually located in some nondescript building nearby.)

During telephone conversation the microphone built into the handset converts speech into an analog signal transmitted to the central telephone exchange, from where it goes either to another subscriber loop or to other switching devices if the called number is outside the coverage area of ​​this exchange. When dialing a number, the telephone generates in-band signals transmitted over the same primary channel to indicate to whom the call is intended.

During its existence telephone companies accumulated extensive experience in speech transmission. It has been established that the frequency range from 300 to 3100 Hz is generally sufficient for this task. Recall that hi-fi class audio systems are capable of reproducing sound without distortion in the frequency range of 20-20,000 Hz, which means that the telephone range is usually only enough for the subscriber to recognize the caller by voice (for other applications, this range is likely to be too narrow - for transmitting music, for example, telephone communications totally unsuitable). The telephone companies provide a smooth decrease in the amplitude-frequency characteristic at high and low frequencies using an analog telephone channel of 4000 Hz.

The central telephone exchange, as a rule, digitizes the signal intended for further transmission over telephone network. With the exception of Gilbet County (Arkansas) and Rat Fork (Wyoming), in all American telephone networks, the signal between central stations is transmitted in digital form. Although many companies use digital private exchanges and data communications, and all ISDN facilities are based on digital encryption, local loops are still the "last resort" of analog communications. This is explained by the fact that most telephones in private homes do not have the means of digitizing the signal and cannot work with lines with a bandwidth of more than 4000 Hz.

WHAT DOES 4000 Hz DO?

A modem is a device that converts digital computer signals into analog signals at frequencies within the bandwidth of a telephone line. The maximum bandwidth of a channel is directly related to the bandwidth. More precisely, the amount of throughput (in bits/sec) is determined by the bandwidth and the allowance for the signal-to-noise ratio. Currently, the maximum throughput of modems - 33.6 Kbps - is already close to this limit. Users of 28.8 Kbps modems are well aware that noisy analog lines rarely provide their full throughput, which often turns out to be much lower. Compression, caching, and other evasions help to rectify the situation somewhat, and yet we will live to see the invention of perpetual motion rather than the appearance of modems with a bandwidth of 50 or at least 40 Kbps on ordinary analog lines.

Telephone companies solve the inverse problem - they digitize the analog signal. To transmit the resulting digital signal, channels with a bandwidth of 64 Kbps are used (this is the world standard). Such a channel, called DS0 (digital signal, zero level), is the basic building block from which all other telephone lines are built. For example, you can combine (the correct term is multiplex) 24 DS0 channels into a DS1 channel. By renting a T-1 line, the user actually receives a DS1 channel. When calculating the total throughput of DS1, we must remember that after every 192 information bits (that is, 8000 times per second), one bit of synchronization is transmitted: in total, 1.544 Mbps is obtained (64000 times 24 plus 8000).

LEASED LINES, SWITCHED LINES

In addition to the T-1 line, the client can rent leased lines or use regular switching lines. By leasing a T-1 circuit or a low-speed data line, such as a dataphone digital service (DDS) line, from a telephone company, the subscriber is effectively leasing a direct connection and as a result becomes the only user of a 1.544 Mbit/s (T-1) channel. ) or 56 kbps (low speed line).

Although the frame relay technology involves the switching of individual frames, the corresponding services are offered to the user in the form of virtual communication channels between fixed endpoints. From a network architecture point of view, a frame relay should be considered more like a dedicated rather than a switched line; important is the fact that the price of such a service with the same bandwidth is significantly lower.

Switching services (an example of which is a home telephone service) are services purchased from the telephone company. Upon request, the subscriber is provided with a connection to any node of the telephone network carried out using a network of public switches. Unlike the situation with leased lines, the fee in this case is charged for the connection time or the actual amount of traffic and depends largely on the frequency and volume of network use. Switching services digital communications can be provided based on X.25, Switched 56, ISDN Basic Rate Interface (BRI), ISDN Primary Rate Interface (PRI), Switched Multimegabit Data Service (SMDS), and ATM protocols. Some organizations, such as universities, railways or municipal organizations, create private networks using their own switches and leased, and sometimes even their own lines.

If the line received from the telephone company is digital, there is no need to convert digital signals to and therefore, the need for a modem is eliminated. Nevertheless, in this case, the use of the telephone network imposes certain requirements on the subscriber. Specifically, ensure that the local loop is terminated correctly, that traffic is forwarded correctly, and that diagnostics performed by the telephone company are supported.

A line that supports the ISDN BRI protocol must be connected to a device called NT1 (network termination 1). In addition to terminating the line and supporting diagnostic routines, the NT1 provides a 2-wire loop termination to a 4-wire digital terminal system. When using leased T-1 or DDS digital lines and digital communications services, use a channel service unit (CSU) as the line load. The CSU acts as a terminator, ensures that the line is correctly loaded and processes diagnostic commands. The customer's end equipment interacts with a data service unit (DSU) that converts the digital signals to a standard form and transmits them to the CSU. Structurally, CSU and DSU are often combined into one unit called CSU / DSU. The DSU can be built into a router or multiplexer. Thus, in this case (although modems are not needed here), the installation of certain interface devices will be required.

CARRIERS FOR TELEPHONE COMMUNICATIONS

Most analog local loops can only provide 33.6 Kbps throughput under very favorable conditions. On the other hand, the same twisted pair, which connects the office to the central office, could well be used for ISDN BRI, which gives 128 Kbps of data throughput and another 16 Kbps for management and configuration. What's the matter here? The signal transmitted over analog telephone lines is filtered to suppress all frequencies above 4 kHz. When using digital lines, such filtering is not required, so the bandwidth of the twisted pair turns out to be significantly wider, and, consequently, the throughput also increases.

Leased lines with a bandwidth of 56 and 64 Kbps are two-wire or four-wire digital lines (in the latter case, one pair is used for transmission and the other for reception). These same lines are suitable as carriers for digital communication services, such as frame relay or Switched 56. Four-wire lines or even optical cables are often used as carriers for T-1, as well as ISDN PRI and frame relay. T-3 lines sometimes represent coaxial cable, but more often they are still performed on the basis of optical.

Although ISDN continues to receive the most attention as a means of high-speed signal transmission over long distances, newer means of communication for the "last mile" (ie local loop) have recently appeared. PairGain and AT&T Paradyne offer products based on Bellcore's high bit-rate digital subscriber loop (HDSL) technology. These products allow you to equalize the capabilities of all existing subscriber loops; by installing HDSL devices at both ends of the line, you can get DS1 bandwidth (1.544 Mbps) on almost all existing subscriber loops. (HDSL up to 3.7 km long can be used on subscriber loops without repeaters in the case of standard 24-gauge wires. For normal T-1 lines to work, repeaters must be installed every kilometer and a half). An alternative to HDSL in achieving DS1 throughput on the "last mile" is to either use optical cable (which is very expensive) or install multiple repeaters on each line (this is not as expensive as fiber optic equipment, but still not cheap). In addition, in this case, the costs of the telephone company, and hence the client, to maintain the line in working order increase significantly.

But even HDSL is not the latest technology in the field of increasing throughput on the "last mile". The successor to HDSL, asymmetrical digital subscriber line (ASDL) technology, is expected to be able to deliver 6 Mbps in one direction; the bandwidth of the other is significantly lower - something around 64 Kbps. Ideally, or at least in the absence of anyone's monopoly - assuming that the cost of a service to a customer roughly corresponds to its cost to the telephone company - a large proportion of customers could use ISDN PRI (or other T-1-based services) at a price , comparable to the current price of ISDN BRI.

Today, however, ISDN supporters probably have nothing to worry about; in most cases, telephone companies will choose to increase the capacity of the lines and pocket all the profits without reducing the cost of service to the customer. It is not at all obvious that tariffs for services should be based on common sense.

Table 1. Types of telephone services

Bandwidth (transparency)- frequency range, within which the amplitude-frequency characteristic (AFC) of an acoustic, radio engineering, optical or mechanical device is uniform enough to ensure signal transmission without significant distortion of its shape. Sometimes, instead of the term "bandwidth", the term "effectively transmitted bandwidth (ETB)" is used. The main energy of the signal is concentrated in the EPFC (at least 90%). This frequency range is set for each signal experimentally in accordance with quality requirements.

Basic Bandwidth Options

The main parameters that characterize the frequency bandwidth are the bandwidth and the unevenness of the frequency response within the band.

Bandwidth

Bandwidth - the frequency band within which the unevenness of the frequency response does not exceed the specified one.

Bandwidth is usually defined as the difference between the upper and lower cutoff frequencies of the frequency response section f 2 − f 1 (\displaystyle f_(2)-f_(1)), where the oscillation amplitude is equal to 1 2 (\displaystyle (\frac (1)(\sqrt (2))))(or equivalently 1 2 (\displaystyle (\frac (1)(2))) for power) from the maximum. This level corresponds approximately to −3 dB.

The bandwidth is expressed in units of frequency (eg, hertz).

In radio communications and information transmission devices, the expansion of the bandwidth allows you to transmit large quantity information.

Frequency response unevenness

The uneven frequency response characterizes the degree of its deviation from a straight line parallel to the frequency axis.

The weakening of the frequency response unevenness in the band improves the reproduction of the transmitted signal shape.

Distinguish:

• Absolute bandwidth: 2Δω = Sa
• Relative bandwidth: 2Δω/ωo = So

Specific examples

In antenna theory, bandwidth is the frequency range at which an antenna operates effectively, usually around the center (resonant) frequency. Depends on the type of antenna, its geometry. In practice, the bandwidth is usually determined by a given level of SWR (standing wave ratio), for example, equal to 2.

From the definition of the bandwidth, it can be seen that the dispersion imposes a limit on the transmission distance and on the upper frequency of the transmitted signals.

Bandwidth Requirements various devices determined by their purpose. For example, for telephone communications, a bandwidth of about 3 kHz (300-3400 Hz) is sufficient, for high-quality playback of musical works - at least 30-16000 Hz, and for television broadcasting - up to 8 MHz wide)

Virtually all electrical signals that display real messages contain an infinite spectrum of frequencies. Undistorted transmission of such signals would require a channel with infinite bandwidth. On the other hand, the loss of at least one component of the spectrum at the reception leads to a distortion of the time waveform. Therefore, the task is to transmit a signal in a limited channel bandwidth in such a way that signal distortions meet the requirements and quality of information transmission. Thus, the frequency band is a limited (based on technical and economic considerations and requirements for transmission quality) signal spectrum.

The 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 limitation. So, for a periodic sequence of rectangular pulses, the signal band can approximately be 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 (enhanced regions of the frequency spectrum), most of which are located in the 300 ... 3400 Hz band. Therefore, on the recommendation of the International Consultative Committee on Telephony and Telegraphy (CCITT), an effectively transmitted frequency band of 300 ... 3400 Hz has been adopted for telephone transmission. Such a signal is called a tone frequency (PM) signal. At the same time, the quality of the transmitted signals is quite high - syllabic intelligibility is about 90%, and intelligibility of phrases is 99%.

2.Sound broadcast signals . The sources of sound in the transmission of broadcast programs are musical instruments or the human voice. Range sound signal occupies the frequency band 20…20000 Hz.

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

3. In broadcast television adopted a method of sequentially converting each element of the image into electrical signal with subsequent transmission of this signal over one communication channel. To implement this principle, special cathode-ray tubes are used on the transmitting side, which convert optical image of the transmitted object into a time-expanded electrical video signal.

Figure 2.6 - Design of the transfer tube

As an example, Figure 2.6 shows a simplified version of a transmission tube. Inside a glass flask under high vacuum, a semitransparent photocathode (target) and an electron searchlight (ED) are located. Outside, a deflecting system (OS) is put on the neck of the tube. The searchlight forms a thin electron beam, which is directed towards the target under the influence of the accelerating field. With the help of a deflecting 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 a 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, as if erases (neutralizes) its potential. The current that flows through the load resistance R n will depend on the illumination of the target area on which the electron beam hits, and a video signal U s will be emitted on the load (Figure 2.7). The voltage of the video signal will change from the "black" level, corresponding to the darkest parts of the transmitted image, to the "white" level, corresponding to the brightest parts of the image.

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

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

The television signal is a pulsed unipolar (because it is a function of brightness, which cannot be bipolar) signal. It has a complex shape, and it can be represented as the sum of the constant and harmonic components of oscillations of various frequencies.
The level of the constant component characterizes the average brightness of the transmitted image. When transmitting moving images, the value of the DC component will continuously change according to the illumination. These changes take place with low frequencies(0-3 Hz). The lower frequencies of the video signal spectrum reproduce large image details.

Television, as well as light cinema, became possible thanks to the inertia of vision. The nerve endings of the retina continue to remain excited for some time after the cessation of the light stimulus. At a frame rate Fk ≥ 50 Hz, the eye does not notice the discontinuity of the image change. In television, the reading time of all N lines (frame time - T c) is chosen equal to T c = s. Interlacing is used to reduce image flicker. First, during the half-frame time equal to T p/k = = s, all odd lines are read in turn, then, for the same time, all even lines. The frequency of the video signal spectrum 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 is a pulse close to a rectangular shape. The minimum frequency of this signal at interlaced frequency fields, i.e.

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

With the help of high frequencies, the smallest details of the image are transmitted. Such an image can be represented as 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. This image will contain maximum amount 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 for drawing one line (Fig. 2.9, b) will be equal to . A line-changing signal is obtained when black and white squares alternate. The minimum signal period will be equal to the reading time of a pair of squares:

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

The number of squares (n) per line will be:

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

b is the width, h is the 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 value of the line decomposition frequency (line frequency) is 15.625 kHz. The upper frequency of the TV signal will be equal to 6.5 MHz.

According to the standard adopted in our country, the voltage of the full video signal U TV, consisting of synchronization pulses U C , a luminance signal and quenching pulses U P is U TV = U P + U C = 1V. In this case, U C \u003d 0.3 U TV, and U P \u003d 0.7 U TV. As can be seen from Figure 2.10, the signal sound accompaniment is located higher in the spectrum (fn SV ​​= 8 MHz) of the video signal. Typically, the video signal is transmitted using amplitude modulation (AM), and the audio signal is transmitted using frequency modulation (FM).

Sometimes, in order to save the 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 fn sound = 6.5 MHz.

Figure 2.10 - Placement of the spectra of image and sound signals in the radio channel of television broadcasting.

Practicum (similar tasks are included in exam tickets)

Task number 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 number 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 value of the duration of one line of a standard TV signal. During this time there will be a change 5 pulses corresponding to the black level and 5 pulses corresponding to the white level (you can calculate their duration). Thus, it is possible to determine the frequency of the pulse change and the bandwidth of the signal.

When solving problem No. 2, proceed from the total number of lines in the frame, determine how many lines fall on one horizontal strip, keep in mind that scanning is carried out interlaced. So you determine the duration of the pulse corresponding to the level of black or white. Further, as in task No. 1

When preparing the final work, for convenience, use a graphical representation of signals and spectra.

4. Fax signals. Fax (phototelegraphic) communication is the transmission of still images (drawings, drawings, photographs, texts, newspaper pages, and so on). The facsimile 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 facsimile

Where 1 – facsimile channel; 2 - drive, synchronizing and phasing devices; 3 - transfer drum, on which the original of the transmitted image on paper is placed; FEP - photoelectronic converter of the reflected light flux into an electrical signal; OS - optical system to form 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 frequency of the pattern. The frequency of the pattern, Hz, reaches its maximum value when transmitting an image whose elements and the gaps 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 duration of the image element transmission, 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 sweep pitch (the diameter of the sweeping beam), then there are π·D/S elements in the line. With N revolutions per minute of a drum having a diameter D, the pixel transmission time, measured in seconds:

The minimum frequency of the pattern (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. Wherein

F pus min = N/60, (2.14)

To perform phototelegraph communications of satisfactory quality, it is sufficient to transmit frequencies from F fig min to F fig max . The International Advisory Committee for Telegraphy and Telephony recommends N = 120, 90 and 60 rpm for facsimile machines; S = 0.15 mm; D = 70 mm. From (2.13) and (2.14) it follows that at N = 120 F fig max = 1466 Hz; F fig min = 2 Hz; at N \u003d 60 F fig max \u003d 733 Hz; F fig min = 1 Hz; The dynamic range of the facsimile signal is 25 dB.

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

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

For data transmission, more complex codes are used that allow you to detect and correct errors in the received combination of pulses arising from interference.

Devices for converting telegraphy signals and transmitting data into messages, according to the received combinations of pulses and pauses, restore the signs of the message in accordance with the code table and issue them to the printer or display screen.

The shorter the duration of the pulses that display messages, the more of them will be transmitted per unit of time. The reciprocal of the pulse duration is called the telegraphy speed: B = 1/τ and, where τ and is the pulse duration, s. The unit of telegraphy speed was called the baud. With a pulse duration τ and = 1 s, the speed B = 1 Baud. Telegraphy uses pulses with a duration of 0.02 s, which corresponds to the 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 sufficient to fix only the sign of the pulse with a bipolar signal, or the presence or absence - with a unipolar signal. Pulses can be reliably captured if they are transmitted using a bandwidth that is numerically equal to the baud rate. For a standard telegraphy rate of 50 baud, the width of the spectrum of the telegraph signal will be 50 Hz. At 2400 baud (medium speed data transmission), the signal spectrum width is approximately 2400 Hz.

5. Average power of messages P SR is determined by averaging the results of measurements over a long period of time.

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

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

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

For ease of calculation, 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. With this in mind, the absolute average power level is:

Peak power ppeak (ε %) – ύ is the message power value that can be exceeded for ε % of the time.

The crest factor of the signal 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 R e, taking into account (2.17) and (2.19), we define the crest factor as the difference between the absolute levels of peak and average power:

Under the dynamic range D (ε%) understand 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 tone frequency signal measured in the busiest hour (PHO), taking into account the control signals - dialing, calling, and so on - is 32 μW, which corresponds to the level (compared to 1 mW) pav = -15 dBm

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

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

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

Primary discrete signals usually have the form of rectangular pulses of constant or alternating current, usually with two allowed states (binary or on-off).

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

В = 1/τ and, (2.23)

where τ and is the duration of an elementary message.

The information transfer rate 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.

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

C max \u003d V, bit / s (2.25)

Control questions

1. Define the terms "information", "message", "signal".

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

3. What kinds of signals are there?

4. What is different discrete signal from continuous?

5. What is the difference between the spectrum of a periodic signal and the spectrum non-periodic signal?

6. Define the signal bandwidth.

7. Explain the essence of facsimile transmission of messages.

8. How is the TV image scanned?

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

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

11. Explain the composition of the complete TV signal.

12. Give the concept of dynamic range?

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

2.1.1. Analog telephone networks

Analogue telephone networks are global networks circuit-switched, which were created to provide public telephone services to the public. Analogue telephone networks are focused on the connection, which is established before the start of conversations (voice transmission) between subscribers. The telephone network is formed (switched) using switches of automatic telephone exchanges.

Telephone networks consist of:

• automatic telephone exchanges (ATS);
• telephone sets;
• trunk communication lines (communication lines between automatic telephone exchanges);
• subscriber lines (lines connecting telephone sets with automatic telephone exchange).

The subscriber has a dedicated line that connects his telephone to the PBX. Trunk communication lines are used by subscribers in turn.

Analog telephone networks are also used for data transmission as:

• networks for accessing packet-switched networks, for example, Internet connections (both dial-up and leased telephone lines are used);
• backbones of packet networks (mainly leased telephone lines are used).

The analog circuit-switched telephone network provides services for the packet network. physical layer, which after switching is a physical point-to-point link.

Ordinary telephone network or POTS(Plain Old Telephone Service - the old “flat” telephone service) provides the transmission of a voice signal between subscribers with a frequency range of up to 3.1 kHz, which is quite sufficient for a normal conversation. Used to communicate with subscribers two-wire line, along which the signals of both subscribers during a conversation go simultaneously in opposite directions.

The telephone network consists of many stations that have hierarchical connections to each other. The switches of these exchanges pave the way between the exchange of the calling and called subscribers under the control of information provided by the signaling system. Trunk communication lines between telephone exchanges must provide the possibility of simultaneous transmission of a large amount of information (support a large number of connections).

It is not advisable to allocate a separate trunk line for each connection, and for more efficient use of physical lines, the following is used:

• method of frequency multiplexing of channels;
• digital channels and multiplexing of digital streams from multiple subscribers.

Frequency division multiplexing method (FDM - Frequency Division Multiplexing)

In this case, many channels are transmitted over one cable, in which a low-frequency voice signal modulates the signal of a high-frequency generator. Each channel has its own oscillator, and the frequencies of these oscillators are separated from each other so as to transmit signals in a bandwidth of up to 3.1 kHz with a normal level of separation from each other.

The use of digital channels for trunk transmissions

To do this, an analog signal from subscriber line is digitized at the telephone exchange and then delivered in digital form to the addressee's telephone exchange. There it is converted back and transmitted to the analog subscriber line.

To ensure two-way communication at the telephone exchange, each end of the subscriber line has a pair of converters - ADC (analog-to-digital) and DAC (digital-to-analog). For voice communication with a standard bandwidth (3.1 kHz), a quantization frequency of 8 kHz is adopted. Acceptable dynamic range(the ratio of the maximum signal to the minimum) is provided with an 8-bit conversion.

In total, it turns out that each telephone channel requires a data transfer rate of 64 kbps (8 bits x 8 kHz).

Often, 7-bit samples are also limited to signal transmission, and the eighth (least significant) bit is used for signaling purposes. In this case, the pure voice stream is reduced to 56 kbps.

For efficient use of trunk lines, digital streams from multiple subscribers at telephone exchanges are multiplexed into channels of various capacities connecting telephone exchanges to each other. At the other end of the channel, demultiplexing is performed - the selection of the required stream from the channel.

Multiplexing and demultiplexing, of course, takes place at both ends at the same time, since telephone communication is two-way. Multiplexing is carried out using time division multiplexing (TDM).

In the main channel, information is organized as a continuous sequence of frames. Each subscriber channel in each frame is assigned the time interval during which the data of this channel is transmitted.

Thus, in modern analog telephone lines, analog signals are transmitted through the subscriber line, and digital signals are transmitted in trunk lines.

Modems for dial-up analog telephone lines

Public telephone networks, in addition to voice transmission, allow the transmission of digital data using modems.

A modem (modulator-demodulator) is used to transmit data over long distances using leased and switched telephone lines.

The modulator converts the binary information coming from the computer into analog signals with frequency or phase modulation, the spectrum of which corresponds to the bandwidth of conventional voice telephone lines. The demodulator extracts the encoded binary information from this signal and transmits it to the receiving computer.

The fax modem (fax-modem) allows you to send and receive fax images that are compatible with conventional fax machines.

Modems for leased telephone lines

Leased physical lines have a much wider bandwidth than dial-up lines. For them, special modems are produced that provide data transfer at speeds up to 2048 kbps and over long distances.

xDSL technologies

xDSL technologies are based on the transformation of a subscriber line of a conventional telephone network from analog to digital xDSL (Digital Subscriber Line). The essence of this technology lies in the fact that at both ends of the subscriber line - at the PBX and at the subscriber - separation filters (splitter) are installed.

The low-frequency (up to 3.5 kHz) component of the signal is sent to ordinary telephone equipment (the PBX port and the telephone set at the subscriber), and the high-frequency (above 4 kHz) is used to transmit data using xDSL modems.

xDSL technologies allow the simultaneous use of the same telephone line and for data transmission, and for voice transmission (telephone conversations), which conventional dial-up modems do not allow.