The division of stations into analog and digital is carried out according to 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), has long been the only means of transmitting voice messages over a distance. The possibility of sampling (in time) and quantization (in terms of level) of the parameters of an analog electrical signal (amplitude, frequency or phase) made it possible to convert an analog signal into a digital (discrete), process it software methods and transmit over digital telecommunications networks.
For the transmission of an analog voice signal between two subscribers in the PSTN network (public switched telephone networks), a so-called standard tone frequency (PM) channel is provided, the bandwidth of which is 3100 Hz. In a digital telephony system, the operations of sampling (by time), quantization (by level), coding and redundancy removal (compression) are performed on an analog electrical signal, after which the data stream generated in this way is sent to the receiving subscriber and, upon “arrival” at the destination, is subjected to reverse procedures.
The speech signal is converted according to the appropriate protocol, depending on which network it is transmitted over. At present, the most efficient transmission of a stream 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).
It is said that the concept of voice transmission using digital technologies originated in 1993 at the University of Illinois (USA). During the next flight of the shuttle Endeavor 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 already thousands of people have downloaded Internet program Phone with home page VocalTec and started chatting.
Naturally, other companies very quickly appreciated the prospects that opened up the opportunity to talk, being in different hemispheres and not paying for international calls. Such prospects could not go unnoticed, and already in 1995 the market was hit by a flood of products designed for voice transmission over the Net.
Today, there are several standardized methods for transmitting information that are most widely used in the digital telephony market: these are ISDN, VoIP, DECT, GSM and some others. Let's try to briefly describe the features of each of them.
So what is ISDN?
The abbreviation ISDN stands for Integrated Services Digital Network - a digital network with integrated services. This is a modern generation of the worldwide telephone network, which has the ability to carry any type of information, including fast and correct data transmission (including voice) 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 termination, and each can have its own landline number.
An ordinary telephone is connected to the telephone exchange with a pair of conductors. In this case, only one telephone conversation can be conducted on one pair. At the same time, noise, interference, radio, extraneous voices can be heard in the handset - the disadvantages of analog telephone communication, which "collects" all the interference in its path. In the case of using ISDN, the network termination is set to the subscriber, and the sound, converted by a special decoder into a digital format, is transmitted via a specially designated (also fully 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 (which also provides for the possibility of data transmission with packet switching) with a data transfer rate of 64 kbps. 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 has a bandwidth of 144 Kbps. When working with PRI, the entire digital communication backbone (DS1) is fully used, which gives throughput 2 Mbps. The high speeds offered by ISDN make it ideal for a wide range of modern communications services, including high speed data, screen sharing, video conferencing, large media files, 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 - 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 acquiring and implementing a solution from the manufacturer of this exchange was not commensurate with the convenience achieved.
The first signs of service CTI-applications were the systems of electronic secretaries (autoattended) and automatic interactive voice greetings (menu), corporate voice mail, answering machine and call recording systems. To add the service of one or another CTI application, a computer was connected to the company's existing telephone exchange. A specialized board was installed in it (first on the ISA bus, then on the PCI bus), which was connected to the telephone exchange via a standard telephone interface. Computer software 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 thus provided the implementation of an additional corporate telephony service. Almost at the same time, a standard was developed software interface for computer-telephony integration - 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 translates (translates) telephone signals, the state of the telephone line and its changes into a “software” form: messages, events, variables, constants. The transmission of the telephone component occurs over the telephone network, and the software component - over the data transmission network, IP-network.
And what does the process of integration into IP-telephony look like?
First of all, it should be noted that with the advent of IP-telephony, the very perception of the telephone exchange (Private Branch eXchange - PBX) has changed. IP PBX is nothing more than another IP network service, and, like most IP network services, it operates in accordance with the principles of client-server technology, i.e., it assumes the presence of a service and a client part. So, for example, the service email in 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 similarly arranged: the service part - the IP PBX server and the client part - the IP phone (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 - a rich functionality, the ability to significantly improve the interaction of employees and at the same time simplify the maintenance of the system.
In addition, IP communications are developing in an open manner due to the standardization of protocols and the global penetration of IP. Thanks to the principle of openness in the IP-telephony system, it is possible to expand the services provided, integrate with existing and planned services.
IP-telephony allows you to build a single centralized control system for all subsystems with access rights and operate subsystems in regional divisions by local staff.
The modularity of the IP communications system, its openness, integration and independence of components (unlike traditional telephony) give additional features to build for real fault-tolerant systems, as well as systems with a distributed territorial structure.
DECT wireless communication systems:
The DECT (Digital Enhanced Cordless Telecommunications) wireless access standard is the most popular system mobile communications V corporate network, the cheapest and easiest option for installation. It allows you to organize wireless communication throughout the territory of the enterprise, which is so necessary for "mobile" users (for example, the security of the enterprise or the heads of workshops, 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 a once purchased DECT base, 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 in open space.
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 with any base station cellular network trying to contact several phones located in its vicinity at once. Therefore, stations must provide "multiple access", that is, the simultaneous operation of several phones at once without mutual interference.
In cellular systems of the first generation (standards NMT, AMPS, N-AMPS, etc.), multiple access is implemented frequency method– FDMA (Frequency Division Multiple Access): the base station has multiple receivers and transmitters, each operating on a different 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 in to them. This is no different from the way you tune a particular radio wave.
However, the number of channels that can be allocated at the base station is not very large, especially since neighboring stations of the cellular network must have different sets of frequencies so as not to create mutual interference. In most cellular networks of the second generation, the frequency-time method of channel separation, TDMA (Time Division Multiple Access), began to be used. In such systems (and these are networks of GSM, D-AMPS, etc.) different frequencies are also used, but only each such channel is allocated to the phone not for the entire time of communication, but only for short periods of time. The rest of the same intervals are alternately used by other phones. Helpful information in such systems (including speech signals) is transmitted in a "compressed" form and in digital form.
Sharing each frequency channel by several phones allows you to provide service more subscribers, but the frequencies are still not enough. CDMA technology, built on the principle of code division of signals, was able to significantly improve this situation.
The essence of the method of code division of signals used in CDMA is that all phones and base stations simultaneously use the same (and at the same time all at once) the 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 "color" that ensures its confident selection from the background of others.
Over the past five years, CDMA technology has been tested, standardized, licensed and commercialized by most wireless equipment vendors and is already in use worldwide. 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 opportunities "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, for GSM standard in Russia, frequencies above 900 MHz are assigned, so the operating ranges of CDMA and GSM networks do not intersect 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 conventional wired one. If we also take into account the fact that the number of such applicants in big companies can easily exceed a thousand, then we get a set of requirements that only a powerful modern digital exchange (UPBX) can satisfy.
There are many solutions on the market today 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 these criteria: they have the ability to switch broadband channels, packet switching, simply integrate with computer systems(CTI) and allow organizing wireless microcells within corporations (DECT).
Which of the following types of communication is better? Decide for yourself.
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.
First, the lines are analog and digital. analog signal changes continuously; he always has certain value A that represents, for example, the volume and pitch of the transmitted voice, or the color and brightness of a particular 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 a central telephone exchange. (The central office is not a shiny skyscraper in the center of the city; the length of the local loop does not exceed 2.5 miles (four kilometers) on average, so the "central office" is usually located in some nondescript building nearby.)
During a telephone conversation, the microphone built into the handset converts speech into an analog signal transmitted to the central telephone exchange, from where it enters either another subscriber loop or 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 their existence, telephone companies have accumulated a lot of experience in voice 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). 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 the telephone network. With the exception of Gilbet County (Arkansas) and Rat Fork (Wyoming), in all American telephone networks, the signal between the 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 phones 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.
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 is often 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).
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 the telephone company, the subscriber is effectively leasing a direct connection and as a result becomes the only user of a 1.544 Mbps (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. Digital communications switching services 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 procedures, 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.
Most analog local loops can only provide 33.6 Kbps throughput under very favorable conditions. On the other hand, the same twisted-pair cable connecting the office to the central office could well be used for ISDN BRI, giving 128Kbps of data throughput and another 16Kbps 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). The 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 are sometimes coaxial cable, but more often they are still based on optical.
Although ISDN continues to attract the most attention as a means of high-speed signal transmission to 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. Regular T-1 lines require repeaters every kilometer and a half to work). An alternative to HDSL in achieving DS1 throughput on the "last mile" is either to use optical cable (which is very expensive) or to install several repeaters on each line (this is not as expensive as fiber optic equipment, but still not cheap). Besides, in this case the expenses of the telephone company, and consequently the client, for maintaining 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 cost of , 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.
line type | Service | Switching type | Subscriber loop carrier |
analog line | Line switching | 2-wire twisted pair |
|
DS0(64 Kbps) | DDS (leased line) | Dedicated line | |
Switched PVC | Two- or four-wire twisted pair |
||
Switching | Two- or four-wire twisted pair |
||
Line switching | Two- or four-wire twisted pair |
||
Line switching | Two- or four-wire twisted pair |
||
Line switching | 2-wire twisted pair |
||
Multiple DS0s (from 64 Kbps to 1536 Mbps Step 64 Kbps) | Dedicated line | Two- or four-wire twisted pair |
|
Switched PVC | Two- or four-wire twisted pair |
||
(1544 Mbps) (24 lines DS0) | Leased line T-1 | Dedicated line | |
Switched PVC | 4-wire twisted pair or fiber optic |
||
Packet switching | 4-wire twisted pair or fiber optic |
||
Line switching | 4-wire twisted pair or fiber optic |
||
(44736 Mbps) (28 lines DS1, 672 DS0 lines) | Cellular switching | ||
Packet switching | Coaxial cable or fiber optic |
Answers: 9
Question for connoisseurs: What is the band of transmitted audio frequencies in telephone communications
Sincerely, Nurslan
Nikolay Ivanov:
300 Hz - 3400 Hz. or narrowed 0.3 - 2.7 kHz
What does sound frequency mean, the frequency is in the transmission channel - wireless or wired - this is the frequency of the electromagnetic wave, and the sound frequency depends on the speaker in the handset. Sound is not transmitted in connected channels))
battalion commander:
effectively transmitted frequency band of telephone channels 0.3-3.4 kHz (standard telephone channel), narrowed channels 0.3-2.7 kHz
This video will help you understand
Vladimir Nikolaev:
if the signal has a sinusoidal form, then its band is one frequency of this sinusoid; if the signal is pulsed, then it can be expanded into a Fourier series, it will represent several sinusoidal frequencies, so the entire band occupied by these frequencies is called the band
An Drew:
Ruslan Mamyshev:
your words were not found - start with this, and when you understand - ask a more interesting question ...
Free wind:
Well, the question itself is the answer - the frequency band, in short from now to now .... It’s already scary to go to Wikipedia, felt boots and there, ultrasound was pulled up from 20 kHz to 1 GHz, I almost fell, and also hypersound over 1 GHz ....)))))))))))) What kind of felt boot did they hit him with? What is written in the wiki ....
Everything is fine:
Any signal finite in time has an INFINITELY large spectrum width.
Should talk about
effective spectral width in which 90% of the energy is concentrated (by agreement)
signal.
osnovy-electrotechniki. en/energeticheskie-xarakteristiki/
The tone frequency channel (eng. voice frequency circuit) is a set of technical means and distribution medium that provides transmission electrical signals communications in the effectively transmitted frequency band (EPCH) 0.3 - 3.4 kHz. In telephony and communications, the abbreviation KTC is often used. A voice-frequency channel is a unit of measure for the capacitance (compression) of analog transmission systems (eg K-24, K-60, K-120). At the same time for digital systems transmission (for example, IKM-30, IKM-480, IKM-1920) the unit of capacity is the main digital channel.
Effectively transmitted frequency band - a frequency band, the residual attenuation at the 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 inherent in this system.
The width of the EPFC determines the quality of the telephone transmission, and the possibility of using the telephone channel for the transmission of other types of communication. In accordance with the international standard for telephone channels of multichannel equipment, EPFC is installed from 300 to 3400 Hz. With such a band, a high degree of intelligibility of speech is ensured, a good naturalness of its sound, and great opportunities are created for the secondary compaction of telephone channels.
Date:2016/4/18 16:13:20 Hits:
Jan Poole
Notes and details on passband, spectrum and sideband FM and their impact on FM usage.
Bandwidth, spectrum and sidebands are of great importance when using frequency modulation.
The sidebands of the modulated frequency signal extend on either side of the main carrier, and cause the bandwidth of the overall signal to increase far beyond that of the unmodulated carrier.
As the modulation of the carrier changes, so do the sidebands and hence the bandwidth and overall spectrum of the signal.
Bessel function modulation frequency and sidebands
Any signal that is modulated produces sidebands. In the case of an amplitude modulated signal, they are easy to determine, but for frequency modulation the situation is not so simple. , They depend on not only the deflection, but also the level of the deflection, that is, the modulation index M. The full spectrum is an infinite series of discrete spectral components expressed by a complex formula using a Bessel function of the first kind.
Relative levels of carrier and sidebands for a frequency modulated signal
For small values of the modulation index, when using narrowband FM, and the FM signal consists of a carrier and two sidebands spaced at the modulation frequency on both sides of the carrier. This looks the same as the AM signal, but the difference is that the lower sideband is 180 degrees out of phase.
As the modulation index increases, it is found that other sidebands with twice the modulation frequency begin to appear. As the index increases, other additional sidebands can also be seen.
Spectra of an FM signal with different modulation index levels
At certain modulation levels, where the modulation index is equal to figures 2.41, 5.53, 8.65 and other higher specific levels, the carrier falls on the actual figure figures of zero, then the signal consists simply of sidebands.
frequency modulation bandwidth
In the case of an amplitude modulated signal, the required bandwidth is twice the maximum modulation frequency. Although the same is true for a narrowband FM signal, the situation is not true for a wideband FM signal. Here, the required bandwidth can be very large, with detectable sidebands spreading over large amounts of frequency spectrum. It is usually necessary to limit the bandwidth of the signal so that it does not cause unnecessary interference to stations on either side.
As a frequency modulated signal has sidebands that extend to infinity, it is normally accepted practice to define a bandwidth that contains approximately 98% of the signal's power.
The Rule of Thumb, often referred to as Carson's Rule states that 98% of the signal power is contained in a bandwidth equal to the deflection frequency plus twice the modulation frequency, i.e.:
Typically, the bandwidth of a broadband FM signal is limited by the Carson Rule limit - this reduces interference and does not introduce any unreasonable signal distortion. In other words, for a VHF-FM broadcast station, this should be (2 x 75) + 15 kHz, i.e. 175 kHz. With this in mind, a total of 200 kHz is usually allowed, which allows stations to have a small guardband and their center frequencies at integers of 100 kHz.
Key Points for Modulation Bandwidth and Sidebands
There are several points of interest with respect to the total modulation bandwidth:
The bandwidth of the modulated signal varies with both the frequency deviation and the modulation factor.
Increasing the modulation frequency reduces the modulation index - this reduces the number of sidebands with significant amplitude and, therefore, the bandwidth.
Increasing the modulation frequency increases the frequency separation between the sidebands.
The frequency of the modulation bandwidth increases with the modulation frequency, but is not directly proportional to it.
Remote meetings with poor audio quality 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 teleconferencing. The following is short description various technical requirements for sound quality.
A person is 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 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 it is called narrowband communication. For analog telephony, this means the loss of a significant part of the speech frequency range. This makes speech less natural and difficult to understand than if the frequency range were greater. Compare this to FM radio, which has a frequency range of up to 15 kHz, allowing both voices and music to be reproduced much more naturally.
Analog telephony has an extremely limited frequency response(about 3.2 kHz). The analog signal is perceived by some as more natural, although the digital signal generally has a wider frequency range. This is because the human ear perceives artificial sound very well.
The term "bandwidth" refers to the amount of information per second that is transmitted over the network. The term "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 bandwidth are not the same thing. What's more, bandwidth can be expressed in both Hertz and bits per second (you'll usually see Mbps on the network). The sound is converted to digital networks. The audio signal is measured thousands of times per second and converted into a digital signal.
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. Audio on 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 using broadband standards and have launched what is known as HD Voice technology. However, for this technology to work, the phone must also support this standard. Poor transmission and reception conditions can also affect the sound quality. In this case, the system automatically reduces the transmission speed on the network. This has a negative effect on the sound quality.
Telephony over a data network is called VoIP (Voice over IP). The sound in digital networks was originally of the same quality as in the old analog technology, ie. the audio bandwidth was 3.2 kHz (narrowband). This was necessary in the early digital networks, as data bandwidth was clearly limited.
On digital networks, sound quality is limited primarily by the codec that has been chosen. The codec is part software on the phone, which compresses outgoing analog audio into data packets and converts incoming data packets into analog audio. Thus, modern phones, which support wideband codecs, are able to provide the best sound. The last 10-15 years have seen fantastic advances in VoIP.
Common designations for various codecs are wideband codec (7 kHz), super wideband codec (14 kHz), and full bandwidth codec (20 kHz). There is also a wide range of technical solutions and standards: G.718, G.722.2, G.729.1, etc.
Of course, throughput broadband network and/or mobile network in the office 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 the bandwidth of the audio channel. It can be wireless systems, such as DECT and Bluetooth®, or older network products.
Bluetooth® is a standard that was originally developed to allow various accessories to connect over wireless network to a mobile phone or computer. Bluetooth® only works over short distances between mobile phone and accessories. There is additional audio data compression that can adversely affect audio quality. The trend is increasingly towards state-of-the-art Bluetooth® technology supporting HD audio.
DECT solutions for wireless telephony in offices and factories were originally developed for use with analog telephony. On a DECT network, it is not possible to get better sound quality than standard telephone quality (3.2 kHz). It hardly matters for regular phone calls, but if you want to hold meetings where sound quality is especially important, using direct connections (cables) to a VoIP network might 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 a 7 kHz audio bandwidth.
Konftel products always provide optimum sound quality. If the network distributes HD audio, you will get HD audio on Konftel conference phones.
This shows that there is reason to analyze the communication needs of your business and organization before choosing a network and upgrading your telephony and data infrastructure. For example, a VoIP network with wideband codecs (7 kHz) is better equipped to deliver superior sound than analog or older mobile network. This may be obvious, but on the other hand, portability and simplicity can be key 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. With an analog DECT connection, it can transmit a bandwidth of 3.2 kHz, while a USB connection for a computer can use wideband codecs (7 kHz). You can also connect it to your mobile phone with a cable.
The same device also provides wireless HD audio (wideband) in IP telephony when the DECT 10 base station from Konftel is connected via SIP. It can have up to 5 registered Konftel 300Wx calls. It is possible to configure the Konftel 300Wx with base stations IP DECT provided by third parties supported by Konftel. However, Konftel IP DECT 10 offers unique advantages and makes the job easier.
Whatever your needs, the Konftel range has products that make teleconferencing at your desk and large meetings in conference rooms easier and faster.
