1.
Multiplexing
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In telecommunications and computer networks, multiplexing is a method by which multiple analog or digital signals are combined into one signal over a shared medium. The aim is to share an expensive resource, for example, in telecommunications, several telephone calls may be carried using one wire. Multiplexing originated in telegraphy in the 1870s, and is now applied in communications. In telephony, George Owen Squier is credited with the development of telephone carrier multiplexing in 1910, the multiplexed signal is transmitted over a communication channel such as a cable. The multiplexing divides the capacity of the channel into several logical channels. A reverse process, known as demultiplexing, extracts the original channels on the receiver end, a device that performs the multiplexing is called a multiplexer, and a device that performs the reverse process is called a demultiplexer. Multiple variable bit rate digital bit streams may be transferred efficiently over a fixed bandwidth channel by means of statistical multiplexing. This is an asynchronous mode time-domain multiplexing which is a form of time-division multiplexing, digital bit streams can be transferred over an analog channel by means of code-division multiplexing techniques such as frequency-hopping spread spectrum and direct-sequence spread spectrum. In wired communication, space-division multiplexing, also known as Space-division multiple access is the use of separate point-to-point electrical conductors for each transmitted channel, in wireless communication, space-division multiplexing is achieved with multiple antenna elements forming a phased array antenna. Examples are multiple-input and multiple-output, single-input and multiple-output and multiple-input and single-output multiplexing, different antennas would give different multi-path propagation signatures, making it possible for digital signal processing techniques to separate different signals from each other. These techniques may also be utilized for space diversity or beamforming rather than multiplexing Frequency-division multiplexing is inherently an analog technology, FDM achieves the combining of several signals into one medium by sending signals in several distinct frequency ranges over a single medium. In FDM the signals are electrical signals, one of the most common applications for FDM is traditional radio and television broadcasting from terrestrial, mobile or satellite stations, or cable television. Only one cable reaches a customers residential area, but the provider can send multiple television channels or signals simultaneously over that cable to all subscribers without interference. Receivers must tune to the frequency to access the desired signal. A variant technology, called wavelength-division multiplexing is used in optical communications, time-division multiplexing is a digital technology which uses time, instead of space or frequency, to separate the different data streams. TDM involves sequencing groups of a few bits or bytes from each individual input stream, one after the other, if done sufficiently quickly, the receiving devices will not detect that some of the circuit time was used to serve another logical communication path. Consider an application requiring four terminals at an airport to reach a central computer, each terminal communicated at 2400 baud, so rather than acquire four individual circuits to carry such a low-speed transmission, the airline has installed a pair of multiplexers. A pair of 9600 baud modems and one dedicated analog communications circuit from the ticket desk back to the airline data center are also installed
2.
Modulation
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Most radio systems in the 20th century used frequency modulation or amplitude modulation to make the carrier carry the radio broadcast. Modulation of a sine waveform transforms a narrow frequency range baseband message signal into a passband signal, a modulator is a device that performs modulation. A demodulator is a device that performs demodulation, the inverse of modulation, a modem can perform both operations. The aim of digital modulation is to transfer a digital bit stream over an analog bandpass channel, in music synthesizers, modulation may be used to synthesise waveforms with an extensive overtone spectrum using a small number of oscillators. In this case the carrier frequency is typically in the order or much lower than the modulating waveform. In analog modulation, the modulation is applied continuously in response to the information signal. Where an AM carrier also carries a variable phase f. TM is f In digital modulation, a carrier signal is modulated by a discrete signal. Digital modulation methods can be considered as digital-to-analog conversion, and the corresponding demodulation or detection as analog-to-digital conversion, the changes in the carrier signal are chosen from a finite number of M alternative symbols. A simple example, A telephone line is designed for transferring audible sounds, for example tones, computers may however communicate over a telephone line by means of modems, which are representing the digital bits by tones, called symbols. If there are four symbols, the first symbol may represent the bit sequence 00, the second 01, the third 10. If the modem plays a melody consisting of 1000 tones per second, since each tone represents a message consisting of two digital bits in this example, the bit rate is twice the symbol rate, i. e.2000 bits per second. This is similar to the used by dialup modems as opposed to DSL modems. According to one definition of digital signal, the signal is a digital signal. According to another definition, the modulation is a form of digital-to-analog conversion, most textbooks would consider digital modulation schemes as a form of digital transmission, synonymous to data transmission, very few would consider it as analog transmission. The most fundamental digital modulation techniques are based on keying, PSK, FSK, a finite number of frequencies are used. ASK, a number of amplitudes are used. QAM, a number of at least two phases and at least two amplitudes are used
3.
Amplitude modulation
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Amplitude modulation is a modulation technique used in electronic communication, most commonly for transmitting information via a radio carrier wave. In amplitude modulation, the amplitude of the wave is varied in proportion to the waveform being transmitted. That waveform may, for instance, correspond to the sounds to be reproduced by a loudspeaker and this technique contrasts with frequency modulation, in which the frequency of the carrier signal is varied, and phase modulation, in which its phase is varied. AM was the earliest modulation method used to transmit voice by radio and it was developed during the first two decades of the 20th century beginning with Roberto Landell De Moura and Reginald Fessendens radiotelephone experiments in 1900. It remains in use today in many forms of communication, for example it is used in two way radios, VHF aircraft radio, Citizens Band Radio, and in computer modems. AM is often used to refer to mediumwave AM radio broadcasting, when it reaches its destination, the information signal is extracted from the modulated carrier by demodulation. In amplitude modulation, the amplitude or strength of the oscillations is what is varied. For example, in AM radio communication, a continuous wave signal has its amplitude modulated by an audio waveform before transmission. The audio waveform modifies the amplitude of the wave and determines the envelope of the waveform. In the frequency domain, amplitude modulation produces a signal with power concentrated at the carrier frequency, each sideband is equal in bandwidth to that of the modulating signal, and is a mirror image of the other. Standard AM is thus sometimes called double-sideband amplitude modulation to distinguish it more sophisticated modulation methods also based on AM. One disadvantage of all amplitude modulation techniques is that the receiver amplifies and detects noise, increasing the received signal to noise ratio, say, by a factor of 10, thus would require increasing the transmitter power by a factor of 10. For this reason AM broadcast is not favored for music and high fidelity broadcasting, another disadvantage of AM is that it is inefficient in power usage, at least two-thirds of the power is concentrated in the carrier signal. The carrier signal contains none of the information being transmitted. However its presence provides a means of demodulation using envelope detection, providing a frequency. The receiver may regenerate a copy of the frequency from a greatly reduced pilot carrier to use in the demodulation process. Even with the carrier totally eliminated in double-sideband suppressed-carrier transmission, carrier regeneration is possible using a Costas phase-locked loop and this doesnt work however for single-sideband suppressed-carrier transmission, leading to the characteristic Donald Duck sound from such receivers when slightly detuned. Single sideband is used widely in amateur radio and other voice communications both due to its power efficiency and bandwidth efficiency
4.
Frequency modulation
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In telecommunications and signal processing, frequency modulation is the encoding of information in a carrier wave by varying the instantaneous frequency of the wave. This contrasts with amplitude modulation, in which the amplitude of the wave varies. This modulation technique is known as frequency-shift keying, FSK is widely used in modems and fax modems, and can also be used to send Morse code. Frequency modulation is used for FM radio broadcasting. For this reason, most music is broadcast over FM radio, frequency modulation has a close relationship with phase modulation, phase modulation is often used as an intermediate step to achieve frequency modulation. Mathematically both of these are considered a case of quadrature amplitude modulation. While most of the energy of the signal is contained within fc ± fΔ, the frequency spectrum of an actual FM signal has components extending infinitely, although their amplitude decreases and higher-order components are often neglected in practical design problems. Mathematically, a baseband modulated signal may be approximated by a continuous wave signal with a frequency fm. This method is also named as Single-tone Modulation. As in other systems, the modulation index indicates by how much the modulated variable varies around its unmodulated level. e. The maximum deviation of the frequency from the carrier frequency. For a sine wave modulation, the index is seen to be the ratio of the peak frequency deviation of the carrier wave to the frequency of the modulating sine wave. If h ≪1, the modulation is called narrowband FM, sometimes modulation index h<0.3 rad is considered as Narrowband FM otherwise Wideband FM. In the case of digital modulation, the carrier f c is never transmitted, rather, one of two frequencies is transmitted, either f c + Δ f or f c − Δ f, depending on the binary state 0 or 1 of the modulation signal. If h ≫1, the modulation is called wideband FM, if the frequency deviation is held constant and the modulation frequency increased, the spacing between spectra increases. The carrier and sideband amplitudes are illustrated for different modulation indices of FM signals, for particular values of the modulation index, the carrier amplitude becomes zero and all the signal power is in the sidebands. Since the sidebands are on sides of the carrier, their count is doubled, and then multiplied by the modulating frequency to find the bandwidth. For example,3 kHz deviation modulated by a 2.2 kHz audio tone produces an index of 1.36. Suppose that we limit ourselves to only those sidebands that have an amplitude of at least 0.01
5.
Quadrature amplitude modulation
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Quadrature amplitude modulation is both an analog and a digital modulation scheme. The modulated waves are summed, and the waveform is a combination of both phase-shift keying and amplitude-shift keying, or, in the analog case, of phase modulation. In the digital QAM case, a number of at least two phases and at least two amplitudes are used. PSK modulators are often designed using the QAM principle, but are not considered as QAM since the amplitude of the carrier signal is constant. QAM is used extensively as a scheme for digital telecommunication systems. Arbitrarily high spectral efficiencies can be achieved with QAM by setting a suitable size, limited only by the noise level. QAM is being used in optical systems as bit rates increase, QAM16. Like all modulation schemes, QAM conveys data by changing some aspect of a signal, or the carrier wave. In the case of QAM, the amplitude of two waves of the frequency, 90° out-of-phase with each other are changed to represent the data signal. Amplitude modulating two carriers in quadrature can be viewed as both amplitude modulating and phase modulating a single carrier. Phase modulation and phase-shift keying can be regarded as a case of QAM. This can also be extended to frequency modulation and frequency-shift keying, at the receiver, these two modulating signals can be demodulated using a coherent demodulator. Such a receiver multiplies the received signal separately with both a cosine and sine signal to produce the received estimates of I and Q respectively, because of the orthogonality property of the carrier signals, it is possible to detect the modulating signals independently. This filtered signal is unaffected by Q, showing that the component can be received independently of the quadrature component. Similarly, we may multiply s by a wave and then low-pass filter to extract Q. Analog QAM suffers from the problem as Single-sideband modulation, the exact phase of the carrier is required for correct demodulation at the receiver. If the demodulating phase is even a little off, it results in crosstalk between the modulated signals and this issue of carrier synchronization at the receiver must be handled somehow in QAM systems. The coherent demodulator needs to be exactly in phase with the received signal and this is achieved typically by transmitting a burst subcarrier or a Pilot signal
6.
Single-sideband modulation
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Amplitude modulation produces an output signal that has twice the bandwidth of the original baseband signal. Single-sideband modulation avoids this bandwidth doubling, and the power wasted on a carrier, at the cost of increased device complexity, radio transmitters work by mixing a radio frequency signal of a specific frequency, the carrier wave, with the signal to be broadcast. That is, the signal has a spectrum with twice the bandwidth of the original input signal. In conventional AM radio, this signal is sent to the radio frequency amplifier. Due to the nature of the process, the quality of the resulting signal can be defined by the difference between the maximum and minimum signal energy. Normally the maximum signal energy will be the carrier itself, perhaps twice as powerful as the mixed signals, SSB takes advantage of the fact that the entire original signal is encoded in either one of these sidebands. It is not necessary to broadcast the entire mixed signal, a receiver can extract the entire signal from either the upper or lower sideband. This means that the amplifier can be used more efficiently. A transmitter can choose to only the upper or lower sideband. By doing so, the amplifier only has to work effectively on one half the bandwidth, as a result, SSB transmissions use the available amplifier energy more efficiently, providing longer-range transmission with little or no additional cost. The first U. S. patent for SSB modulation was applied for on December 1,1915 by John Renshaw Carson, the U. S. Navy experimented with SSB over its radio circuits before World War I. SSB first entered service on January 7,1927 on the longwave transatlantic public radiotelephone circuit between New York and London. The high power SSB transmitters were located at Rocky Point, New York and Rugby, the receivers were in very quiet locations in Houlton, Maine and Cupar Scotland. SSB was also used over long distance lines, as part of a technique known as frequency-division multiplexing. FDM was pioneered by telephone companies in the 1930s and this enabled many voice channels to be sent down a single physical circuit, for example in L-carrier. SSB allowed channels to be spaced just 4,000 Hz apart, amateur radio operators began serious experimentation with SSB after World War II. The Strategic Air Command established SSB as the standard for its aircraft in 1957. It has become a de facto standard for voice radio transmissions since then
7.
Circuit switching
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Circuit switching is a method of implementing a telecommunications network in which two network nodes establish a dedicated communications channel through the network before the nodes may communicate. The circuit guarantees the full bandwidth of the channel and remains connected for the duration of the communication session, the circuit functions as if the nodes were physically connected as with an electrical circuit. The defining example of a network is the early analog telephone network. When a call is made from one telephone to another, switches within the telephone exchanges create a continuous wire circuit between the two telephones, for as long as the call lasts. Circuit switching contrasts with packet switching which divides the data to be transmitted into packets transmitted through the network independently. In circuit switching, the bit delay is constant during a connection, as opposed to packet switching, no circuit can be degraded by competing users because it is protected from use by other callers until the circuit is released and a new connection is set up. Even if no actual communication is taking place, the channel remains reserved and protected from competing users and its advantage is that it provides for continuous transfer without the overhead associated with packets making maximal use of available bandwidth for that communication. Its disadvantage is that it can be inefficient because unused capacity guaranteed to a connection cannot be used by other connections on the same network. For call setup and control, it is possible to use a separate dedicated signalling channel from the end node to the network, ISDN is one such service that uses a separate signalling channel while plain old telephone service does not. Early telephone exchanges are an example of circuit switching. The subscriber would ask the operator to connect to another subscriber, whether on the exchange or via an inter-exchange link. In any case, the end result was an electrical connection between the two subscribers telephones for the duration of the call. The copper wire used for the connection could not be used to other calls at the same time, even if the subscribers were in fact not talking. Circuit switching contrasts with packet switching which divides the data to be transmitted into small units, called packets, packet switching shares available network bandwidth between multiple communication sessions. In contrast, packet switching is the process of segmenting data to be transmitted into several smaller packets. Each packet is labeled with its destination and a number for ordering related packets. Each packet is dispatched independently and each may be routed via a different path, at the destination, the original message is reordered based on the packet number to reproduce the original message. As a result, datagram packet switching networks do not require a circuit to be established, public switched telephone network B channel of ISDN Circuit Switched Data and High-Speed Circuit-Switched Data service in cellular systems such as GSM Datakit X
8.
Frequency-division multiplexing
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This allows a single transmission medium such as the radio spectrum, a cable or optical fiber to be shared by multiple independent signals. Another use is to carry separate serial bits or segments of a higher signal in parallel. The most natural example of frequency-division multiplexing is radio and television broadcasting, another example is cable television, in which many television channels are carried simultaneously on a single cable. The multiple separate information signals that are sent over an FDM system, at the source end, for each frequency channel, an electronic oscillator generates a carrier signal, a steady oscillating waveform at a single frequency that serves to carry information. The carrier is much higher in frequency than the baseband signal, the carrier signal and the baseband signal are combined in a modulator circuit. The modulator alters some aspect of the signal, such as its amplitude, frequency, or phase, with the baseband signal. The result of modulating the carrier with the signal is to generate sub-frequencies near the carrier frequency, at the sum. The information from the signal is carried in sidebands on each side of the carrier frequency. Therefore, all the information carried by the channel is in a band of frequencies clustered around the carrier frequency. Similarly, additional signals are used to modulate carriers at other frequencies. The carriers are spaced far apart in frequency that the band of frequencies occupied by each channel. All the channels are sent through the medium, such as a coaxial cable, optical fiber. As long as the frequencies are spaced far enough apart that none of the passbands overlap. Thus the available bandwidth is divided into slots or channels, each of which can carry a modulated signal. The frequencies subtract, producing the signal for that channel again. The resulting baseband signal is filtered out of the other frequencies, for shorter distances, cheaper balanced pair cables were used for various systems including Bell System K- and N-Carrier. By the end of the 20th Century, FDM voice circuits had become rare, modern telephone systems employ digital transmission, in which time-division multiplexing is used instead of FDM. Since the late 20th century digital subscriber lines have used a Discrete multitone system to divide their spectrum into frequency channels, the concept corresponding to frequency-division multiplexing in the optical domain is known as wavelength-division multiplexing
9.
Wavelength-division multiplexing
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This technique enables bidirectional communications over one strand of fiber, as well as multiplication of capacity. This is purely convention because wavelength and frequency communicate the same information, a WDM system uses a multiplexer at the transmitter to join the several signals together, and a demultiplexer at the receiver to split them apart. With the right type of fiber it is possible to have a device that does both simultaneously, and can function as an optical add-drop multiplexer, the optical filtering devices used have conventionally been etalons. As there are three different WDM types, whereof one is called WDM, we would use the notation xWDM when discussing the technology as such. The concept was first published in 1978, and by 1980 WDM systems were being realized in the laboratory, the first WDM systems combined only two signals. Modern systems can handle 160 signals and can expand a basic 100 Gbit/s system over a single fiber pair to over 16 Tbit/s. A system of 320 channels in also present WDM systems are popular with telecommunications companies because they allow them to expand the capacity of the network without laying more fiber. By using WDM and optical amplifiers, they can accommodate several generations of development in their optical infrastructure without having to overhaul the backbone network. Capacity of a link can be expanded simply by upgrading the multiplexers and demultiplexers at each end. This is often done by use of optical-to-electrical-to-optical translation at the edge of the transport network. Most WDM systems operate on single-mode fiber optical cables, which have a diameter of 9 µm. Certain forms of WDM can also be used in multi-mode fiber cables which have diameters of 50 or 62.5 µm. Early WDM systems were expensive and complicated to run, however, recent standardization and better understanding of the dynamics of WDM systems have made WDM less expensive to deploy. Optical receivers, in contrast to laser sources, tend to be wideband devices, therefore, the demultiplexer must provide the wavelength selectivity of the receiver in the WDM system. WDM systems are divided into three different wavelength patterns, normal, coarse and dense, normal WDM uses the two normal wavelengths 1310 and 1550 on one fiber. Coarse WDM provides up to 16 channels across multiple transmission windows of silica fibers, dense wavelength division multiplexing uses the C-Band transmission window but with denser channel spacing. Channel plans vary, but a typical DWDM system would use 40 channels at 100 GHz spacing or 80 channels with 50 GHz spacing, some technologies are capable of 12.5 GHz spacing. New amplification options enable the extension of the wavelengths to the L-band
10.
Polarization-division multiplexing
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It is used in microwave links such as satellite television downlinks to double the bandwidth by using two orthogonally polarized feed antennas in satellite dishes. It is also used in fiber optic communication by transmitting separate left, polarization techniques have long been used in radio transmission to reduce interference between channels, particularly at VHF frequencies and beyond. Under some circumstances, the rate of a radio link can be doubled by transmitting two separate channels of radio waves on the same frequency, using orthogonal polarization. These two separate channels can be received by vertical and horizontal feed antennas at the receiving station, for satellite communications, orthogonal circular polarization is often used instead, as the sense of circular polarization is not changed by the relative orientation of the antenna in space. Although two separate single-polarization antennas can be used for PDM, radiating two independent polarization states can be easily achieved by means of a single dual-polarization antenna. The circular or square waveguide port is needed so that at least two modes are supported. An ad-hoc component must be introduced in such situations to merge two separate single-polarized signals into one dual-polarized physical interface, namely an ortho-mode transducer. Companies working on commercial PDM technology include Siae Microelettronica, Huawei, such operation is carried out by a cross-polarization-interference cancellation, typically implemented as a baseband digital stage. An XPIC typically acts on one of the received signals C containing the signal as dominant term. The XPIC algorithm multiplies the X by a coefficient and then adds it to the received C. The complex recombination coefficient is adjusted adaptively to maximize the MMSE as measured on the recombination, once the MMSE is improved to the required level, the two terminals can switch to high-order modulations. Polarization-division multiplexing is used together with phase modulation or optical QAM. Sets of PDM wavelength signals can then be carried over wavelength-division multiplexing infrastructure, multiple polarization signals can be combined to form new states of polarization, which is known as parallel polarization state generation. Over a long-distance system, these drifts accumulate progressively without limit, resulting in rapid, and cross-polarization modulation are other phenomena that can cause problems in PDM systems. Modulations used include PDM-QPSK and PDM-DQPSK, companies working on commercial PDM technology include Alcatel-Lucent, Ciena, Cisco Systems, Huawei and Infinera. Polarization scrambler Wavelength-division multiplexing Orbital angular momentum multiplexing
11.
Spatial multiplexing
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Therefore, the space dimension is reused, or multiplexed, more than one time. If the transmitter is equipped with N t antennas and the receiver has N r antennas and this means that N s streams can be transmitted in parallel, ideally leading to an N s increase of the spectral efficiency. The practical multiplexing gain can be limited by spatial correlation, which means some of the parallel streams may have very weak channel gains. An often encountered problem in open loop spatial multiplexing is to guard against instance of high channel correlation, one such extension which is being considered for DVB-NGH systems is the so-called enhanced Spatial Multiplexing scheme. A closed-loop MIMO system utilizes Channel State Information at the transmitter, in most cases, only partial CSI is available at the transmitter because of the limitations of the feedback channel. A precoding matrix W is used to precode the symbols in the vector to enhance the performance, the column dimension N s of W can be selected smaller than N t which is useful if the system requires N s streams because of several reasons. Examples of the reasons are as follows, either the rank of the MIMO channel or the number of antennas is smaller than the number of transmit antennas. Space–time code Space–time trellis code WiMAX MIMO 3G MIMO
12.
Orbital angular momentum multiplexing
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Orbital angular momentum is one of two forms of angular momentum of light. OAM is distinct from, and should not be confused with, OAM multiplexing can access a potentially unbounded set of OAM quantum states, and thus offer a much larger number of channels, subject only to the constraints of real-world optics. OAM multiplexing was demonstrated using light beams in free space as early as 2004, since then, research into OAM has proceeded in two areas, radio frequency and optical transmission. An experiment in 2011 demonstrated OAM multiplexing of two incoherent radio signals over a distance of 442 m.5 metres and these results agree well with predictions about severely limited distances made by Edfors et al. OAM multiplexing is used in the optical domain, work is ongoing on applying OAM techniques to long-range practical free-space optical communication links. OAM multiplexing can not be implemented in the existing optical fiber systems, since these systems are based on single-mode fibers. Instead, few-mode or multi-mode fibers need to be used, because of this mode-instability, direct-detection OAM multiplexing has not yet been realized in long-haul communications. In 2012, transmission of OAM states with 97% purity after 20 meters over specialty fibers was demonstrated by researchers at Boston University, later experiments have shown stable propagation of these modes over distances of 50 meters, and further improvements of this distance are the subject of ongoing work. Et al. published in Science in 2013 claims the successful demonstration of an OAM multiplexed fiber optic transmission system over a 1.1 km test path. The test system was capable of using up to four different OAM channels simultaneously and they also demonstrated combined OAM and WDM using the same apparatus, but using only two OAM modes. Optical vortex Vorticity Angular momentum of light Polarization-division multiplexing Wavelength-division multiplexing Siae Microelettronica patent
13.
Statistical multiplexing
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Statistical multiplexing is a type of communication link sharing, very similar to dynamic bandwidth allocation. In statistical multiplexing, a channel is divided into an arbitrary number of variable bitrate digital channels or data streams. The link sharing is adapted to the traffic demands of the data streams that are transferred over each channel. This is an alternative to creating a fixed sharing of a link, such as in general time division multiplexing, when performed correctly, statistical multiplexing can provide a link utilization improvement, called the statistical multiplexing gain. Statistical multiplexing is facilitated through packet mode or packet-oriented communication, which among others is utilized in packet switched computer networks, each stream is divided into packets that normally are delivered asynchronously in a first-come first-served fashion. In alternative fashion, the packets may be delivered according to some scheduling discipline for fair queuing or differentiated and/or guaranteed quality of service, statistical multiplexing normally implies on-demand service rather than one that preallocates resources for each data stream. Statistical multiplexing schemes do not control user data transmissions, statistical multiplexing allows the bandwidth to be divided arbitrarily among a variable number of channels. Statistical multiplexing ensures that slots will not be wasted, the transmission capacity of the link will be shared by only those users who have packets. Static TDM and other circuit switching is carried out at the layer in the OSI model and TCP/IP model, while statistical multiplexing is carried out at the data link layer. In statistical multiplexing, each packet or frame contains a channel/data stream identification number, examples of statistical multiplexing are, The MPEG transport stream for digital TV transmission. Statistical multiplexing is used to allow several video, audio and data streams of different data rates to be transmitted over a bandwidth-limited channel, the channel number is denoted Program ID. The UDP and TCP protocols, where streams from several application processes are multiplexed together. The packets may have varying lengths, the port numbers constitute channel identification numbers. The X.25 and Frame relay packet-switching protocols, where the packets have varying lengths, and the channel number is denoted Virtual Connection Identifier. The international collection of X.25 providers, using the X.25 protocol suite was known as the Packet switched network in the 1980s. The Asynchronous Transfer Mode packet-switched protocol, where the packets have fixed length, the channel identification number consists of a Virtual Connection Identifier and a Virtual Path Identifier. The multiplexer allocates to each service the bandwidth required for its real-time needs so that services with complex scenes receive more bandwidth than services with complex ones. This bandwidth sharing technique produces the best video quality at the lowest possible aggregate bandwidth
14.
Packet switching
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Packet switching increases network efficiency and robustness, and enables technological convergence of many applications operating on the same network. Packets are composed of a header and payload, Information in the header is used by networking hardware to direct the packet to its destination where the payload is extracted and used by application software. This concept contrasted and contradicted then-established principles of pre-allocation of network bandwidth, the new concept found little resonance among network implementers until the independent work of British computer scientist Donald Davies at the National Physical Laboratory in the late 1960s. Packet mode communication may be implemented with or without intermediate forwarding nodes, in case of a shared physical medium, the packets may be delivered according to a multiple access scheme. In the late 1950s, the US Air Force established a wide area network for the Semi-Automatic Ground Environment radar defense system and they sought a system that might survive a nuclear attack to enable a response, thus diminishing the attractiveness of the first strike advantage by enemies. Report P-2626 described a general architecture for a large-scale, distributed, Barans work was known to Robert Taylor and J. C. R. Licklider at the Information Processing Technology Office, who advocated wide area networks, starting in 1965, Donald Davies at the National Physical Laboratory, UK, independently developed the same message routing methodology as developed by Baran. He called it packet switching, a more accessible name than Barans and he gave a talk on the proposal in 1966, after which a person from the Ministry of Defence told him about Barans work. A member of Davies team met Lawrence Roberts at the 1967 ACM Symposium on Operating System Principles, Davies had chosen some of the same parameters for his original network design as did Baran, such as a packet size of 1024 bits. In 1966, Davies proposed that a network should be built at the laboratory to serve the needs of NPL, the NPL Data Communications Network entered service in 1970. In 1974, Vint Cerf and Bob Kahn published the specifications for Transmission Control Protocol, Packet switching may be classified into connectionless packet switching, also known as datagram switching, and connection-oriented packet switching, also known as virtual circuit switching. Examples of connectionless protocols are Ethernet, Internet Protocol, and the User Datagram Protocol, connection-oriented protocols include X.25, Frame Relay, Multiprotocol Label Switching, and the Transmission Control Protocol. In connectionless mode each packet includes complete addressing information, the packets are routed individually, sometimes resulting in different paths and out-of-order delivery. Each packet is labeled with an address, source address. It may also be labeled with the number of the packet. At the destination, the original message/data is reassembled in the correct order, connection-oriented transmission requires a setup phase in each involved node before any packet is transferred to establish the parameters of communication. The packets include a connection identifier rather than address information and are negotiated between endpoints so that they are delivered in order and with error checking, the signaling protocols used allow the application to specify its requirements and discover link parameters. Acceptable values for service parameters may be negotiated, routing a packet requires the node to look up the connection id in a table
15.
Time-division multiple access
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Time-division multiple access is a channel access method for shared-medium networks. It allows several users to share the same channel by dividing the signal into different time slots. The users transmit in rapid succession, one after the other and this allows multiple stations to share the same transmission medium while using only a part of its channel capacity. It is also used extensively in satellite systems, combat-net radio systems, for usage of Dynamic TDMA packet mode communication, see below. TDMA is a type of multiplexing, with the special point that instead of having one transmitter connected to one receiver. GSM, D-AMPS, PDC, iDEN, and PHS are examples of TDMA cellular systems, GSM combines TDMA with Frequency Hopping and wideband transmission to minimize common types of interference. In the GSM system, the synchronization of the phones is achieved by sending timing advance commands from the base station which instructs the mobile phone to transmit earlier. This compensates for the delay resulting from the light speed velocity of radio waves. The mobile phone is not allowed to transmit for its time slot. As the transmission moves into the period, the mobile network adjusts the timing advance to synchronize the transmission. Initial synchronization of a phone requires even more care, before a mobile transmits there is no way to actually know the offset required. For this reason, a time slot has to be dedicated to mobiles attempting to contact the network. The mobile attempts to broadcast at the beginning of the time slot, if the mobile is located next to the base station, there will be no time delay and this will succeed. If, however, the phone is at just less than 35 km from the base station. In that case, the mobile will be instructed to broadcast its messages starting nearly a whole time slot earlier than would be expected otherwise. Finally, if the mobile is beyond the 35 km cell range in GSM, then the RACH will arrive in a time slot. It is this feature, rather than limitations of power, that limits the range of a GSM cell to 35 km when no special extension techniques are used, in G. hn, a master device allocates Contention-Free Transmission Opportunities to other slave devices in the network. Only one device can use a CFTXOP at a time, thus avoiding collisions, flexRay protocol which is also a wired network used for safety-critical communication in modern cars, uses the TDMA method for data transmission control
16.
Frequency-hopping spread spectrum
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It is used as a multiple access method in the code division multiple access scheme frequency-hopping code division multiple access. FHSS is a technology that spreads its signal over rapidly changing frequencies. Each available frequency band is divided into sub-frequencies, signals rapidly change among these in a pre-determined order. Interference at a frequency will only affect the signal during that short interval. FHSS can, however, cause interference with adjacent direct-sequence spread spectrum systems, a sub-type of FHSS used in Bluetooth wireless data transfer is adaptive frequency hopping spread spectrum. A spread-spectrum transmission offers three main advantages over a transmission, Spread-spectrum signals are highly resistant to narrowband interference. The process of re-collecting a spread signal spreads out the interfering signal, Spread-spectrum signals are difficult to intercept. A spread-spectrum signal may simply appear as an increase in the noise to a narrowband receiver. An eavesdropper may have difficulty intercepting a transmission in real time if the sequence is not known. Spread-spectrum transmissions can share a band with many types of conventional transmissions with minimal interference. The spread-spectrum signals add minimal noise to the communications. As a result, bandwidth can be used more efficiently, Spread-spectrum signals are highly resistant to deliberate jamming, unless the adversary has knowledge of the spreading characteristics. Military radios use cryptographic techniques to generate the sequence under the control of a secret Transmission Security Key that the sender and receiver share in advance. By itself, frequency hopping provides only limited protection against eavesdropping and jamming, most modern military frequency hopping radios also employ separate encryption devices such as the KY-57. U. S. military radios that use frequency hopping include the JTIDS/MIDS family, HAVE QUICK, some walkie-talkies that employ frequency-hopping spread spectrum technology have been developed for unlicensed use on the 900 MHz band. Several such radios were marketed under the name eXtreme Radio Service, despite the names similarity to the FRS allocation, the system is a proprietary design, rather than an official FCC allocated service. Motorola has deployed a business-banded, license-free digital radio that uses FHSS technology, the overall bandwidth required for frequency hopping is much wider than that required to transmit the same information using only one carrier frequency. However, because transmission occurs only on a portion of this bandwidth at any given time
17.
Direct-sequence spread spectrum
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In telecommunications, direct-sequence spread spectrum is a spread spectrum modulation technique used to reduce overall signal interference. The spreading of this makes the resulting wideband channel more noisy. A method of achieving the spreading of a signal is provided by the modulation scheme. This modulation of the message signal scrambles and spreads the pieces of data, in this context, the duration of the radio pulse for the PN code is referred to as the chip duration. The smaller this duration, the larger the bandwidth of the resulting DSSS signal, some practical and effective uses of DSSS include the Code Division Multiple Access channel access method and the IEEE802. 11b specification used in Wi-Fi networks. DSSS phase-shifts a sine wave pseudorandomly with a string of pseudonoise code symbols called chips. That is, each bit is modulated by a sequence of much faster chips. Therefore, the rate is much higher than the information signal bit rate. DSSS uses a structure in which the sequence of chips produced by the transmitter is already known by the receiver. The receiver can use the same PN sequence to counteract the effect of the PN sequence on the received signal in order to reconstruct the information signal. Direct-sequence spread-spectrum transmissions multiply the data being transmitted by a noise signal and this noise signal is a pseudorandom sequence of 1 and −1 values, at a frequency much higher than that of the original signal. The resulting signal resembles white noise, like a recording of static. However, this signal is used to exactly reconstruct the original data at the receiving end. This process, known as de-spreading, mathematically constitutes a correlation of the transmitted PN sequence with the PN sequence that the receiver knows the transmitter is using. The resulting effect of enhancing signal to noise ratio on the channel is called process gain and this effect can be made larger by employing a longer PN sequence and more chips per bit, but physical devices used to generate the PN sequence impose practical limits on attainable processing gain. While for useful process gain the transmitted DSSS signal must occupy much wider bandwidth than simple a. m, if an undesired transmitter transmits on the same channel but with a different PN sequence, the de-spreading process has reduced processing gain for that signal. It is the most widely used type of CDMA, cordless phones operating in the 900 MHz,2.4 GHz and 5.8 GHz bands IEEE802. 11b 2.4 GHz Wi-Fi, and its predecessor 802. 11-1999. NTIA Manual of Regulations and Procedures for Federal Radio Frequency Management Civil Spread Spectrum History
18.
OFDMA
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Orthogonal frequency-division multiple access is a multi-user version of the popular orthogonal frequency-division multiplexing digital modulation scheme. Multiple access is achieved in OFDMA by assigning subsets of subcarriers to individual users and this allows simultaneous low-data-rate transmission from several users. The advantages and disadvantages summarized below are further discussed in the Characteristics, see also the list of OFDM key features. Allows simultaneous low-data-rate transmission from several users, lower maximal transmission power for low-data-rate users. Further improves OFDM robustness to fading and interference, flexibility of deployment across various frequency bands with little needed modification to the air interface. Averaging interferences from neighboring cells, by using different basic carrier permutations between users in different cells, interferences within the cell are averaged by using allocation with cyclic permutations. Enables single-frequency network coverage, where coverage problem exists and gives excellent coverage, offers frequency diversity by spreading the carriers all over the used spectrum. Higher sensitivity to frequency offsets and phase noise, asynchronous data communication services such as web access are characterised by short communication bursts at high data rate. Few users in a base station cell are transferring data simultaneously at low constant data rate, the OFDM diversity gain and resistance to frequency-selective fading may partly be lost if very few sub-carriers are assigned to each user, and if the same carrier is used in every OFDM symbol. Adaptive sub-carrier assignment based on fast feedback information about the channel, dealing with co-channel interference from nearby cells is more complex in OFDM than in CDMA. It would require dynamic channel allocation with advanced coordination among adjacent base stations, the fast channel feedback information and adaptive sub-carrier assignment is more complex than CDMA fast power control. Based on feedback information about the conditions, adaptive user-to-subcarrier assignment can be achieved. OFDMA can be seen as an alternative to combining OFDM with time-division multiple access or time-domain statistical multiplexing communication, low-data-rate users can send continuously with low transmission power instead of using a pulsed high-power carrier. Constant delay, and shorter delay, can be achieved, in spectrum sensing cognitive radio, OFDMA is a possible approach to filling free radio frequency bands adaptively. Timo A. Weiss and Friedrich K. Jondral of the University of Karlsruhe proposed a pooling system in which free bands sensed by nodes were immediately filled by OFDMA subbands. The radio interface was formerly named High Speed OFDM Packet Access, the Qualcomm Flarion Technologies Mobile Flash-OFDM the now defunct Qualcomm/3GPP2 Ultra Mobile Broadband project, intended as a successor of CDMA2000, but replaced by LTE. OFDMA is also an access method for the IEEE802.22 Wireless Regional Area Networks. The project aims at designing the first cognitive radio based standard operating in the VHF-low UHF spectrum, the term OFDMA is claimed to be a registered trademark by Runcom Technologies Ltd. with various other claimants to the underlying technologies through patents
19.
Media access control
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In the IEEE802 reference model of computer networking, the medium access control or media access control layer is the lower sublayer of the data link layer of the seven-layer OSI model. g. The hardware that implements the MAC is referred to as a media access controller, the MAC sublayer acts as an interface between the logical link control sublayer and the networks physical layer. The MAC layer emulates a full-duplex logical communication channel in a multi-point network and this channel may provide unicast, multicast or broadcast communication service. According to IEEE Std 802-2001 section 6.2. 3-2002 section 4.1, a MAC address is intended as a unique serial number. MAC addresses are assigned to network interface hardware at the time of manufacture. The most significant part of the address identifies the manufacturer, who assigns the remainder of the address, thus provide a potentially unique address. This makes it possible for frames to be delivered on a link that interconnects hosts by some combination of repeaters, hubs, bridges and switches. Examples of physical networks are Ethernet networks and Wi-Fi networks, both of which are IEEE802 networks and use IEEE802 48-bit MAC addresses, a MAC layer is not required in full-duplex point-to-point communication, but address fields are included in some point-to-point protocols for compatibility reasons. The channel access control mechanisms provided by the MAC layer are known as a multiple access protocol. This makes it possible for several stations connected to the physical medium to share it. Examples of shared physical media are bus networks, ring networks, hub networks, wireless networks, the channel access control mechanism relies on a physical layer multiplex scheme. The most widespread multiple access protocol is the contention based CSMA/CD protocol used in Ethernet networks and this mechanism is only utilized within a network collision domain, for example an Ethernet bus network or a hub-based star topology network. An Ethernet network may be divided into several domains, interconnected by bridges and switches. A multiple access protocol is not required in a switched network, such as todays switched Ethernet networks. The MAC protocol in cellular networks is designed to maximize the utilization of the expensive licensed spectrum, the air interface of a cellular network is at layers 1 and 2 of the OSI model, at layer 2, it is divided into multiple protocol layers. In UMTS and LTE, those protocols are the Packet Data Convergence Protocol, the Radio Link Control protocol, the base station has the absolute control over the air interface and schedules the downlink access as well as the uplink access of all devices. The MAC protocol is specified by 3GPP in TS25.321 for UMTS and TS36.321 for LTE.3 or later
20.
Telecommunications
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Telecommunication is the transmission of signs, signals, messages, writings, images and sounds or intelligence of any nature by wire, radio, optical or other electromagnetic systems. Telecommunication occurs when the exchange of information between communication participants includes the use of technology and it is transmitted either electrically over physical media, such as cables, or via electromagnetic radiation. Such transmission paths are divided into communication channels which afford the advantages of multiplexing. The term is used in its plural form, telecommunications. Early means of communicating over a distance included visual signals, such as beacons, smoke signals, semaphore telegraphs, signal flags, other examples of pre-modern long-distance communication included audio messages such as coded drumbeats, lung-blown horns, and loud whistles. Zworykin, John Logie Baird and Philo Farnsworth, the word telecommunication is a compound of the Greek prefix tele, meaning distant, far off, or afar, and the Latin communicare, meaning to share. Its modern use is adapted from the French, because its use was recorded in 1904 by the French engineer. Communication was first used as an English word in the late 14th century, in the Middle Ages, chains of beacons were commonly used on hilltops as a means of relaying a signal. Beacon chains suffered the drawback that they could pass a single bit of information. One notable instance of their use was during the Spanish Armada, in 1792, Claude Chappe, a French engineer, built the first fixed visual telegraphy system between Lille and Paris. However semaphore suffered from the need for skilled operators and expensive towers at intervals of ten to thirty kilometres, as a result of competition from the electrical telegraph, the last commercial line was abandoned in 1880. Homing pigeons have occasionally used throughout history by different cultures. Pigeon post is thought to have Persians roots and was used by the Romans to aid their military, frontinus said that Julius Caesar used pigeons as messengers in his conquest of Gaul. The Greeks also conveyed the names of the victors at the Olympic Games to various cities using homing pigeons, in the early 19th century, the Dutch government used the system in Java and Sumatra. And in 1849, Paul Julius Reuter started a service to fly stock prices between Aachen and Brussels, a service that operated for a year until the gap in the telegraph link was closed. Sir Charles Wheatstone and Sir William Fothergill Cooke invented the telegraph in 1837. Also, the first commercial electrical telegraph is purported to have constructed by Wheatstone and Cooke. Both inventors viewed their device as an improvement to the electromagnetic telegraph not as a new device, samuel Morse independently developed a version of the electrical telegraph that he unsuccessfully demonstrated on 2 September 1837
21.
Telegraphy
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Telegraphy is the long-distance transmission of textual or symbolic messages without the physical exchange of an object bearing the message. Thus semaphore is a method of telegraphy, whereas pigeon post is not, telegraphy requires that the method used for encoding the message be known to both sender and receiver. Such methods are designed according to the limits of the medium used. The use of signals, beacons, reflected light signals. In the 19th century, the harnessing of electricity led to the invention of electrical telegraphy, the advent of radio in the early 20th century brought about radiotelegraphy and other forms of wireless telegraphy. The word telegraph was first coined by the French inventor of the Semaphore line, Claude Chappe, a telegraph is a device for transmitting and receiving messages over long distances, i. e. for telegraphy. The word telegraph alone now generally refers to an electrical telegraph, Wireless telegraphy is also known as CW, for continuous wave, as opposed to the earlier radio technique of using a spark gap. Contrary to the definition used by Chappe, Morse argued that the term telegraph can strictly be applied only to systems that transmit. This is to be distinguished from semaphore, which transmits messages. Smoke signals, for instance, are to be considered semaphore, according to Morse, telegraph dates only from 1832 when Pavel Schilling invented one of the earliest electrical telegraphs. A telegraph message sent by a telegraph operator or telegrapher using Morse code was known as a telegram. A cablegram was a sent by a submarine telegraph cable. Later, a Telex was a sent by a Telex network. A wire picture or wire photo was a picture that was sent from a remote location by a facsimile telegraph. A diplomatic telegram, also known as a cable, is the term given to a confidential communication between a diplomatic mission and the foreign ministry of its parent country. These continue to be called telegrams or cables regardless of the used for transmission. Commercial electrical telegraphs were introduced from 1837, the first telegraphs came in the form of optical telegraph, including the use of smoke signals, beacons, or reflected light, which have existed since ancient times. Early proposals for a telegraph system were made to the Royal Society by Robert Hooke in 1684 and were first implemented on an experimental level by Sir Richard Lovell Edgeworth in 1767
22.
Electrical telegraph
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An electrical telegraph is a telegraph that uses electrical signals, usually conveyed via dedicated telecommunication lines or radio. From early studies of electricity, electrical phenomena were known to travel with great speed, in 1753 an anonymous writer in the Scots Magazine suggested an electrostatic telegraph. In 1774 Georges-Louis Le Sage realised an early electric telegraph, the telegraph had a separate wire for each of the 26 letters of the alphabet and its range was only between two rooms of his home. In 1800 Alessandro Volta invented the voltaic pile, allowing for a current of electricity for experimentation. Both their designs employed multiple wires to represent almost all Latin letters, thus, messages could be conveyed electrically up to a few kilometers, with each of the telegraph receivers wires immersed in a separate glass tube of acid. The telegraph receivers operator would watch the bubbles and could record the transmitted message. This is in contrast to later telegraphs that used a single wire, hans Christian Ørsted discovered in 1820 that an electric current produces a magnetic field which will deflect a compass needle. In the same year Johann Schweigger invented the galvanometer, with a coil of wire around a compass, in 1824, Peter Barlow said that such a system only worked to a distance of about 200 feet, and so was impractical. In 1825 William Sturgeon invented the electromagnet, with a winding of uninsulated wire on a piece of varnished iron. During his tenure at The Albany Academy from 1826 to 1832, in 1835 Joseph Henry and Edward Davy invented the critical electrical relay. Davys relay used a needle which dipped into a mercury contact when an electric current passed through the surrounding coil. This allowed a weak current to switch a larger current to operate a powerful local electromagnet over very long distances, Davy demonstrated his telegraph system in Regents Park in 1837 and was granted a patent on 4 July 1838. He also developed an electric relay, the first working telegraph was built by the English inventor Francis Ronalds in 1816 and used static electricity. At the family home on Hammersmith Mall, he set up a subterranean system in a 175 yard long trench as well as an eight mile long overhead telegraph. The lines were connected at both ends to revolving dials marked with the letters of the alphabet and electrical impulses sent along the wire were used to transmit messages, offering his invention to the Admiralty in July 1816, it was rejected as “wholly unnecessary”. Elements of Ronalds’ design were utilised in the subsequent commercialisation of the telegraph over 20 years later, the telegraph invented by Baron Schilling von Canstatt in 1832 had a transmitting device which consisted of a keyboard with 16 black-and-white keys. These served for switching the electric current, the receiving instrument consisted of six galvanometers with magnetic needles, suspended from silk threads. Both stations of Shillings telegraph were connected by eight wires, six were connected with the galvanometers, one served for the return current, when at the starting station the operator pressed a key, the corresponding pointer was deflected at the receiving station
23.
Hughes telegraph
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David Edward Hughes, was a British-American inventor, practical experimenter, and professor of music known for his work on the printing telegraph and the microphone. He is generally considered to have born in London but his family moved around that time so he may have been born in Corwen. His family moved to the U. S. while he was a child, in 1855 he patented a printing telegraph. He moved back to London in 1857 and further pursued experimentation and invention, in 1879 he identified what seemed to be a new phenomenon during his experiments, sparking in one device could be heard in a separate portable microphone apparatus he had set up. Hughes was born in 1831, the son of a talented family hailing originally from Y Bala. At only six years old, he is known to have played the harp, Hughes also worked as a practical experimenter, coming up with the printing telegraph in 1855. He moved back to London in 1857 to sell his invention and he worked on microphones and on the invention of the induction balance. Despite Hughes facility as an experimenter, he had little mathematical training and he was a friend of William Henry Preece. In 1855, Hughes designed a printing telegraph system, in Europe, the Hughes Telegraph System became an international standard. In 1878 Hughes published his work on the effects of sound on the electronic sound pickups, called transmitters. Based on its ability to pick up extremely weak sounds, Hughes referred to it as a microphone effect and he conducted a simple demonstration of this principle of loose contact by laying an iron nail across two other nails connected to a battery and galvanometer. His paper was read before the Royal Society of London by Thomas Henry Huxley on May 8,1878, Hughes published his work during the time that Thomas Edison was working on a carbon telephone transmitter and Emile Berliner was working on a loose-contact transmitter. Both Hughes and Edison may have based their work on Philipp Reis telephone work, Hughes seems to have come across the phenomenon of radio waves nine years before they were proven to exist by Heinrich Hertz in 1888. In 1879 while working in London Hughes discovered that a bad contact in a Bell telephone he was using in his experiments seemed to be sparking when he worked on a nearby induction balance. By trial and error experiments he found he could pick up these aerial waves as he carried his telephone device down the street out to a range of 500 yards. Stokes was convinced the phenomenon Hughes was demonstrating was merely electromagnetic induction, Hughes was not a physicist and seems to have accepted Stokes observations and did not pursue the experiments any further. The world will be right in placing his name on the highest pinnacle, in the same publication Elihu Thomson put forward a claim that Hughes was really the first to transmit radio. The carbon rod and two blocks, which he would refer to as a coherer in 1899 is also similar to devices known as crystal radio detectors
24.
Telephone exchange
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A telephone exchange is a telecommunications system used in the public switched telephone network or in large enterprises. In historical perspective, telecommunication terms have been used with different semantics over time, the term telephone exchange is often used synonymously with central office, a Bell System term. Often, an office is defined as a building used to house the inside plant equipment of potentially several telephone exchanges. Such an area has also referred to as the exchange. Central office locations may also be identified in North America as wire centers, All central offices within a larger region, typically aggregated by state, were assigned a common numbering plan area code. For corporate or enterprise use, a telephone exchange is often referred to as a private branch exchange. Smaller installations might deploy a PBX or key telephone system in the office of a receptionist and this made it possible for subscribers to call each other at homes, businesses, or public spaces. These made telephony an available and comfortable communication tool for everyday use, one of the first to propose a telephone exchange was Hungarian Tivadar Puskás in 1877 while he was working for Thomas Edison. The first experimental telephone exchange was based on the ideas of Puskás, the worlds first commercial telephone exchange opened on November 12,1877 in Friedrichsberg close to Berlin. George W. Coy designed and built the first commercial US telephone exchange opened in New Haven. The switchboard was built from carriage bolts, handles from teapot lids and bustle wire, Charles Glidden is also credited with establishing an exchange in Lowell, MA. with 50 subscribers in 1878. In Europe other early telephone exchanges were based in London and Manchester, Belgium had its first International Bell exchange a year later. In 1887 Puskás introduced the multiplex switchboard, later exchanges consisted of one to several hundred plug boards staffed by switchboard operators. Each operator sat in front of a panel containing banks of ¼-inch tip-ring-sleeve jacks. In front of the jack panel lay a horizontal panel containing two rows of patch cords, each connected to a cord circuit. When a calling party lifted the receiver, the loop current lit a signal lamp near the jack. The operator responded by inserting the rear cord into the jack and switched her headset into the circuit to ask, Number. For a local call, the operator inserted the front cord of the pair into the partys local jack
25.
Pulse-amplitude modulation
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Pulse-amplitude modulation, is a form of signal modulation where the message information is encoded in the amplitude of a series of signal pulse. It is an analog pulse modulation scheme in which the amplitudes of a train of pulses are varied according to the sample value of the message signal. Demodulation is performed by detecting the level of the carrier at every single period. There are two types of pulse amplitude modulation, Single polarity PAM, In this a suitable fixed DC bias is added to the signal to ensure all the pulses are positive. Double polarity PAM, In this the pulses are positive and negative. In particular, all telephone modems faster than 300 bit/s use quadrature amplitude modulation, the number of possible pulse amplitudes in analog PAM is theoretically infinite. Digital PAM reduces the number of pulse amplitudes to some power of two, some versions of the Ethernet communication standard are an example of PAM usage. Pulse-amplitude modulation has also developed for the control of light-emitting diodes. Pulse-amplitude modulation LED drivers are able to synchronize pulses across multiple LED channels to enable perfect colour matching. Due to the inherent nature of PAM in conjunction with the switching speed of LEDs it is possible to use LED lighting as a means of wireless data transmission at high speed. The North American Advanced Television Systems Committee standards for digital television uses a form of PAM to broadcast the data that makes up the television signal. This system, known as 8VSB, is based on a three-level PAM like 100BASE-TX, using a single 6 MHz channel allocation, as defined in the previous NTSC analog standard, 8VSB is capable of transmitting 32 Mbit/s. After accounting for error correcting codes and other overhead, the rate in the signal is 19.39 Mbit/s
26.
Pulse-code modulation
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Pulse-code modulation is a method used to digitally represent sampled analog signals. It is the form of digital audio in computers, compact discs, digital telephony. In a PCM stream, the amplitude of the signal is sampled regularly at uniform intervals. Linear pulse-code modulation is a type of PCM where the quantization levels are linearly uniform. This is in contrast to PCM encodings where quantization levels vary as a function of amplitude, though PCM is a more general term, it is often used to describe data encoded as LPCM. Early electrical communications started to sample signals in order to interlace samples from multiple telegraphy sources, the American inventor Moses G. Farmer conveyed telegraph time-division multiplexing as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals and he obtained intelligible speech from channels sampled at a rate above 3500–4300 Hz, lower rates proved unsatisfactory. This was TDM, but pulse-amplitude modulation rather than PCM, in 1920 the Bartlane cable picture transmission system, named after its inventors Harry G. Bartholomew and Maynard D. In 1926, Paul M. Rainey of Western Electric patented a machine which transmitted its signal using 5-bit PCM. The machine did not go into production, british engineer Alec Reeves, unaware of previous work, conceived the use of PCM for voice communication in 1937 while working for International Telephone and Telegraph in France. He described the theory and advantages, but no practical application resulted, Reeves filed for a French patent in 1938, and his US patent was granted in 1943. By this time Reeves had started working at the Telecommunications Research Establishment, the first transmission of speech by digital techniques, the SIGSALY encryption equipment, conveyed high-level Allied communications during World War II. In 1943 the Bell Labs researchers who designed the SIGSALY system became aware of the use of PCM binary coding as already proposed by Alec Reeves. In 1949 for the Canadian Navys DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances, PCM in the late 1940s and early 1950s used a cathode-ray coding tube with a plate electrode having encoding perforations. The plate collected or passed the beam, producing current variations in binary code, rather than natural binary, the grid of Goodalls later tube was perforated to produce a glitch-free Gray code, and produced all bits simultaneously by using a fan beam instead of a scanning beam. Patent 2,801,281 filed in 1946 and 1952, another patent by the same title was filed by John R. Pierce in 1945, and issued in 1948, U. S. The three of them published The Philosophy of PCM in 1948, PCM is the method of encoding generally used for uncompressed audio, although there are other methods such as pulse-density modulation. The 4ESS switch introduced time-division switching into the US telephone system in 1976, LPCM is used for the lossless encoding of audio data in the Compact disc Red Book standard, introduced in 1982
27.
T-carrier
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The T-carrier is a member of the series of carrier systems developed by AT&T Bell Laboratories for digital transmission of multiplexed telephone calls. The first version, the Transmission System 1, was introduced in 1962 in the Bell System, subsequent specifications carried multiples of the basic T1 data rates, such as T2 with 96 channels, T3 with 672 channels, and others. T-1 is a specification for telecommunications trunking. A trunk is a transmission channel between two points on the network, each point is either a switching center or a node. Initially, T-1 trunks were used only to connect major telephone exchanges, one transmit pair, one receive pair. If the exchanges were too far apart, a repeater boosted the signal, before the digital T-1 system, carrier wave systems such as 12-channel carrier systems worked by frequency division multiplexing, each call was an analog signal. A T-1 trunk could transmit 24 telephone calls at a time, dS-1 is a communications protocol for multiplexing the bitstreams of up to 24 telephone calls, along with two special bits, a framing bit and a maintenance-signaling bit. T-1s maximum data rate is 1.544 megabits per second. Throughout Europe and most of the rest of the world there is a transmission system called E-carrier. Existing frequency-division multiplexing carrier systems worked well for connections between distant cities, but required expensive modulators, demodulators and filters for every voice channel, for connections within metropolitan areas, Bell Labs in the late 1950s sought cheaper terminal equipment. Pulse-code modulation allowed sharing a coder and decoder among several voice trunks, the most common legacy of this system is the line rate speeds. T1 now means any data circuit that runs at the original 1.544 Mbit/s line rate, the T2 and T3 circuit channels carry multiple T1 channels multiplexed, resulting in transmission rates of 6.312 and 44.736 Mbit/s, respectively. A T3 line comprises 28 T1 lines, each operating at total signaling rate of 1.544 Mbit/s. It is possible to get a fractional T3 line, meaning a T3 line with some of the 28 lines turned off, resulting in a transfer rate. Supposedly, the 1.544 Mbit/s rate was chosen because tests done by AT&T Long Lines in Chicago were conducted underground, the optimum bit rate was chosen empirically—the capacity was increased until the failure rate was unacceptable, then reduced to leave a margin. Companding allowed acceptable audio performance with seven bits per PCM sample in this original T1/D1 system. The standard does not allow an all zero sample which would produce a string of binary zeros. However, when carrying data there could be long strings of zeros, a more detailed understanding of how the rate of 1.544 Mbit/s was divided into channels is as follows
28.
E-carrier
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The E-carrier is a member of the series of carrier systems developed for digital transmission of many simultaneous telephone calls by time-division multiplexing. It was widely adopted in almost all countries outside the US, Canada, E-carrier deployments have steadily been replaced by Ethernet as telecommunication networks transitions towards all IP. An E1 link operates two separate sets of wires, usually Unshielded twisted pair or using coaxial. A nominal 3 volt peak signal is encoded with pulses using a method avoiding long periods without polarity changes, the line data rate is 2.048 Mbit/s which is split into 32 timeslots, each being allocated 8 bits in turn. Thus each timeslot sends and receives an 8-bit PCM sample, usually encoded according to A-law algorithm,8,000 times per second and this is ideal for voice telephone calls where the voice is sampled at that data rate and reconstructed at the other end. The timeslots are numbered from 0 to 31, the E1 frame defines a cyclical set of 32 time slots of 8 bits. The time slot 0 is devoted to management and time slot 16 for signaling. The main characteristics of the 2-Mbit/s frame are described in the following, one timeslot is reserved for framing purposes, and alternately transmits a fixed pattern. This allows the receiver to lock onto the start of each frame, the standards allow for a full cyclic redundancy check to be performed across all bits transmitted in each frame, to detect if the circuit is losing bits, but this is not always used. An alarm signal may also be transmitted using timeslot TS0, finally, some bits are reserved for national use. One timeslot is reserved for signalling purposes, to control call setup. This includes channel-associated signaling where a set of bits is used to opening and closing the circuit. In an E1 channel, communication consists of sending consecutive frames from the transmitter to the receiver and this way, the bytes received in each slot are assigned to the correct channel. A synchronization process is established, and it is known as frame alignment. In order to implement the frame alignment system so that the receiver of the frame can tell where it begins, in the 2 Mbit/s frame system, the FAS is a combination of seven fixed bits transmitted in the first time slot in the frame. For the alignment mechanism to be maintained, the FAS does not need to be transmitted in every frame, instead, this signal can be sent in alternate frames. In this case, TS0 is used as the synchronization slot, the TS0 of the rest of the frames is therefore available for other functions, such as the transmission of the alarms. In the TS0 of frames with FAS, the first bit is dedicated to carrying the cyclic redundancy checksum and it tells us whether there are one or more bit errors in a specific group of data received in the previous block of eight frames known as submultiframe
29.
Synchronous optical networking
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At low transmission rates data can also be transferred via an electrical interface. SONET generic criteria are detailed in Telcordia Technologies Generic Requirements document GR-253-CORE, Generic criteria applicable to SONET and other transmission systems are found in Telcordia GR-499-CORE. The primary difficulty in doing this prior to SONET/SDH was that the sources of these various circuits were different. This meant that circuit was actually operating at a slightly different rate. SONET/SDH allowed for the transport of many different circuits of differing origin within a single framing protocol. SONET/SDH is not a protocol in itself, but a transport protocol. It quickly evolved mapping structures and concatenated payload containers to transport ATM connections, both SDH and SONET are widely used today, SONET in the United States and Canada, and SDH in the rest of the world. Although the SONET standards were developed before SDH, it is considered a variation of SDH because of SDHs greater worldwide market penetration, SONET is subdivided into four sublayer with some factor such as the path, line, section and physical layer. The SDH standard was defined by the European Telecommunications Standards Institute, and is formalized as International Telecommunication Union standards G.707, G.783, G.784. The SONET standard was defined by Telcordia and American National Standards Institute standard T1.105, which define the set of transmission formats and transmission rates in the range above 51.840 Mbit/s. SDH differs from Plesiochronous Digital Hierarchy in that the rates that are used to transport the data on SONET/SDH are tightly synchronized across the entire network. This synchronization system allows entire inter-country networks to operate synchronously, greatly reducing the amount of buffering required between elements in the network. Therefore, it is inaccurate to think of SDH or SONET as communications protocols in and of themselves, they are generic, the basic format of a SONET/SDH signal allows it to carry many different services in its virtual container, because it is bandwidth-flexible. SONET and SDH often use different terms to describe identical features or functions and this can cause confusion and exaggerate their differences. With a few exceptions, SDH can be thought of as a superset of SONET, SONET is a set of transport containers that allow for delivery of a variety of protocols, including traditional telephony, ATM, Ethernet, and TCP/IP traffic. SONET therefore is not in itself a native communications protocol and should not be confused as being necessarily connection-oriented in the way that term is usually used, the protocol is a heavily multiplexed structure, with the header interleaved between the data in a complex way. This permits the data to have its own frame rate and be able to float around relative to the SDH/SONET frame structure. This interleaving permits a low latency for the encapsulated data
30.
Integrated Services Digital Network
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It was first defined in 1988 in the CCITT red book. Prior to ISDN, the system was viewed as a way to transport voice. The key feature of ISDN is that it integrates speech and data on the same lines, the ISDN standards define several kinds of access interfaces, such as Basic Rate Interface, Primary Rate Interface, Narrowband ISDN, and Broadband ISDN. It offers circuit-switched connections, and packet-switched connections, in increments of 64 kilobit/s, in some countries, ISDN found major market application for Internet access, in which ISDN typically provides a maximum of 128 kbit/s bandwidth in both upstream and downstream directions. Channel bonding can achieve a data rate, typically the ISDN B-channels of three or four BRIs are bonded. ISDN is employed as the network, data-link and physical layers in the context of the OSI model, or could be considered a suite of digital services existing on layers 1,2, and 3 of the OSI model. In a videoconference, ISDN provides simultaneous voice, video, Integrated services refers to ISDNs ability to deliver at minimum two simultaneous connections, in any combination of data, voice, video, and fax, over a single line. Multiple devices can be attached to the line, and used as needed and that means an ISDN line can take care of most peoples complete communications needs at a much higher transmission rate, without forcing the purchase of multiple analog phone lines. It also refers to integrated switching and transmission in that telephone switching, the entry level interface to ISDN is the Basic Rate Interface, a 128 kbit/s service delivered over a pair of standard telephone copper wires. The 144 kbit/s payload rate is broken down into two 64 kbit/s bearer channels and one 16 kbit/s signaling channel and this is sometimes referred to as 2B+D. The T interface is an interface between a computing device and a terminal adapter, which is the digital equivalent of a modem. The S interface is a bus that ISDN consumer devices plug into. The R interface defines the point between a device and a terminal adapter which provides translation to and from such a device. BRI-ISDN is very popular in Europe but is less common in North America. It is also common in Japan — where it is known as INS64, the other ISDN access available is the Primary Rate Interface, which is carried over an E1 in most parts of the world. An E1 is 30 B channels of 64 kbit/s, one D channel of 64 kbit/s and this is often referred to as 30B+2D. In North America PRI service is delivered on one or more T1 carriers of 1544 kbit/s, a PRI has 23 B channels and 1 D channel for signalling. Inter-changeably but incorrectly, a PRI is referred to as T1 because it uses the T1 carrier format, a true T1 uses 24 channels of 64 kbit/s of in-band signaling
31.
Global system for mobile communications
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As of 2014 it has become the de facto global standard for mobile communications – with over 90% market share, operating in over 219 countries and territories. This expanded over time to include communications, first by circuit-switched transport, then by packet data transport via GPRS. Subsequently, the 3GPP developed third-generation UMTS standards followed by fourth-generation LTE Advanced standards, GSM is a trademark owned by the GSM Association. It may also refer to the most common voice codec used, the decision to develop a continental standard eventually resulted in a unified, open, standard-based network which was larger than that in the United States. In February 1987 Europe produced the very first agreed GSM Technical Specification, the MoU drew-in mobile operators from across Europe to pledge to invest in new GSM networks to an ambitious common date. In 1989, the Groupe Spécial Mobile committee was transferred from CEPT to the European Telecommunications Standards Institute, in parallel, France and Germany signed a joint development agreement in 1984 and were joined by Italy and the UK in 1986. In 1986 the European Commission proposed reserving the 900 MHz spectrum band for GSM, the former Finnish prime minister Harri Holkeri made the worlds first GSM call on July 1,1991, calling Kaarina Suonio using a network built by Telenokia and Siemens and operated by Radiolinja. In the following year,1992, saw the sending of the first short messaging service message, work began in 1991 to expand the GSM standard to the 1800 MHz frequency band and the first 1800 MHz network became operational in the UK by 1993 called and DCS1800. Also that year, Telecom Australia became the first network operator to deploy a GSM network outside Europe and the first practical hand-held GSM mobile phone became available. In 1995, fax, data and SMS messaging services were launched commercially, in the same year, the GSM Association formed. Pre-paid GSM SIM cards were launched in 1996 and worldwide GSM subscribers passed 100 million in 1998, in 2000 the first commercial GPRS services were launched and the first GPRS-compatible handsets became available for sale. In 2001 the first UMTS network was launched, a 3G technology that is not part of GSM, worldwide GSM subscribers exceeded 500 million. In 2002 the first Multimedia Messaging Service were introduced and the first GSM network in the 800 MHz frequency band became operational, EDGE services first became operational in a network in 2003 and the number of worldwide GSM subscribers exceeded 1 billion in 2004. By 2005, GSM networks accounted for more than 75% of the cellular network market. In 2005 the first HSDPA-capable network also became operational, the first HSUPA network launched in 2007. Worldwide GSM subscribers exceeded three billion in 2008, GSM is a second-generation standard employing time-division multiple-Access spectrum-sharing, issued by the European Telecommunications Standards Institute. GSM, for the first time, set a standard for Europe for wireless networks. It was also adopted by many countries outside Europe and this allowed subscribers to use other GSM networks that have roaming agreements with each other
32.
Synchronous Digital Hierarchy
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At low transmission rates data can also be transferred via an electrical interface. SONET generic criteria are detailed in Telcordia Technologies Generic Requirements document GR-253-CORE, Generic criteria applicable to SONET and other transmission systems are found in Telcordia GR-499-CORE. The primary difficulty in doing this prior to SONET/SDH was that the sources of these various circuits were different. This meant that circuit was actually operating at a slightly different rate. SONET/SDH allowed for the transport of many different circuits of differing origin within a single framing protocol. SONET/SDH is not a protocol in itself, but a transport protocol. It quickly evolved mapping structures and concatenated payload containers to transport ATM connections, both SDH and SONET are widely used today, SONET in the United States and Canada, and SDH in the rest of the world. Although the SONET standards were developed before SDH, it is considered a variation of SDH because of SDHs greater worldwide market penetration, SONET is subdivided into four sublayer with some factor such as the path, line, section and physical layer. The SDH standard was defined by the European Telecommunications Standards Institute, and is formalized as International Telecommunication Union standards G.707, G.783, G.784. The SONET standard was defined by Telcordia and American National Standards Institute standard T1.105, which define the set of transmission formats and transmission rates in the range above 51.840 Mbit/s. SDH differs from Plesiochronous Digital Hierarchy in that the rates that are used to transport the data on SONET/SDH are tightly synchronized across the entire network. This synchronization system allows entire inter-country networks to operate synchronously, greatly reducing the amount of buffering required between elements in the network. Therefore, it is inaccurate to think of SDH or SONET as communications protocols in and of themselves, they are generic, the basic format of a SONET/SDH signal allows it to carry many different services in its virtual container, because it is bandwidth-flexible. SONET and SDH often use different terms to describe identical features or functions and this can cause confusion and exaggerate their differences. With a few exceptions, SDH can be thought of as a superset of SONET, SONET is a set of transport containers that allow for delivery of a variety of protocols, including traditional telephony, ATM, Ethernet, and TCP/IP traffic. SONET therefore is not in itself a native communications protocol and should not be confused as being necessarily connection-oriented in the way that term is usually used, the protocol is a heavily multiplexed structure, with the header interleaved between the data in a complex way. This permits the data to have its own frame rate and be able to float around relative to the SDH/SONET frame structure. This interleaving permits a low latency for the encapsulated data
This article incorporates public domain material from the General Services Administration document "Federal Standard 1037C" (in support of MIL-STD-188).
This article incorporates 
