ADV.

COMMUNICATION LAB 6 TH SEM E&C

the line-coded signal can directly be put on a transmission line, in the form of variations of the voltage or current (often using differential signaling).

the line-coded signal (the "base-band signal") undergoes further pulse shaping (to reduce its frequency bandwidth) and then modulated (to shift its frequency bandwidth) to create the "RF signal" that can be sent through free space.

LINE CODING
Line coding consists of representing the digital signal to be transported by an amplitude- and time-discrete signal that is optimally tuned for the specific properties of the physical channel (and of the receiving equipment). The waveform pattern of voltage or current used to represent the 1s and 0s of a digital data on a transmission link is called line encoding. The common types of line encoding are unipolar, polar, bipolar and Manchester encoding. For reliable clock recovery at the receiver, one usually imposes a maximum run length constraint on the generated channel sequence, i.e. the maximum number of consecutive ones or zeros is bounded to a reasonable number. A clock period is recovered by observing transitions in the received sequence, so that a maximum run length guarantees such clock recovery, while sequences without such a constraint could seriously hamper the detection quality. After line coding, the signal is put through a "physical channel", either a "transmission medium" or "data storage medium". Sometimes the characteristics of two very different-seeming channels are similar enough that the same line code is used for them. The most common physical channels are:

the line-coded signal can be used to turn on and off a light in Free Space Optics, most commonly infrared remote control. the line-coded signal can be printed on paper to create a bar code. the line-coded signal can be converted to a magnetized spots on a hard drive or tape drive. the line-coded signal can be converted to a pits on optical disc. Unfortunately, most long-distance communication channels cannot transport a DC component. The DC component is also called the disparity, the bias, or the DC coefficient. The simplest possible line code, called unipolar because it has an unbounded DC component, gives too many errors on such systems. Most line codes eliminate the DC component such codes are called DC balanced, zero-DC, zero-bias or DC equalized etc. There are two ways of eliminating the DC component: Use a constant-weight code. In other words, design each transmitted code word such that every code word that contains some positive or negative levels also contains

enough of the opposite levels, such that the average level over each code word is zero. For example, Manchester code and Interleaved 2 of 5.

AMI Modified AMI codes: B8ZS, B6ZS, B3ZS, HDB3 2B1Q 4B5B 4B3T 6b/8b encoding Hamming Code 8b/10b encoding 64b/66b encoding 128b/130b encoding Coded mark inversion (CMI) Conditioned Diphase Eight-to-Fourteen Modulation (EFM) used in Compact Disc EFMPlus used in DVD RZ Return-to-zero NRZ Non-return-to-zero NRZI Non-return-to-zero, inverted Manchester code (also variants Differential Manchester & Biphase mark code) Miller encoding (also known as Delay encoding or Modified Frequency Modulation, and has variant Modified Miller encoding)

Use a paired disparity code. In other words, design the receiver such that every code word that averages to a negative level is paired with another code word that averages to a positive level. Design the receiver so that either code word of the pair decodes to the same data bits. Design the transmitter to keep track of the running DC buildup, and always pick the code word that pushes the DC level back towards zero. For example, AMI, 8B10B, 4B3T, etc.

Line coding should make it possible for the receiver to synchronize itself to the phase of the received signal. If the synchronization is not ideal, then the signal to be decoded will not have optimal differences (in amplitude) between the various digits or symbols used in the line code. This will increase the error probability in the received data. It is also preferred for the line code to have a structure that will enable error detection. Note that the line-coded signal and a signal produced at a terminal may differ, thus requiring translation. A line code will typically reflect technical requirements of the transmission medium, such as optical fiber or shielded twisted pair. These requirements are unique for each medium, because each one has different behavior related to interference, distortion, capacitance and loss of amplitude.

MLT-3 Encoding Hybrid Ternary Codes Surround by complement (SBC) TC-PAM

Optical line codes:

Carrier-Suppressed Return-to-Zero Alternate-Phase Return-to-Zero

[EDIT ] COMMON LINE CODES

NON-RETURN-TO-ZERO
From Wikipedia, the free encyclopedia Jump to: navigation, search

mechanisms for bit synchronization when a separate clock signal is not available. NRZ-Level itself is not a synchronous system but rather an encoding that can be used in either a synchronous or asynchronous transmission environment, that is, with or without an explicit clock signal involved. Because of this, it is not strictly necessary to discuss how the NRZ-Level encoding acts "on a clock edge" or "during a clock cycle" since all transitions happen in the given amount of time representing the actual or implied integral clock cycle. The real question is that of sampling--the high or low state will be received correctly provided the transmission line has stabilized for that bit when the physical line level is sampled at the receiving end. However, it is helpful to see NRZ transitions as happening on the trailing (falling) clock edge in order to compare NRZ-Level to other encoding methods, such as the mentioned Manchester code, which requires clock edge information (is the XOR of the clock and NRZ, actually) and to see the difference between NRZMark and NRZ-Inverted.

The binary signal is encoded using rectangular pulse amplitude modulation with polar non-return-to-zero code In telecommunication, a non-return-to-zero (NRZ) line code is a binary code in which 1's are represented by one significant condition (usually a positive voltage) and 0's are represented by some other significant condition (usually a negative voltage), with no other neutral or rest condition. The pulses have more energy than a RZ code. Unlike RZ, NRZ does not have a rest state. NRZ is not inherently a self-synchronizing code, so some additional synchronization technique (for example a run length limited constraint, or a parallel synchronization signal) must be used to avoid bit slip. For a given data signaling rate, i.e., bit rate, the NRZ code requires only half the bandwidth required by the Manchester code. When used to represent data in an asynchronous communication scheme, the absence of a neutral state requires other

CONTENTS

1 Unipolar Non-Return-to-Zero Level 2 Bipolar Non-Return-to-Zero Level 3 Non-Return-to-Zero Space 4 Non-Return-to-Zero Inverted (NRZI) 5 See also 6 References

[EDIT ] UNIPOLAR NON-RETURN-TO-ZERO LEVEL

Main article: On-off keying "One" is represented by one physical level (such as a DC bias on the transmission line). "Zero" is represented by another level (usually a positive voltage).

An example of this is RS-232, where "one" is 5V to 12V and "zero" is +5 to +12V.

[EDIT ] NON-RETURN-TO-ZERO SPACE

Non-Return-to-Zero Space "One" is represented by no change in physical level. In clock language, "one" transitions or remains high on the trailing clock edge of the previous bit and "zero" transitions or remains low on the trailing clock edge of the previous bit, or just the opposite. This allows for long series without change, which makes synchronization difficult. One solution is to not send bytes without transitions. Disadvantages of an on-off keying are the waste of power due to the transmitted DC level and the power spectrum of the transmitted signal does not approach zero at zero frequency. See RLL "Zero" is represented by a change in physical level. In clock language, the level transitions on the trailing clock edge of the previous bit to represent a "zero." This "change-on-zero" is used by High-Level Data Link Control and USB. They both avoid long periods of no transitions (even when the data contains long sequences of 1 bits) by using zerobit insertion. HDLC transmitters insert a 0 bit after five contiguous 1 bits (except when transmitting the frame delimiter '01111110'). USB transmitters insert a 0 bit after six consecutive 1 bits. The receiver at the far end uses every transition both "One" is represented by one physical level (usually a negative voltage). "Zero" is represented by another level (usually a positive voltage). In clock language, in bipolar NRZ-Level the voltage "swings" from positive to negative on the trailing edge of the previous bit clock cycle. from 0 bits in the data and these extra non-data 0 bits to maintain clock synchronization. The receiver otherwise ignores these non-data 0 bits.

[EDIT ] BIPOLAR NON-RETURN-TO-ZERO LEVEL

[EDIT ] NON-RETURN-TO-ZERO INVERTED (NRZI)

Example NRZI encoding

From Wikipedia, the free encyclopedia Jump to: navigation, search In telecommunication, Manchester code (also known as Phase

NRZ-transition occurs for a zero Non return to zero, inverted (NRZI) is a method of mapping a binary signal to a physical signal for transmission over some transmission media. The two level NRZI signal has a transition at a clock boundary if the bit being transmitted is a logical 1, and does not have a transition if the bit being transmitted is a logical 0. "One" is represented by a transition of the physical level. "Zero" has no transition. Also, NRZI might take the opposite convention, as in Universal Serial Bus (USB) signalling, when in Mode 1 (transition when signalling zero and steady level when signalling one). The transition occurs on the leading edge of the clock for the given bit. This distinguishes NRZI from NRZ-Mark. However, even NRZI can have long series of zeros (or ones if transitioning on "zero"), so clock recovery can be difficult unless some form of run length limited (RLL) coding is used on top. Magnetic disk and tape storage devices generally use fixed-rate RLL codes, while USB uses bit stuffing, which is efficient, but results in a variable data rate: it takes slightly longer to send a long string of 1 bits over USB than it does to send a long string of 0 bits. (USB inserts an additional 0 bit after 6 consecutive 1 bits.)

Encoding, or PE) is a line code in which the encoding of each data bit has at least one transition and occupies the same time. It therefore has no DC component, and is self-clocking, which means that it may be inductively or capacitively coupled, and that a clock signal can be recovered from the encoded data. Manchester code is widely used (e.g. in Ethernet; see also RFID). There are more complex codes, such as 8B/10B encoding, that use less bandwidth to achieve the same data rate but may be less tolerant of frequency errors and jitter in the transmitter and receiver reference clocks.

CONTENTS

1 Features 2 Description o 2.1 Manchester encoding as phase-shift keying o 2.2 Conventions for representation of data 3 References 4 See also

[EDIT ] FEATURES
Manchester code ensures frequent line voltage transitions, directly proportional to the clock rate. This helps clock recovery. The DC component of the encoded signal is not dependent on

MANCHESTER CODE

the data and therefore carries no information, allowing the signal

to be conveyed conveniently by media (e.g. Ethernet) which usually do not convey a DC component.

A 0 is expressed by a low-to-high transition, a 1 by highto-low transition (according to G.E. Thomas' convention -in the IEEE 802.3 convention, the reverse is true).

[EDIT ] DESCRIPTION

The transitions which signify 0 or 1 occur at the midpoint of a period. Transitions at the start of a period are overhead and don't signify data.

Manchester code always has a transition at the middle of each bit period and may (depending on the information to be transmitted) have a transition at the start of the period also. The direction of the mid-bit transition indicates the data. Transitions at the period boundaries do not carry information. They exist only to place the signal in the correct state to allow the mid-bit transition. The existence of guaranteed transitions allows the An example of Manchester encoding showing both conventions Extracting the original data from the received encoded bit (from Manchester as per 802.3):
original data XOR clock 0 0 1 1 0 1 0 1 = Manchester value 0

signal to be self-clocking, and also allows the receiver to align correctly; the receiver can identify if it is misaligned by half a bit period, as there will no longer always be a transition during each bit period. The price of these benefits is a doubling of the bandwidth requirement compared to simpler NRZ coding schemes (or see also NRZI). In the Thomas convention, the result is that the first half of a bit

1 1 0

period matches the information bit and the second half is its complement.

[EDIT] MANCHESTER ENCODING AS PHASE-SHIFT KEYING

Manchester encoding is a special case of binary phase-shift keying (BPSK), where the data controls the phase of a square wave carrier whose frequency is the data rate. Such a signal is easy to generate.

Summary:

Each bit is transmitted in a fixed time (the "period").

[EDIT] CONVENTIONS FOR REPRESENTATION OF DATA

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Encoding of 11011000100 in Manchester code (as per G. E. Thomas) There are two opposing conventions for the representations of data. The first of these was first published by G. E. Thomas in 1949 and is followed by numerous authors (e.g., Tanenbaum). It specifies that for a 0 bit the signal levels will be Low-High (assuming an amplitude physical encoding of the data) - with a low level in the first half of the bit period, and a high level in the second half. For a 1 bit the signal levels will be High-Low.

A universal asynchronous receiver/transmitter (usually abbreviated UART and pronounced /jurt/) is a type of "asynchronous receiver/transmitter", a piece of computer hardware that translates data between parallel and serial forms. UARTs are commonly used in conjunction with communication standards such as EIA RS-232, RS-422 or RS-485. The universal designation indicates that the data format and transmission speeds are configurable and that the actual electric signaling levels and methods (such as differential signaling etc) typically are handled by a special driver circuit external to the UART. A UART is usually an individual (or part of an) integrated circuit

The second convention is also followed by numerous authors (e.g., Stallings) as well as by IEEE 802.4 (token bus) and lower speed versions of IEEE 802.3 (Ethernet) standards. It states that a logic 0 is represented by a High-Low signal sequence and a logic 1 is represented by a Low-High signal sequence. If a Manchester encoded signal is inverted in communication, it is transformed from one convention to the other. This ambiguity can be overcome by using differential Manchester encoding.

used for serial communications over a computer or peripheral device serial port. UARTs are now commonly included in microcontrollers. A dual UART, or DUART, combines two UARTs into a single chip. Many modern ICs now come with a UART that can also communicate synchronously; these devices are called USARTs (universal synchronous/asynchronous receiver/transmitter).

CONTENTS

UNIVERSAL ASYNCHRONOUS RECEIVER/TRANSMITTER

1 Transmitting and receiving serial data

2 3 4 5

6 7 8

1.1 Character framing 1.2 Receiver 1.3 Transmitter 1.4 Application Synchronous transmission History Structure Special receiver conditions o 5.1 Overrun error o 5.2 Underrun error o 5.3 Framing error o 5.4 Parity error o 5.5 Break condition UART models See also References
o o o o

standards for voltage signaling are RS-232, RS-422 and RS-485 from the EIA. Historically, the presence or absence of current (in current loops) was used in telegraph circuits. Some signaling schemes do not use electrical wires. Examples of such are optical fiber, IrDA (infrared), and (wireless) Bluetooth in its Serial Port Profile (SPP). Some signaling schemes use modulation of a carrier signal (with or without wires). Examples are modulation of audio signals with phone line modems, RF modulation with data radios, and the DC-LIN for power line communication. Communication may be "full duplex" (both send and receive at the same time) or "half duplex" (devices take turns transmitting and receiving).

9 External links

[EDIT] CHARACTER FRAMING [EDIT ] TRANSMITTING AND RECEIVING SERIAL DATA

See also: Asynchronous serial communication The Universal Asynchronous Receiver/Transmitter (UART) takes bytes of data and transmits the individual bits in a sequential fashion. At the destination, a second UART re-assembles the bits into complete bytes. Each UART contains a shift register which is the fundamental method of conversion between serial and parallel forms. Serial transmission of digital information (bits) through a single wire or other medium is much more cost effective than parallel transmission through multiple wires. The UART usually does not directly generate or receive the external signals used between different items of equipment. Separate interface devices are used to convert the logic level signals of the UART to and from the external signaling levels. External signals may be of many different forms. Examples of Each character is sent as a logic low start bit, a configurable number of data bits (usually 7 or 8, sometimes 5), an optional parity bit, and one or more logic high stop bits. The start bit signals the receiver that a new character is coming. The next five to eight bits, depending on the code set employed, represent the character. Following the data bits may be a parity bit. The next one or two bits are always in the mark (logic high, i.e., '1') condition and called the stop bit(s). They signal the receiver that the character is completed. Since the start bit is logic low (0) and the stop bit is logic high (1) then there is always a clear demarcation between the previous character and the next one.

[EDIT] RECEIVER

All operations of the UART hardware are controlled by a clock signal which runs at a multiple (say, 16) of the data rate - each data bit is as long as 16 clock pulses. The receiver tests the state of the incoming signal on each clock pulse, looking for the beginning of the start bit. If the apparent start bit lasts at least one-half of the bit time, it is valid and signals the start of a new character. If not, the spurious pulse is ignored. After waiting a further bit time, the state of the line is again sampled and the resulting level clocked into a shift register. After the required number of bit periods for the character length (5 to 8 bits, typically) have elapsed, the contents of the shift register is made available (in parallel fashion) to the receiving system. The UART will set a flag indicating new data is available, and may also generate a processor interrupt to request that the host processor transfers the received data. In some common types of UART, a small first-in, first-out FIFO buffer memory is inserted between the receiver shift register and the host system interface. This allows the host processor more time to handle an interrupt from the UART and prevents loss of received data at high rates.

interrupt. Since full-duplex operation requires characters to be sent and received at the same time, practical UARTs use two different shift registers for transmitted characters and received characters.

[EDIT] APPLICATION
Transmitting and receiving UARTs must be set for the same bit speed, character length, parity, and stop bits for proper operation. The receiving UART may detect some mismatched settings and set a "framing error" flag bit for the host system; in exceptional cases the receiving UART will produce an erratic stream of mutilated characters and transfer them to the host system. Typical serial ports used with personal computers connected to modems use eight data bits, no parity, and one stop bit; for this configuration the number of ASCII characters per second equals the bit rate divided by 10. Some very low-cost home computers or embedded systems dispensed with a UART and used the CPU to sample the state of an input port or directly manipulate an output port for data transmission. While very CPU-intensive, since the CPU timing was critical, these schemes avoided the purchase of a costly UART chip. The technique was known as a bit-banging serial port.

[EDIT] TRANSMITTER
Transmission operation is simpler since it is under the control of the transmitting system. As soon as data is deposited in the shift register after completion of the previous character, the UART hardware generates a start bit, shifts the required number of data bits out to the line,generates and appends the parity bit (if used), and appends the stop bits. Since transmission of a single character may take a long time relative to CPU speeds, the UART will maintain a flag showing busy status so that the host system does not deposit a new character for transmission until the previous one has been completed; this may also be done with an

[EDIT ] SYNCHRONOUS TRANSMISSION

USART chips have both synchronous and asynchronous modes. In synchronous transmission, the clock data is recovered separately from the data stream and no start/stop bits are used. This improves the efficiency of transmission on suitable channels

since more of the bits sent are usable data and not character framing. An asynchronous transmission sends no characters over the interconnection when the transmitting device has nothing to send; but a synchronous interface must send "pad" characters to maintain synchronization between the receiver and transmitter. The usual filler is the ASCII "SYN" character. This may be done automatically by the transmitting device.

Depending on the manufacturer, different terms are used to identify devices that perform the UART functions. Intel called their 8251 device a "Programmable Communication Interface". MOS Technology 6551 was known under the name "Asynchronous Communications Interface Adapter" (ACIA). The term "Serial Communications Interface" (SCI) was first used at Motorola around 1975 to refer to their start-stop asynchronous serial interface device, which others were calling a UART.

[EDIT ] HISTORY [EDIT ] STRUCTURE

Some early telegraph schemes used variable-length pulses (as in Morse code) and rotating clockwork mechanisms to transmit alphabetic characters. The first UART-like devices (with fixedlength pulses) were rotating mechanical switches (commutators). These sent 5-bit Baudot codes for mechanical teletypewriters, and replaced morse code. Later, ASCII required a seven bit code. When IBM built computers in the early 1960s with 8-bit characters, it became customary to store the ASCII code in 8 bits. Gordon Bell designed the UART for the PDP series of computers. Western Digital made the first single-chip UART WD1402A around 1971; this was an early example of a medium scale integrated circuit. An example of an early 1980s UART was the National Semiconductor 8250. In the 1990s, newer UARTs were developed with on-chip buffers. This allowed higher transmission speed without data loss and without requiring such frequent attention from the computer. For example, the popular National Semiconductor 16550 has a 16 byte FIFO, and spawned many variants, including the 16C550, 16C650, 16C750, and 16C850.

A UART usually contains the following components: a clock generator, usually a multiple of the bit rate to allow sampling in the middle of a bit period. input and output shift registers transmit/receive control read/write control logic transmit/receive buffers (optional) parallel data bus buffer (optional) First-in, first-out (FIFO) buffer memory (optional)

[EDIT ] SPECIAL RECEIVER CONDITIONS [EDIT] OVERRUN ERROR

An "overrun error" occurs when the receiver cannot process the character that just came in before the next one arrives. Various devices have different amounts of buffer space to hold received characters. The CPU must service the UART in order to remove characters from the input buffer. If the CPU does not service the

UART quickly enough and the buffer becomes full, an Overrun Error will occur.

A "break condition" occurs when the receiver input is at the "space" level for longer than some duration of time, typically, for more than a character time. This is not necessarily an error, but

[EDIT] UNDERRUN ERROR

An "underrun error" occurs when the UART transmitter has completed sending a character and the transmit buffer is empty. In asynchronous modes this is treated as an indication that no data remains to be transmitted, rather than an error, since additional stop bits can be appended. This error indication is commonly found in USARTs, since an underrun is more serious in synchronous systems.

appears to the receiver as a character of all zero bits with a framing error. Some equipment will deliberately transmit the "break" level for longer than a character as an out-of-band signal. When signaling rates are mismatched, no meaningful characters can be sent, but a long "break" signal can be a useful way to get the attention of a mismatched receiver to do something (such as resetting itself). Unix-like systems can use the long "break" level as a request to change the signaling rate, to support dial-in access at multiple signaling rates. Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing, or modulating, the phase of a reference signal (the carrier wave). Any digital modulation scheme uses a finite number of distinct signals to represent digital data. PSK uses a finite number of phases, each assigned a unique pattern of binary digits. Usually, each phase encodes an equal number of bits. Each pattern of bits forms the symbol that is represented by the particular phase. The demodulator, which is designed specifically for the symbol-set used by the modulator, determines the phase of the received signal and maps it back to the symbol it represents, thus recovering the original data. This requires the receiver to be able to compare the phase of the received signal to a reference signal such a system is termed coherent (and referred to as CPSK).

[EDIT] FRAMING ERROR

A "framing error" occurs when the designated "start" and "stop" bits are not valid. As the "start" bit is used to identify the beginning of an incoming character, it acts as a reference for the remaining bits. If the data line is not in the expected idle state when the "stop" bit is expected, a Framing Error will occur.

[EDIT] PARITY ERROR

A "parity error" occurs when the number of "active" bits does not agree with the specified parity configuration of the USART, producing a Parity Error. Because the "parity" bit is optional, this error will not occur if parity has been disabled. Parity error is set when the parity of an incoming data character does not match the expected value.

[EDIT] BREAK CONDITION

Alternatively, instead of using the bit patterns to set the phase of the wave, it can instead be used to change it by a specified amount. The demodulator then determines the changes in the phase of the received signal rather than the phase itself. Since this scheme depends on the difference between successive phases, it is termed differential phase-shift keying (DPSK). DPSK can be significantly simpler to implement than ordinary PSK since there is no need for the demodulator to have a copy of the reference signal to determine the exact phase of the received signal (it is a non-coherent scheme). In exchange, it produces more erroneous demodulations. The exact requirements of the particular scenario under consideration determine which scheme is used.

o 6.3 Example: Differentially-encoded BPSK 7 Channel capacity 8 See also 9 Notes

10 References

[EDIT ] INTRODUCTION
There are three major classes of digital modulation techniques used for transmission of digitally represented data:

Amplitude-shift keying (ASK) Frequency-shift keying (FSK) Phase-shift keying (PSK)

CONTENTS

All convey data by changing some aspect of a base signal, the carrier wave (usually a sinusoid), in response to a data signal. In the case of PSK, the phase is changed to represent the data signal. There are two fundamental ways of utilizing the phase of a signal in this way:

1 Introduction o 1.1 Definitions 2 Applications 3 Binary phase-shift keying (BPSK) o 3.1 Implementation o 3.2 Bit error rate 4 Quadrature phase-shift keying (QPSK) o 4.1 Implementation o 4.2 Bit error rate o 4.3 QPSK signal in the time domain o 4.4 Variants 4.4.1 Offset QPSK (OQPSK) 4.4.2 /4QPSK 4.4.3 SOQPSK 4.4.4 DPQPSK 5 Higher-order PSK o 5.1 Bit error rate 6 Differential phase-shift keying (DPSK) o 6.1 Differential encoding o 6.2 Demodulation

By viewing the phase itself as conveying the information, in which case the demodulator must have a reference signal to compare the received signal's phase against; or

By viewing the change in the phase as conveying information differential schemes, some of which do not need a reference carrier (to a certain extent).

A convenient way to represent PSK schemes is on a constellation diagram. This shows the points in the Argand plane where, in this context, the real and imaginary axes are termed the in-phase and quadrature axes respectively due to their 90 separation. Such a representation on perpendicular axes lends itself to

straightforward implementation. The amplitude of each point along the in-phase axis is used to modulate a cosine (or sine) wave and the amplitude along the quadrature axis to modulate a sine (or cosine) wave. In PSK, the constellation points chosen are usually positioned with uniform angular spacing around a circle. This gives maximum phase-separation between adjacent points and thus the best immunity to corruption. They are positioned on a circle so that they can all be transmitted with the same energy. In this way, the moduli of the complex numbers they represent will be the same and thus so will the amplitudes needed for the cosine and sine waves. Two common examples are "binary phase-shift keying" (BPSK) which uses two phases, and "quadrature phaseshift keying" (QPSK) which uses four phases, although any number of phases may be used. Since the data to be conveyed are usually binary, the PSK scheme is usually designed with the number of constellation points being a power of 2.

Q(x) will give the probability that a single sample taken from a
random process with zero-mean and unit-variance Gaussian probability density function will be greater or equal to x. It is a scaled form of the complementary Gaussian error function:

. The error-rates quoted here are those in additive white Gaussian noise (AWGN). These error rates are lower than those computed in fading channels, hence, are a good theoretical benchmark to compare with.

[EDIT ] APPLICATIONS
Owing to PSK's simplicity, particularly when compared with its competitor quadrature amplitude modulation, it is widely used in existing technologies. The wireless LAN standard, IEEE 802.11b-1999[1][2], uses a variety of different PSKs depending on the data-rate required. At the basic-rate of 1 Mbit/s, it uses DBPSK (differential BPSK). To provide the extended-rate of 2 Mbit/s, DQPSK is used. In reaching 5.5 Mbit/s and the full-rate of 11 Mbit/s, QPSK is employed, but has to be coupled with complementary code keying. The higherspeed wireless LAN standard, IEEE 802.11g-2003[1][3] has eight data rates: 6, 9, 12, 18, 24, 36, 48 and 54 Mbit/s. The 6 and 9 Mbit/s modes use OFDM modulation where each sub-carrier is BPSK modulated. The 12 and 18 Mbit/s modes use OFDM with QPSK. The fastest four modes use OFDM with forms of quadrature amplitude modulation.

[EDIT] DEFINITIONS
For determining error-rates mathematically, some definitions will be needed:

Eb = Energy-per-bit Es = Energy-per-symbol = nEb with n bits per symbol Tb = Bit duration Ts = Symbol duration N0 / 2 = Noise power spectral density (W/Hz) Pb = Probability of bit-error Ps = Probability of symbol-error

Because of its simplicity BPSK is appropriate for low-cost passive transmitters, and is used in RFID standards such as ISO/IEC 14443 which has been adopted for biometric passports, credit cards such as American Express's ExpressPay, and many other applications[4]. Bluetooth 2 will use

Constellation diagram example for BPSK. BPSK (also sometimes called PRK, Phase Reversal Keying, or 2PSK) is the simplest form of phase shift keying (PSK). It uses two phases which are separated by 180 and so can also be termed 2-PSK. It does not particularly matter exactly where the

/ 4-DQPSK at its lower rate (2 Mbit/s) and

constellation points are positioned, and in this figure they are shown on the real axis, at 0 and 180. This modulation is the most robust of all the PSKs since it takes the highest level of noise or distortion to make the demodulator reach an incorrect decision. It is, however, only able to modulate at 1 bit/symbol (as seen in the figure) and so is unsuitable for high data-rate applications when bandwidth is limited. In the presence of an arbitrary phase-shift introduced by the communications channel, the demodulator is unable to tell which constellation point is which. As a result, the data is often differentially encoded prior to modulation.

8-DPSK at its higher rate (3 Mbit/s) when the link between the two devices is sufficiently robust. Bluetooth 1 modulates with Gaussian minimum-shift keying, a binary scheme, so either modulation choice in version 2 will yield a higher data-rate. A similar technology, IEEE 802.15.4 (the wireless standard used by ZigBee) also relies on PSK. IEEE 802.15.4 allows the use of two frequency bands: 868915 MHz using BPSK and at 2.4 GHz using OQPSK. Notably absent from these various schemes is 8-PSK. This is because its error-rate performance is close to that of 16-QAM it is only about 0.5 dB better[citation needed] but its data rate is only three-quarters that of 16-QAM. Thus 8-PSK is often omitted from standards and, as seen above, schemes tend to 'jump' from QPSK to 16-QAM (8-QAM is possible but difficult to implement).

[EDIT] IMPLEMENTATION
The general form for BPSK follows the equation:

[EDIT ] BINARY PHASE-SHIFT KEYING (BPSK)

This yields two phases, 0 and . In the specific form, binary data is often conveyed with the following signals:

f or binary "0"

for binary "1" where fc is the frequency of the carrier-wave. Hence, the signal-space can be represented by the single basis function Constellation diagram for QPSK with Gray coding. Each adjacent symbol only differs by one bit. Sometimes this is known as quaternary PSK, quadriphase PSK, 4PSK, or 4-QAM. (Although the root concepts of QPSK and 4-QAM are different, the resulting modulated radio waves are exactly the same.) QPSK uses four points on the constellation diagram, The use of this basis function is shown at the end of the next section in a signal timing diagram. The topmost signal is a BPSKmodulated cosine wave that the BPSK modulator would produce. The bit-stream that causes this output is shown above the signal (the other parts of this figure are relevant only to QPSK). The mathematical analysis shows that QPSK can be used either equispaced around a circle. With four phases, QPSK can encode two bits per symbol, shown in the diagram with gray coding to minimize the bit error rate (BER) sometimes misperceived as twice the BER of BPSK.

where 1 is represented by

and 0 is represented by

. This assignment is, of course, arbitrary.

[EDIT] BIT ERROR RATE

The bit error rate (BER) of BPSK in AWGN can be calculated as :
[5]

to double the data rate compared with a BPSK system while maintaining the same bandwidth of the signal, or to maintain the data-rate of BPSK but halving the bandwidth needed. In this latter case, the BER of QPSK is exactly the same as the BER of BPSK - and deciding differently is a common confusion when

or Since there is only one bit per symbol, this is also the symbol error rate.

considering or describing QPSK. Given that radio communication channels are allocated by agencies such as the Federal Communication Commission giving a prescribed (maximum) bandwidth, the advantage of QPSK over BPSK becomes evident: QPSK transmits twice the data rate in a given bandwidth compared to BPSK - at the same BER. The

[EDIT ] QUADRATURE PHASE-SHIFT KEYING (QPSK)

engineering penalty that is paid is that QPSK transmitters and receivers are more complicated than the ones for BPSK. However, with modern electronics technology, the penalty in cost is very moderate. As with BPSK, there are phase ambiguity problems at the receiving end, and differentially encoded QPSK is often used in practice.

Hence, the signal constellation consists of the signal-space 4 points

The factors of 1/2 indicate that the total power is split equally between the two carriers. Comparing these basis functions with that for BPSK shows clearly how QPSK can be viewed as two independent BPSK signals. Note that the signal-space points for BPSK do not need to split the symbol (bit) energy over the two carriers in the scheme shown in the BPSK constellation diagram. QPSK systems can be implemented in a number of ways. An illustration of the major components of the transmitter and receiver structure are shown below.

[EDIT] IMPLEMENTATION
The implementation of QPSK is more general than that of BPSK and also indicates the implementation of higher-order PSK. Writing the symbols in the constellation diagram in terms of the sine and cosine waves used to transmit them:

This yields the four phases /4, 3/4, 5/4 and 7/4 as needed. This results in a two-dimensional signal space with unit basis functions

Conceptual transmitter structure for QPSK. The binary data stream is split into the in-phase and quadrature-phase The first basis function is used as the in-phase component of the signal and the second as the quadrature component of the signal. components. These are then separately modulated onto two orthogonal basis functions. In this implementation, two sinusoids are used. Afterwards, the two signals are superimposed, and the resulting signal is the QPSK signal. Note the use of polar non-

return-to-zero encoding. These encoders can be placed before for binary data source, but have been placed after to illustrate the conceptual difference between digital and analog signals involved with digital modulation.

However, in order to achieve the same bit-error probability as BPSK, QPSK uses twice the power (since two bits are transmitted simultaneously). The symbol error rate is given by:

. Receiver structure for QPSK. The matched filters can be replaced with correlators. Each detection device uses a reference threshold value to determine whether a 1 or 0 is detected. If the signal-to-noise ratio is high (as is necessary for practical QPSK systems) the probability of symbol error may be approximated:

[EDIT] BIT ERROR RATE

Although QPSK can be viewed as a quaternary modulation, it is easier to see it as two independently modulated quadrature carriers. With this interpretation, the even (or odd) bits are used to modulate the in-phase component of the carrier, while the odd (or even) bits are used to modulate the quadrature-phase component of the carrier. BPSK is used on both carriers and they can be independently demodulated. As a result, the probability of bit-error for QPSK is the same as for BPSK:

[EDIT] QPSK SIGNAL IN THE TIME DOMAIN

The modulated signal is shown below for a short segment of a random binary data-stream. The two carrier waves are a cosine wave and a sine wave, as indicated by the signal-space analysis above. Here, the odd-numbered bits have been assigned to the in-phase component and the even-numbered bits to the quadrature component (taking the first bit as number 1). The total signal the sum of the two components is shown at the bottom. Jumps in phase can be seen as the PSK changes the phase on each component at the start of each bit-period. The topmost waveform alone matches the description given for BPSK above.

Timing diagram for QPSK. The binary data stream is shown beneath the time axis. The two signal components with their bit assignments are shown the top and the total, combined signal at the bottom. Note the abrupt changes in phase at some of the bitperiod boundaries. The binary data that is conveyed by this waveform is: 1 1 0 0 0 1 1 0.

Signal doesn't cross zero, because only one bit of the symbol is changed at a time Offset quadrature phase-shift keying (OQPSK) is a variant of phase-shift keying modulation using 4 different values of the phase to transmit. It is sometimes called Staggered quadrature phase-shift keying (SQPSK).

The odd bits, highlighted here, contribute to the in-phase component: 1 1 0 0 0 1 1 0 The even bits, highlighted here, contribute to the quadrature-phase component: 1 1 0 0 0 1 1 0

[EDIT] VARIANTS
[EDIT ] OFFSET QPSK (OQPSK)

Difference of the phase between QPSK and OQPSK Taking four values of the phase (two bits) at a time to construct a QPSK symbol can allow the phase of the signal to jump by as much as 180 at a time. When the signal is low-pass filtered (as

is typical in a transmitter), these phase-shifts result in large amplitude fluctuations, an undesirable quality in communication systems. By offsetting the timing of the odd and even bits by one bit-period, or half a symbol-period, the in-phase and quadrature components will never change at the same time. In the constellation diagram shown on the right, it can be seen that this will limit the phase-shift to no more than 90 at a time. This yields much lower amplitude fluctuations than non-offset QPSK and is sometimes preferred in practice. The picture on the right shows the difference in the behavior of the phase between ordinary QPSK and OQPSK. It can be seen that in the first plot the phase can change by 180 at once, while in OQPSK the changes are never greater than 90. The modulated signal is shown below for a short segment of a random binary data-stream. Note the half symbol-period offset between the two component waves. The sudden phase-shifts occur about twice as often as for QPSK (since the signals no longer change together), but they are less severe. In other words, the magnitude of jumps is smaller in OQPSK when compared to QPSK.

Timing diagram for offset-QPSK. The binary data stream is shown beneath the time axis. The two signal components with their bit assignments are shown the top and the total, combined signal at the bottom. Note the half-period offset between the two signal components. [EDIT ] /4QPSK

Dual constellation diagram for /4-QPSK. This shows the two separate constellations with identical Gray coding but rotated by 45 with respect to each other. This final variant of QPSK uses two identical constellations which are rotated by 45 (

/ 4 radians, hence the name) with respect

to one another. Usually, either the even or odd symbols are used to select points from one of the constellations and the other symbols select points from the other constellation. This also reduces the phase-shifts from a maximum of 180, but only to a maximum of 135 and so the amplitude fluctuations of QPSK are between OQPSK and non-offset QPSK. One property this modulation scheme possesses is that if the modulated signal is represented in the complex domain, it does not have any paths through the origin. In other words, the signal

/ 4

does not pass through the origin. This lowers the dynamical range of fluctuations in the signal which is desirable when engineering communications signals. On the other hand, telephone systems.

[EDIT ] SOQPSK The license-free shaped-offset QPSK (SOQPSK) is interoperable with Feher-patented QPSK (FQPSK), in the sense that an integrate-and-dump offset QPSK detector produces the same output no matter which kind of transmitter is used[1]. These modulations carefully shape the I and Q waveforms such

/ 4QPSK lends itself to easy demodulation

and has been adopted for use in, for example, TDMA cellular

The modulated signal is shown below for a short segment of a random binary data-stream. The construction is the same as above for ordinary QPSK. Successive symbols are taken from the two constellations shown in the diagram. Thus, the first symbol (1 1) is taken from the 'blue' constellation and the second symbol (0 0) is taken from the 'green' constellation. Note that magnitudes of the two component waves change as they switch between constellations, but the total signal's magnitude remains constant. The phase-shifts are between those of the two previous timing-diagrams.

that they change very smoothly, and the signal stays constantamplitude even during signal transitions. (Rather than traveling instantly from one symbol to another, or even linearly, it travels smoothly around the constant-amplitude circle from one symbol to the next.) The standard description of SOQPSK-TG involves ternary symbols. [EDIT ] DPQPSK Dual-polarization quadrature phase shift keying (DPQPSK) or dual-polarization QPSK - involves the polarization multiplexing of two different QPSK signals, thus improving the spectral efficiency by a factor of 2. This is a cost-effective alternative, to utilizing 16-PSK instead of QPSK to double the spectral the efficiency.

[EDIT ] HIGHER-ORDER PSK

Timing diagram for /4-QPSK. The binary data stream is shown beneath the time axis. The two signal components with their bit assignments are shown the top and the total, combined signal at the bottom. Note that successive symbols are taken alternately from the two constellations, starting with the 'blue' one.

, ,

and

and Constellation diagram for 8-PSK with Gray coding. Any number of phases may be used to construct a PSK constellation but 8-PSK is usually the highest order PSK constellation deployed. With more than 8 phases, the error-rate becomes too high and there are better, though more complex, modulations available such as quadrature amplitude modulation (QAM). Although any number of phases may be used, the fact that the constellation must usually deal with binary data means that the number of symbols is usually a power of 2 this allows an equal number of bits-per-symbol. Gaussian random variables.

are jointly-

[EDIT] BIT ERROR RATE

Bit-error rate curves for BPSK, QPSK, 8-PSK and 16-PSK, AWGN channel.

M-PSK there is no simple expression for the symbol-error probability if M > 4. Unfortunately, it can only be
For the general obtained from:

This may be approximated for high

M and high Eb / N0 by:

. The bit-error probability for where

M-PSK can only be determined

exactly once the bit-mapping is known. However, when Gray coding is used, the most probable error from one symbol to the next produces only a single bit-error and ,

For example, in differentially-encoded BPSK a binary '1' may be . (Using Gray coding allows us to approximate the Lee distance of the errors as the Hamming distance of the errors in the decoded bitstream, which is easier to implement in hardware.) The graph on the left compares the bit-error rates of BPSK, QPSK (which are the same, as noted above), 8-PSK and 16-PSK. It is seen that higher-order modulations exhibit higher error-rates; in exchange however they deliver a higher raw data-rate. Bounds on the error rates of various digital modulation schemes can be computed with application of the union bound to the signal constellation. transmitted by adding 180 to the current phase and a binary '0' by adding 0 to the current phase. Another variant of DPSK is Symmetric Differential Phase Shift keying, SDPSK, where encoding would be +90 for a '1' and -90 for a '0'. In differentially-encoded QPSK (DQPSK), the phase-shifts are 0, 90, 180, -90 corresponding to data '00', '01', '11', '10'. This kind of encoding may be demodulated in the same way as for non-differential PSK but the phase ambiguities can be ignored. Thus, each received symbol is demodulated to one of the difference in phase between this received signal and the preceding one. The difference encodes the data as described above. Symmetric Differential Quadrature Phase Shift Keying (SDQPSK) is like DQPSK, but encoding is symmetric, using phase shift values of -135, -45, +45 and +135. The modulated signal is shown below for both DBPSK and DQPSK as described above. In the figure, it is assumed that the signal starts with zero phase, and so there is a phase shift in both signals at

points in the constellation and a comparator then computes the

[EDIT ] DIFFERENTIAL PHASE-SHIFT KEYING (DPSK)

This article may require cleanup to meet Wikipedia's quality standards. Please improve this article if you can. The talk page may contain suggestions. (May 2009)

[EDIT] DIFFERENTIAL ENCODING

Main article: differential coding Differential phase shift keying (DPSK) is a common form of phase modulation that conveys data by changing the phase of the carrier wave. As mentioned for BPSK and QPSK there is an ambiguity of phase if the constellation is rotated by some effect in the communications channel through which the signal passes. This problem can be overcome by using the data to change rather than set the phase.

t = 0.

Timing diagram for DBPSK and DQPSK. The binary data stream is above the DBPSK signal. The individual bits of the DBPSK signal are grouped into pairs for the DQPSK signal, which only changes every Ts = 2Tb.

Analysis shows that differential encoding approximately doubles

system. This channel will, in general, introduce an unknown phase-shift to the PSK signal; in these cases the differential schemes can yield a better error-rate than the ordinary schemes which rely on precise phase information.

M-PSK but this may be overcome by only a small increase in Eb / N0. Furthermore, this
the error rate compared to ordinary analysis (and the graphical results below) are based on a system in which the only corruption is additive white Gaussian noise(AWGN). However, there will also be a physical channel between the transmitter and receiver in the communication

[EDIT] DEMODULATION

Call the received symbol in the phase

kth timeslot rk and let it have

k. Assume without loss of generality that the phase of the carrier wave is zero. Denote the AWGN term as nk. Then
.

BER comparison between DBPSK, DQPSK and their nondifferential forms using gray-coding and operating in white noise. For a signal that has been differentially encoded, there is an obvious alternative method of demodulation. Instead of demodulating as usual and ignoring carrier-phase ambiguity, the phase between two successive received symbols is compared and used to determine what the data must have been. When differential encoding is used in this manner, the scheme is known as differential phase-shift keying (DPSK). Note that this is subtly different to just differentially-encoded PSK since, upon reception, the received symbols are not decoded one-by-one to constellation points but are instead compared directly to one another.

k 1th symbol and the kth symbol is the phase difference between rk and rk 1. That is, if rk is projected onto rk 1, the decision is taken on the phase of the
The decision variable for the resultant complex number:

where superscript * denotes complex conjugation. In the absence of noise, the phase of this is data transmitted. The probability of error for DPSK is difficult to calculate in general, but, in the case of DBPSK it is:

k k 1, the phase-shift between

the two received signals which can be used to determine the

which, when numerically evaluated, is only slightly worse than ordinary BPSK, particularly at higher

Eb / N0 values.

Using DPSK avoids the need for possibly complex carrierrecovery schemes to provide an accurate phase estimate and can be an attractive alternative to ordinary PSK. In optical communications, the data can be modulated onto the phase of a laser in a differential way. The modulation is a laser which emits a continuous wave, and a Mach-Zehnder modulator which receives electrical binary data. For the case of BPSK for example, the laser transmits the field unchanged for binary '1', and with reverse polarity for '0'. The demodulator consists of a delay line interferometer which delays one bit, so two bits can be compared at one time. In further processing, a photo diode is used to transform the optical field into an electric current, so the information is changed back into its original state. The bit-error rates of DBPSK and DQPSK are compared to their non-differential counterparts in the graph to the right. The loss for using DBPSK is small enough compared to the complexity reduction that it is often used in communications systems that would otherwise use BPSK. For DQPSK though, the loss in performance compared to ordinary QPSK is larger and the system designer must balance this against the reduction in complexity.

kth time-slot call the bit to be modulated bk, the differentially-encoded bit ek and the resulting modulated signal mk(t). Assume that the constellation diagram positions the
At the symbols at 1 (which is BPSK). The differential encoder produces:

where

indicates binary or modulo-2 addition.

BER comparison between BPSK and differentially-encoded BPSK with gray-coding operating in white noise.

[EDIT] EXAMPLE: DIFFERENTIALLY-ENCODED BPSK

ek only changes state (from binary '0' to binary '1' or from binary '1' to binary '0') if bk is a binary '1'. Otherwise it remains in
So its previous state. This is the description of differentially-encoded BPSK given above. The received signal is demodulated to yield

ek = 1 and then the

Differential encoding/decoding system diagram.

differential decoder reverses the encoding procedure and produces:

since binary subtraction is the same as binary addition.

Given a fixed bandwidth, channel capacity vs. SNR for some common modulation schemes Like all M-ary modulation schemes with M = 2b symbols, when given exclusive access to a fixed bandwidth, the channel capacity of any phase shift keying modulation scheme rises to a maximum of b bits per symbol as the SNR increases.

bk = 1 if ek and ek 1 differ and bk = 0 if they are the same. Hence, if both ek and ek 1 are inverted, bk will still be
Therefore, decoded correctly. Thus, the 180 phase ambiguity does not matter. Differential schemes for other PSK modulations may be devised along similar lines. The waveforms for DPSK are the same as for differentially-encoded PSK given above since the only change between the two schemes is at the receiver. The BER curve for this example is compared to ordinary BPSK on the right. As mentioned above, whilst the error-rate is approximately doubled, the increase needed in overcome this is small. The increase in

TIME-DIVISION MULTIPLEXING
From Wikipedia, the free encyclopedia Jump to: navigation, search Time-division multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in which two or more signals or bit streams are transferred apparently simultaneously as subchannels in one communication channel, but are physically taking turns on the channel. The time domain is divided into several recurrent timeslots of fixed length, one for each subchannel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot per sub-channel plus a synchronization channel and sometimes error correction channel before the synchronization. After the last sub-channel, error correction, and synchronization, the cycle starts all over again with a new frame, starting with the second sample, byte or data block from sub-channel 1, etc.

Eb / N0 to

Eb / N0 required to

overcome differential modulation in coded systems, however, is larger - typically about 3 dB. The performance degradation is a result of noncoherent transmission - in this case it refers to the fact that tracking of the phase is completely ignored.

CONTENTS

1 Application examples 2 TDM versus packet mode communication

3 History o 3.1 Transmission using Time Division Multiplexing (TDM) 4 Synchronous time division multiplexing (Sync TDM) 5 Synchronous digital hierarchy (SDH) 6 Statistical time-division multiplexing (Stat TDM) 7 See also 8 Notes 9 References

In its primary form, TDM is used for circuit mode communication with a fixed number of channels and constant bandwidth per channel. Bandwidth Reservation distinguishes time-division multiplexing from statistical multiplexing such as packet mode communication (also known as statistical time-domain multiplexing, see below) i.e. the time-slots are recurrent in a fixed order and pre-allocated to the channels, rather than scheduled on a packet-by-packet basis. Statistical time-domain multiplexing resembles, but should not be considered the same as time-division multiplexing. In dynamic TDMA, a scheduling algorithm dynamically reserves a variable number of timeslots in each frame to variable bit-rate data streams, based on the traffic demand of each data stream. Dynamic TDMA is used in

APPLICATION EXAMPLES

The plesiochronous digital hierarchy (PDH) system, also known as the PCM system, for digital transmission of several telephone calls over the same four-wire copper cable (T-carrier or E-carrier) or fiber cable in the circuit switched digital telephone network

The SDH and synchronous optical networking (SONET) network transmission standards, that have surpassed PDH. The RIFF (WAV) audio standard interleaves left and right stereo signals on a per-sample basis The left-right channel splitting in use for stereoscopic liquid crystal shutter glasses

HIPERLAN/2; Dynamic synchronous Transfer Mode; IEEE 802.16a.

[EDIT ] HISTORY
TDM can be further extended into the time division multiple access (TDMA) scheme, where several stations connected to the same physical medium, for example sharing the same frequency channel, can communicate. Application examples include:

Time-division multiplexing was first developed in telegraphy; see multiplexing in telegraphy: mile Baudot developed a timemultiplexing system of multiple Hughes machines in the 1870s. For the SIGSALY encryptor of 1943, see PCM. In 1962, engineers from Bell Labs developed the first D1 Channel

The GSM telephone system The Tactical Data Links Link 16 and Link 22

[EDIT ] TDM VERSUS PACKET MODE COMMUNICATION

Banks, which combined 24 digitised voice calls over a 4-wire copper trunk between Bell central office analogue switches. A

channel bank sliced a 1.544 Mbit/s digital signal into 8,000 separate frames, each composed of 24 contiguous bytes. Each byte represented a single telephone call encoded into a constant bit rate signal of 64 Kbit/s. Channel banks used a byte's fixed position (temporal alignment) in the frame to determine which call it belonged to.
[1]

higher order multiplex, four TDM frames from the immediate lower order are combined, creating multiplexes with a bandwidth of n x 64 kbit/s, where n = 120, 480, 1920, etc.[2]

[EDIT ] SYNCHRONOUS TIME DIVISION MULTIPLEXING (SYNC TDM)

There are three types of (Sync TDM): T1, SONET/SDH (see below), and ISDN[4].

[EDIT] TRANSMISSION USING TIME DIVISION MULTIPLEXING (TDM)

In circuit switched networks such as the public switched telephone network (PSTN) there exists the need to transmit multiple subscribers calls along the same transmission medium.
[2]

[EDIT ] SYNCHRONOUS DIGITAL HIERARCHY (SDH)

Plesiochronous digital hierarchy (PDH) was developed as a standard for multiplexing higher order frames.[2][3] PDH created larger numbers of channels by multiplexing the standard Europeans 30 channel TDM frames.[2] This solution worked for a while; however PDH suffered from several inherent drawbacks which ultimately resulted in the development of the Synchronous Digital Hierarchy (SDH). The requirements which drove the development of SDH were these:[2][3]

To accomplish this, network designers make use of TDM. TDM

allows switches to create channels, also known as tributaries, within a transmission stream.[2] A standard DS0 voice signal has a data bit rate of 64 kbit/s, determined using Nyquists sampling criterion.[2][3] TDM takes frames of the voice signals and multiplexes them into a TDM frame which runs at a higher bandwidth. So if the TDM frame consists of n voice frames, the bandwidth will be n*64 kbit/s.[2]

Be synchronous All clocks in the system must align with a reference clock. Be service-oriented SDH must route traffic from End Exchange to End Exchange without worrying about exchanges in between, where the bandwidth can be reserved at a fixed level for a fixed period of time.

Each voice sample timeslot in the TDM frame is called a channel .

[2]

In European systems, TDM frames contain 30 digital voice

[2]

channels, and in American systems, they contain 24 channels. Both standards also contain extra bits (or bit timeslots) for signalling (see Signaling System 7) and synchronisation bits.
[2]

Multiplexing more than 24 or 30 digital voice channels is called higher order multiplexing.[2] Higher order multiplexing is accomplished by multiplexing the standard TDM frames.[2] For example, a European 120 channel TDM frame is formed by multiplexing four standard 30 channel TDM frames.[2] At each

Allow frames of any size to be removed or inserted into an SDH frame of any size. Easily manageable with the capability of transferring management data across links. Provide high levels of recovery from faults.

Provide high data rates by multiplexing any size frame, limited only by technology. Give reduced bit rate errors.

therefore extremely fast.[2] Modern optic fibre transmission makes use of Wavelength Division Multiplexing (WDM) where signals transmitted across the fibre are transmitted at different wavelengths, creating additional channels for transmission.[2][3] This increases the speed and capacity of the link, which in turn reduces both unit and total costs.[2]

SDH has become the primary transmission protocol in most PSTN networks.[2][3] It was developed to allow streams 1.544 Mbit/s and above to be multiplexed, in order to create larger SDH frames known as Synchronous Transport Modules (STM).[2] The STM-1 frame consists of smaller streams that are multiplexed to create a 155.52 Mbit/s frame.[2][3] SDH can also multiplex packet based frames e.g. Ethernet, PPP and ATM.[2] While SDH is considered to be a transmission protocol (Layer 1 in the OSI Reference Model), it also performs some switching functions, as stated in the third bullet point requirement listed above.[2] The most common SDH Networking functions are these:

[EDIT ] STATISTICAL TIME-DIVISION MULTIPLEXING (STAT TDM)

STDM is an advanced version of TDM in which both the address of the terminal and the data itself are transmitted together for better routing. Using STDM allows bandwidth to be split over 1 line. Many college and corporate campuses use this type of TDM to logically distribute bandwidth. If there is one 10MBit line coming into the building, STDM can be used to provide 178 terminals with a dedicated 56k connection (178 * 56k = 9.96Mb). A more common use however is to only grant the bandwidth when that much is needed. STDM does not reserve a time slot for each terminal, rather it assigns a slot when the terminal is requiring data to be sent or received. This is also called asynchronous time-division multiplexing[4] (ATDM), in an alternative nomenclature in which "STDM" or "synchronous time division multiplexing" designates the older method that uses fixed time slots.

SDH Crossconnect The SDH Crossconnect is the SDH version of a Time-Space-Time crosspoint switch. It connects any channel on any of its inputs to any channel on any of its outputs. The SDH Crossconnect is used in Transit Exchanges, where all inputs and outputs are connected to other exchanges.[2]

SDH Add-Drop Multiplexer The SDH Add-Drop Multiplexer (ADM) can add or remove any multiplexed frame down to 1.544Mb. Below this level, standard TDM can be performed. SDH ADMs can also perform the task of an SDH Crossconnect and are used in End Exchanges where the channels from subscribers are connected to the core PSTN network.[2]

PULSE-CODE MODULATION
From Wikipedia, the free encyclopedia (Redirected from PCM)

SDH network functions are connected using high-speed optic fibre. Optic fibre uses light pulses to transmit data and is

Jump to: navigation, search "PCM" redirects here. For other uses, see PCM (disambiguation). Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals, which was invented by Alec Reeves in 1937. It is the standard form for digital audio in computers and various Blu-ray, Compact Disc and DVD formats, as well as other uses such as digital telephone systems. A PCM stream is a digital representation of an analog signal, in which the magnitude of the analogue signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a range of digital steps. PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that each sample can take.

[EDIT ] MODULATION

Sampling and quantization of a signal (red) for 4-bit PCM In the diagram, a sine wave (red curve) is sampled and quantized for pulse code modulation. The sine wave is sampled at regular intervals, shown as ticks on the x-axis. For each sample, one of the available values (ticks on the y-axis) is chosen by some algorithm. This produces a fully discrete representation of the input signal (shaded area) that can be easily encoded as digital data for storage or manipulation. For the sine wave example at right, we can verify that the quantized values at the sampling moments are 7, 9, 11, 12, 13, 14, 14, 15, 15, 15, 14, etc. Encoding these values as binary numbers would result in the following set of nibbles: 0111 (230+221+211+201=0+4+2+1=7), 1001, 1011, 1100, 1101, 1110, 1110, 1111, 1111, 1111, 1110, etc. These digital values could then be further processed or analyzed by a purposespecific digital signal processor or general purpose DSP. Several Pulse Code Modulation streams could also be multiplexed into a larger aggregate data stream, generally for transmission of multiple streams over a single physical link. One technique is called time-division multiplexing, or TDM, and is widely used, notably in the modern public telephone system. Another

CONTENTS

1 Modulation 2 Demodulation 3 Limitations 4 Digitization as part of the PCM process 5 Encoding for transmission 6 History 7 Nomenclature 8 See also 9 References 10 Further reading 11 External links

technique is called Frequency-division multiplexing, where the signal is assigned a frequency in a spectrum, and transmitted along with other signals inside that spectrum. Currently, TDM is much more widely used than FDM because of its natural compatibility with digital communication, and generally lower bandwidth requirements. There are many ways to implement a real device that performs this task. In real systems, such a device is commonly implemented on a single integrated circuit that lacks only the clock necessary for sampling, and is generally referred to as an ADC (Analog-to-Digital converter). These devices will produce on their output a binary representation of the input whenever they are triggered by a clock signal, which would then be read by a processor of some sort.

inherent losses in the system compensate for the artifacts or the system simply does not require much precision. The sampling theorem suggests that practical PCM devices, provided a sampling frequency that is sufficiently greater than that of the input signal, can operate without introducing significant distortions within their designed frequency bands. The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. These devices are DACs (digital-to-analog converters), and operate similarly to ADCs. They produce on their output a voltage or current (depending on type) that represents the value presented on their inputs. This output would then generally be filtered and amplified for use.

[EDIT ] LIMITATIONS [EDIT ] DEMODULATION

There are two sources of impairment implicit in any PCM system: To produce output from the sampled data, the procedure of modulation is applied in reverse. After each sampling period has passed, the next value is read and a signal is shifted to the new value. As a result of these transitions, the signal will have a significant amount of high-frequency energy. To smooth out the signal and remove these undesirable aliasing frequencies, the signal would be passed through analog filters that suppress energy outside the expected frequency range (that is, greater than the Nyquist frequency

Choosing a discrete value near the analog signal for each sample leads to quantization error, which swings between -q/2 and q/2. In the ideal case (with a fully linear ADC) it is uniformly distributed over this interval, with zero mean and variance of q2/12. Between samples no measurement of the signal is made; the sampling theorem guarantees non-ambiguous representation and recovery of the signal only if it has no energy at frequency fs/2 or higher (one half the sampling frequency, known as the Nyquist frequency); higher frequencies will generally not be correctly represented or recovered.

fs / 2). Some systems use digital

filtering to remove some of the aliasing, converting the signal from digital to analog at a higher sample rate such that the analog filter required for anti-aliasing is much simpler. In some systems, no explicit filtering is done at all; as it's impossible for any system to reproduce a signal with infinite bandwidth,

As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or decoding clock is not stable, its frequency drift will directly affect the output quality of the device. A slight difference between the encoding and decoding clock frequencies is not generally a major concern; a small constant error is not noticeable. Clock error does become a major issue if the clock is not stable, however. A drifting clock, even with a relatively small error, will cause very obvious distortions in audio and video signals, for example. Extra information: PCM data from a master with a clock frequency that can not be influenced requires an exact clock at the decoding side to ensure that all the data is used in a continuous stream without buffer underrun or buffer overflow. Any frequency difference will be audible at the output since the number of samples per time interval can not be correct. The data speed in a compact disk can be steered by means of a servo that controls the rotation speed of the disk; here the output clock is the master clock. For all "external master" systems like DAB the output stream must be decoded with a regenerated and exact synchronous clock. When the wanted output sample rate differs from the incoming data stream clock then a sample rate converter must be inserted in the chain to convert the samples to the new clock domain.

PCM with linear quantization is known as Linear PCM (LPCM).[1] Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the A/D process; newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques.

DPCM encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM.

Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio.

Delta modulation is a form of DPCM which uses one bit per sample.

In telephony, a standard audio signal for a single phone call is encoded as 8,000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either -law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12 or 13-bit linear PCM sample number is mapped into an 8-bit value. This system is described by international standard G.711. An alternative proposal for a floating point representation, with 5-bit mantissa and 3-bit radix, was abandoned.

[EDIT ] DIGITIZATION AS PART OF THE PCM PROCESS

In conventional PCM, the analog signal may be processed (e.g., by amplitude compression) before being digitized. Once the signal is digitized, the PCM signal is usually subjected to further processing (e.g., digital data compression).

Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8-bit -law or A-law PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Later it was found that even further compression was possible and additional standards were published. Some of these international standards describe systems and ideas which are covered by privately owned patents and thus use of these standards requires payments to the patent holders. Some ADPCM techniques are used in Voice over IP communications.

Another technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend to turn the raw data stream into a stream that looks pseudo-random, but where the raw stream can be recovered exactly by reversing the effect of the polynomial. In this case, long runs of zeroes or ones are still possible on the output, but are considered unlikely enough to be within normal engineering tolerance. In other cases, the long term DC value of the modulated signal is important, as building up a DC offset will tend to bias detector circuits out of their operating range. In this case special measures are taken to keep a count of the cumulative DC offset, and to modify the codes if necessary to make the DC offset always tend back to zero. Many of these codes are bipolar codes, where the pulses can be

[EDIT ] ENCODING FOR TRANSMISSION

Main article: Line code Pulse-code modulation can be either return-to-zero (RZ) or nonreturn-to-zero (NRZ). For a NRZ system to be synchronized using in-band information, there must not be long sequences of identical symbols, such as ones or zeroes. For binary PCM systems, the density of 1-symbols is called ones-density.[2] Ones-density is often controlled using precoding techniques such as Run Length Limited encoding, where the PCM code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into the channel. In other cases, extra framing bits are added into the stream which guarantee at least occasional symbol transitions.

positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes. See also: T-carrier and E-carrier

[EDIT ] HISTORY
In the history of electrical communications, the earliest reason for sampling a signal was to interlace samples from different telegraphy sources, and convey them over a single telegraph cable. Telegraph time-division multiplexing (TDM) was conveyed as early as 1853, by the American inventor Moses B. Farmer. The electrical engineer W. M. Miner, in 1903, used an electromechanical commutator for time-division multiplex of multiple telegraph signals, and also applied this technology to telephony.

He obtained intelligible speech from channels sampled at a rate above 35004300 Hz: below this was unsatisfactory. This was TDM, but pulse-amplitude modulation (PAM) rather than PCM.

produced all bits simultaneously by using a fan beam instead of a scanning beam. The National Inventors Hall of Fame has honored Bernard M.

In 1926, Paul M. Rainey of Western Electric patented a facsimile machine which transmitted its signal using 5-bit PCM, encoded by an opto-mechanical analog-to-digital converter.[3] 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 use resulted. Reeves filed for a French patent in 1938, and his U.S. patent was granted in 1943. The first transmission of speech by digital techniques was the SIGSALY vocoder encryption equipment used for high-level Allied communications during World War II from 1943. 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 Navy's DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances.
[4]

Oliver[7] and Claude Shannon[8] as the inventors of PCM,[9] as described in 'Communication System Employing Pulse Code Modulation,' U.S. Patent 2,801,281 filed in 1946 and 1952, granted in 1956. Another patent by the same title was filed by John R. Pierce in 1945, and issued in 1948: U.S. Patent 2,437,707. The three of them published "The Philosophy of PCM" in 1948.[10] Pulse-code modulation (PCM) was used in Japan by Denon in 1972 for the mastering and production of analogue phonograph records, using a 2-inch Quadruplex-format videotape recorder for its transport, but this was not developed into a consumer product.

[EDIT ] NOMENCLATURE
The word pulse in the term Pulse-Code Modulation refers to the "pulses" to be found in the transmission line. This perhaps is a natural consequence of this technique having evolved alongside two analog methods, pulse width modulation and pulse position modulation, in which the information to be encoded is in fact represented by discrete signal pulses of varying width or position, respectively. In this respect, PCM bears little resemblance to these other forms of signal encoding, except that all can be used in time division multiplexing, and the binary numbers of the PCM codes are represented as electrical pulses. The device that performs the coding and decoding function in a telephone circuit is called a codec.

PCM in the late 1940s and early 1950s used a cathode-ray coding tube with a plate electrode having encoding perforations.
[5][6]

As

in an oscilloscope, the beam was swept horizontally at the sample rate while the vertical deflection was controlled by the input analog signal, causing the beam to pass through higher or lower portions of the perforated plate. The plate collected or passed the beam, producing current variations in binary code, one bit at a time. Rather than natural binary, the grid of Goodall's later tube was perforated to produce a glitch-free Gray code, and

OPTICAL FIBER
An optical fiber or optical fibre is a thin, flexible, transparent fiber that acts as a waveguide, or "light pipe", to transmit light between the two ends of the fiber. The field of applied science and engineering concerned with the design and application of optical fibers is known as fiber optics. Optical fibers are widely used in fiber-optic communications, which permits transmission over longer distances and at higher bandwidths (data rates) than other forms of communication. Fibers are used instead of metal wires because signals travel along them with less loss and are also immune to electromagnetic interference. Fibers are also used for illumination, and are wrapped in bundles so they can be used to carry images, thus allowing viewing in tight spaces. Specially designed fibers are used for a variety of other applications, including sensors and fiber lasers. Optical fiber typically consists of a transparent core surrounded by a transparent cladding material with a lower index of refraction. Light is kept in the core by total internal reflection. This causes the fiber to act as a waveguide. Fibers which support many propagation paths or transverse modes are called multimode fibers (MMF), while those which can only support a single mode are called single-mode fibers (SMF). Multi-mode fibers generally have a larger core diameter, and are used for shortdistance communication links and for applications where high power must be transmitted. Single-mode fibers are used for most communication links longer than 1,050 meters (3,440 ft). Joining lengths of optical fiber is more complex than joining electrical wire or cable. The ends of the fibers must be carefully cleaved, and then spliced together either mechanically or by fusing them together with heat. Special optical fiber connectors are used to make removable connections

[EDIT] OPTICAL FIBER COMMUNICATION

Main article: Fiber-optic communication Optical fiber can be used as a medium for telecommunication and networking because it is flexible and can be bundled as cables. It is especially advantageous for long-distance communications, because light propagates through the fiber with little attenuation compared to electrical cables. This allows long distances to be spanned with few repeaters. Additionally, the per-channel light signals propagating in the fiber have been modulated at rates as high as 111 gigabits per second by NTT,[15]
[16]

although 10 or 40 Gbit/s is typical in deployed systems.[17][18]

Each fiber can carry many independent channels, each using a different wavelength of light (wavelength-division multiplexing (WDM)). The net data rate (data rate without overhead bytes) per fiber is the per-channel data rate reduced by the FEC overhead, multiplied by the number of channels (usually up to eighty in commercial dense WDM systems as of 2008). The current laboratory fiber optic data rate record, held by Bell Labs in Villarceaux, France, is multiplexing 155 channels, each carrying 100 Gbit/s over a 7000 km fiber.[19] Nippon Telegraph and Telephone Corporation have also managed 69.1 Tbit/s over a single 240 km fiber (multiplexing 432 channels, equating to 171 Gbit/s per channel).[20] Bell Labs also broke a 100 Petabit per second kilometer barrier (15.5 Tbit/s over a single 7000 km fiber).[21] For short distance applications, such as creating a network within an office building, fiber-optic cabling can be used to save space in cable ducts. This is because a single fiber can often carry much more data than many electrical cables, such as 4 pair Cat5 Ethernet cabling.[vague] Fiber is also immune to electrical

interference; there is no cross-talk between signals in different cables and no pickup of environmental noise. Non-armored fiber cables do not conduct electricity, which makes fiber a good solution for protecting communications equipment located in high voltage environments such as power generation facilities, or metal communication structures prone to lightning strikes. They can also be used in environments where explosive fumes are present, without danger of ignition. Wiretapping is more difficult compared to electrical connections, and there are concentric dual core fibers that are said to be tap-proof.[22]

particularly useful feature of such fiber optic sensors is that they can, if required, provide distributed sensing over distances of up to one meter. Extrinsic fiber optic sensors use an optical fiber cable, normally a multi-mode one, to transmit modulated light from either a nonfiber optical sensor, or an electronic sensor connected to an optical transmitter. A major benefit of extrinsic sensors is their ability to reach places which are otherwise inaccessible. An example is the measurement of temperature inside aircraft jet engines by using a fiber to transmit radiation into a radiation pyrometer located outside the engine. Extrinsic sensors can also be used in the same way to measure the internal temperature of electrical transformers, where the extreme electromagnetic fields present make other measurement techniques impossible. Extrinsic sensors are used to measure vibration, rotation, displacement, velocity, acceleration, torque, and twisting. A solid state version of the gyroscope using the interference of light has been developed. The fiber optic gyroscope (FOG) has no moving parts and exploits the Sagnac effect to detect mechanical rotation. A common use for fiber optic sensors are in advanced intrusion detection security systems, where the light is transmitted along the fiber optic sensor cable, which is placed on a fence, pipeline or communication cabling, and the returned signal is monitored and analysed for disturbances. This return signal is digitally processed to identify if there is a disturbance, and if an intrusion has occurred an alarm is triggered by the fiber optic security system.

[EDIT] FIBER OPTIC SENSORS

Main article: Fiber optic sensor Fibers have many uses in remote sensing. In some applications, the sensor is itself an optical fiber. In other cases, fiber is used to connect a non-fiberoptic sensor to a measurement system. Depending on the application, fiber may be used because of its small size, or the fact that no electrical power is needed at the remote location, or because many sensors can be multiplexed along the length of a fiber by using different wavelengths of light for each sensor, or by sensing the time delay as light passes along the fiber through each sensor. Time delay can be determined using a device such as an optical time-domain reflectometer. Optical fibers can be used as sensors to measure strain, temperature, pressure and other quantities by modifying a fiber so that the quantity to be measured modulates the intensity, phase, polarization, wavelength or transit time of light in the fiber. Sensors that vary the intensity of light are the simplest, since only a simple source and detector are required. A

] PRINCIPLE OF OPERATION
An optical fiber is a cylindrical dielectric waveguide (nonconducting waveguide) that transmits light along its axis, by

the process of total internal reflection. The fiber consists of a core surrounded by a cladding layer, both of which are made of dielectric materials. To confine the optical signal in the core, the refractive index of the core must be greater than that of the cladding. The boundary between the core and cladding may either be abrupt, in step-index fiber, or gradual, in graded-index fiber

WIRELESS
From Wikipedia, the free encyclopedia In telecommunications, wireless communication may be used to transfer information over short distances (a few meters as in television remote control) or long distances (thousands or millions of kilometers for radio communications). The term is often shortened to "wireless". It encompasses various types of fixed, mobile, and portable two-way radios, cellular telephones, personal digital assistants (PDAs), and wireless networking. Other examples of wireless technology include GPS units, garage door openers and or garage doors, wireless computer mice, keyboards and headsets, satellite television and cordless telephones.

8.2 Cellular telephone (phones and modems) 8.3 Wi-Fi 8.4 Wireless energy transfer 8.5 Computer interface devices 9 Categories of wireless implementations, devices and standards 10 See also 11 References 12 Further reading
o o o o

13 External links

[EDIT ] INTRODUCTION
Handheld wireless radios such as this Maritime VHF radio transceiver use electromagnetic waves to implement a form of wireless communications technology. Wireless operations permits services, such as long range communications, that are impossible or impractical to implement with the use of wires. The term is commonly used in the telecommunications industry to refer to telecommunications systems (e.g. radio transmitters and receivers, remote controls, computer networks, network terminals, etc.) which use some form of energy (e.g. radio frequency (RF), infrared light, laser light, visible light, acoustic energy, etc.) to transfer information without the use of wires.[1] Information is transferred in this manner over both short and long distances.

CONTENTS

1 2 3 4 5 6

Introduction Wireless services Wireless networks Modes Cordless History o 6.1 Photophone o 6.2 Early wireless work o 6.3 Radio 7 The electromagnetic spectrum 8 Applications of wireless technology o 8.1 Security systems

[EDIT ] WIRELESS SERVICES

The term "wireless" has become a generic and all-encompassing word used to describe communications in which electromagnetic waves or RF (rather than some form of wire) carry a signal over

part or the entire communication path. Common examples of wireless equipment in use today include:

connect via satellite. A wireless transmission method is a logical choice to network a LAN segment that must frequently change locations. The following situations justify the use of wireless technology:

Professional LMR (Land Mobile Radio) and SMR (Specialized Mobile Radio) typically used by business, industrial and Public Safety entities.

To span a distance beyond the capabilities of typical cabling, To provide a backup communications link in case of normal network failure, To link portable or temporary workstations, To overcome situations where normal cabling is difficult or financially impractical, or To remotely connect mobile users or networks.

Consumer Two way radio including FRS Family Radio Service, GMRS (General Mobile Radio Service) and Citizens band ("CB") radios.

The Amateur Radio Service (Ham radio). Consumer and professional Marine VHF radios. Cellular telephones and pagers: provide connectivity for portable and mobile applications, both personal and business.

Global Positioning System (GPS): allows drivers of cars and trucks, captains of boats and ships, and pilots of aircraft to ascertain their location anywhere on earth.

[EDIT ] MODES
Wireless communication can be via:

Cordless computer peripherals: the cordless mouse is a common example; keyboards and printers can also be linked to a computer via wireless. radio frequency communication, microwave communication, for example long-range line-ofsight via highly directional antennas, or short-range communication, or

Cordless telephone sets: these are limited-range devices, not to be confused with cell phones. Satellite television: Is broadcast from satellites in geostationary orbit. Typical services use digital broadcasting to provide multiple channels to viewers.

infrared (IR) short-range communication, for example from remote controls or via Infrared Data Association (IrDA).

Applications may involve point-to-point communication, point-tomultipoint communication, broadcasting, cellular networks and other wireless networks.

[EDIT ] WIRELESS NETWORKS

Wireless networking (i.e. the various types of unlicensed 2.4 GHz WiFi devices) is used to meet many needs. Perhaps the most common use is to connect laptop users who travel from location to location. Another common use is for mobile networks that

[EDIT ] CORDLESS
The term "wireless" should not be confused with the term "cordless", which is generally used to refer to powered electrical

or electronic devices that are able to operate from a portable power source (e.g. a battery pack) without any cable or cord to limit the mobility of the cordless device through a connection to the mains power supply. Some cordless devices, such as cordless telephones, are also wireless in the sense that information is transferred from the cordless telephone to the telephone's base unit via some type of wireless communications link. This has caused some disparity in the usage of the term "cordless", for example in Digital Enhanced Cordless Telecommunications.

[EDIT] EARLY WIRELESS WORK

Main article: Wireless telegraphy David E. Hughes, eight years before Hertz's experiments, transmitted radio signals over a few hundred yards by means of a clockwork keyed transmitter. As this was before Maxwell's work was understood, Hughes' contemporaries dismissed his achievement as mere "Induction". In 1885, T. A. Edison used a vibrator magnet for induction transmission. In 1888, Edison deployed a system of signaling on the Lehigh Valley Railroad. In 1891, Edison obtained the wireless patent for this method using inductance (U.S. Patent 465,971). In the history of wireless technology, the demonstration of the theory of electromagnetic waves by Heinrich Hertz in 1888 was important.[2][3] The theory of electromagnetic waves was predicted from the research of James Clerk Maxwell and Michael Faraday. Hertz demonstrated that electromagnetic waves could be transmitted and caused to travel through space at straight lines and that they were able to be received by an experimental apparatus.[2][3] The experiments were not followed up by Hertz. Jagadish Chandra Bose around this time developed an early wireless detection device and helped increase the knowledge of millimeter length electromagnetic waves.[4] Practical applications of wireless radio communication and radio remote control technology were implemented by later inventors, such as Nikola Tesla. Further information: Invention of radio

[EDIT ] HISTORY [EDIT] PHOTOPHONE

Main article: Photophone The world's first, wireless telephone conversation occurred in 1880, when Alexander Graham Bell and Charles Sumner Tainter invented and patented the photophone, a telephone that conducted audio conversations wirelessly over modulated light beams (which are narrow projections of electromagnetic waves). In that distant era when utilities did not yet exist to provide electricity, and lasers had not even been conceived of in science fiction, there were no practical applications for their invention, which was highly limited by the availability of both sunlight and good weather. Similar to free space optical communication, the photophone also required a clear line of sight between its transmitter and its receiver. It would be several decades before the photophone's principles found their first practical applications in military communications and later in fiber-optic communications.

[EDIT] RADIO

Main article: History of radio The term "wireless" came into public use to refer to a radio receiver or transceiver (a dual purpose receiver and transmitter device), establishing its usage in the field of wireless telegraphy early on; now the term is used to describe modern wireless connections such as in cellular networks and wireless broadband Internet. It is also used in a general sense to refer to any type of operation that is implemented without the use of wires, such as "wireless remote control" or "wireless energy transfer", regardless of the specific technology (e.g. radio, infrared, ultrasonic) used. Guglielmo Marconi and Karl Ferdinand Braun were awarded the 1909 Nobel Prize for Physics for their contribution to wireless telegraphy.

[EDIT] SECURITY SYSTEMS

Wireless technology may supplement or replace hard wired implementations in security systems for homes or office buildings.

[EDIT] CELLULAR TELEPHONE (PHONES AND MODEMS)

Perhaps the best known example of wireless technology is the cellular telephone and modems. These instruments use radio waves to enable the operator to make phone calls from many locations worldwide. They can be used anywhere that there is a cellular telephone site to house the equipment that is required to transmit and receive the signal that is used to transfer both voice and data to and from these instruments.

[EDIT ] THE ELECTROMAGNETIC SPECTRUM

Light, colors, AM and FM radio, and electronic devices make use of the electromagnetic spectrum. In the US, the frequencies that are available for use for communication are treated as a public resource and are regulated by the Federal Communications Commission. This determines which frequency ranges can be used for what purpose and by whom. In the absence of such control or alternative arrangements such as a privatized electromagnetic spectrum, chaos might result if, for example, airlines didn't have specific frequencies to work under and an amateur radio operator were interfering with the pilot's ability to land an airplane. Wireless communication spans the spectrum from 9 kHz to 300 GHz. (Also see Spectrum management)

[EDIT] WI-FI
Main article: Wi-Fi Wi-Fi is a wireless local area network that enables portable computing devices to connect easily to the Internet. Standardized as IEEE 802.11 a,b,g,n, Wi-Fi approaches speeds of some types of wired Ethernet. Wi-Fi hot spots have been popular over the past few years. Some businesses charge customers a monthly fee for service, while others have begun offering it for free in an effort to increase the sales of their goods.[5]

[EDIT] WIRELESS ENERGY TRANSFER

Main article: Wireless energy transfer

[EDIT ] APPLICATIONS OF WIRELESS TECHNOLOGY

Wireless energy transfer is a process whereby electrical energy is transmitted from a power source to an electrical load that does

not have a built-in power source, without the use of interconnecting wires.

List of emerging technologies Short-range point-to-point communication : Wireless microphones, Remote controls, IrDA, RFID (Radio Frequency Identification), Wireless USB, DSRC (Dedicated Short Range Communications), EnOcean, Near Field Communication

[EDIT] COMPUTER INTERFACE DEVICES

Answering the call of customers frustrated with cord clutter, many manufactures of computer peripherals turned to wireless technology to satisfy their consumer base. Originally these units used bulky, highly limited transceivers to mediate between a computer and a keyboard and mouse, however more recent generations have used small, high quality devices, some even incorporating Bluetooth. These systems have become so ubiquitous that some users have begun complaining about a lack of wired peripherals.[who?] Wireless devices tend to have a slightly slower response time than their wired counterparts, however the gap is decreasing. Initial concerns about the security of wireless keyboards have also been addressed with the maturation of the technology.

Wireless sensor networks: ZigBee, EnOcean; Personal area networks, Bluetooth, TransferJet, Ultra-wideband (UWB from WiMedia Alliance). Wireless networks: Wireless LAN (WLAN), (IEEE 802.11 branded as Wi-Fi and HiperLAN), Wireless Metropolitan Area Networks (WMAN) and Broadband Fixed Access (BWA) (LMDS, WiMAX, AIDAAS and HiperMAN)

MICROWAVE TRANSMISSION
From Wikipedia, the free encyclopedia

[EDIT ] CATEGORIES OF WIRELESS IMPLEMENTATIONS, DEVICES AND STANDARDS

Radio communication system Broadcasting Amateur radio Land Mobile Radio or Professional Mobile Radio: TETRA, P25, OpenSky, EDACS, DMR, dPMR Communication radio Cordless telephony:DECT (Digital Enhanced Cordless Telecommunications) Cellular networks: 0G, 1G, 2G, 3G, Beyond 3G (4G), Future wireless The atmospheric attenuation of microwaves in dry air with a precipitable water vapor level of 0.001 mm. The downward spikes in the graph correspond to frequencies at which

microwaves are absorbed more strongly, such as by oxygen molecules Microwave transmission refers to the technology of transmitting information by the use of radio waves whose wavelengths are conveniently measured in small numbers of centimeters, by using various electronic technologies. These are called microwaves. This part of the radio spectrum ranges across frequencies of roughly 1.0 gigahertz (GHz) to 30 GHz. These correspond to wavelengths from 30 centimeters down to 1.0 cm. In the microwave frequency band, antennas are usually of convenient sizes and shapes, and also the use of metal waveguides for carrying the radio power works well. Furthermore, with the use of the modern solid-state electronics and traveling wave tube technologies that have been developed since the early 1960s, the electronics used by microwave radio transmission have been readily used by expert electronics engineers. Microwave radio transmission is commonly used by communication systems on the surface of the Earth, in satellite communications, and in deep space radio communications. Other parts of the microwave radio band are used for radars, radio navigation systems, sensor systems, and radio astronomy. The next higher part of the radio electromagnetic spectrum, where the frequencies are above 30 GHz and below 100 GHz, are called "millimeter waves" because their wavelengths are conveniently measured in millimeters, and their wavelengths range from 10 mm down to 3.0 mm. Radio waves in this band are usually strongly attenuated by the Earthly atmosphere and particles contained in it, especially during wet weather. Also, in

wide band of frequencies around 60 GHz, the radio waves are strongly attenuated by molecular oxygen in the atmosphere. The electronic technologies needed in the millimeter wave band are also much more difficult to utilize than those of the microwave band.

CONTENTS

1 2 3 4

Properties Uses Parabolic (microwave) antenna Microwave power transmission o 4.1 History o 4.2 Common safety concerns o 4.3 Proposed uses o 4.4 Current status 5 Microwave radio relay o 5.1 How microwave radio relay links are formed o 5.2 Planning considerations o 5.3 Over-horizon microwave radio relay o 5.4 Usage of microwave radio relay systems o 5.5 Microwave link 5.5.1 Properties of microwave links 5.5.2 Uses of microwave links o 5.6 Tunable microwave device 6 See also 7 References 8 External links

[EDIT ] PROPERTIES

Suitable over line-of-sight transmission links without obstacles Provides good bandwidth[clarification needed]

Affected by rain, vapor, dust, snow, cloud, mist and fog, heavy moisture, depending on chosen frequency (see rain fade)

[EDIT ] MICROWAVE POWER TRANSMISSION

Microwave power transmission (MPT) is the use of microwaves to transmit power through outer space or the

[EDIT ] USES

atmosphere without the need for wires. It is a sub-type of the more general wireless energy transfer methods.

Backbone or backhaul carriers in cellular networks. Used to link BTS-BSC and BSC-MSC. Communication with satellites Microwave radio relay links for television and telephone service providers

[EDIT] HISTORY
Following World War II, which saw the development of highpower microwave emitters known as cavity magnetrons, the idea of using microwaves to transmit power was researched. In 1964, William C. Brown demonstrated a miniature helicopter equipped with a combination antenna and rectifier device called a rectenna. The rectenna converted microwave power into electricity, allowing the helicopter to fly.[1] In principle, the rectenna is capable of very high conversion efficiencies - over 90% in optimal circumstances. Most proposed MPT systems now usually include a phased array microwave transmitter. While these have lower efficiency levels they have the advantage of being electrically steered using no moving parts, and are easier to scale to the necessary levels that a practical MPT system requires. Using microwave power transmission to deliver electricity to communities without having to build cable-based infrastructure is being studied at Grand Bassin on Reunion Island in the Indian Ocean.

[EDIT ] PARABOLIC (MICROWAVE) ANTENNA

Main article: Parabolic antenna A parabolic antenna is a high-gain reflector antenna used for radio, television and data communications, and also for radiolocation (radar), on the UHF and SHF parts of the electromagnetic spectrum. The relatively short wavelength of electromagnetic radiation at these frequencies allows reasonably sized reflectors to exhibit the desired highly directional response for both receiving and transmitting.

[EDIT] COMMON SAFETY CONCERNS

The common reaction to microwave transmission is one of concern, as microwaves are generally perceived by the public as dangerous forms of radiation - stemming from the fact that they are used in microwave ovens. While high power microwaves can be painful and dangerous as in the United States Military's Active Denial System, MPT systems are generally proposed to have only low intensity at the rectenna.

Wireless Power Transmission (using microwaves) is well proven. Experiments in the tens of kilowatts have been performed at Goldstone in California in 1975[3][4][5] and more recently (1997) at Grand Bassin on Reunion Island.[6] In 2008 a long range transmission experiment successfully transmitted 20 watts 92 miles from a mountain on Maui to the main island of Hawaii.[7] [edit] Microwave radio relay

Though this would be extremely safe as the power levels would be about equal to the leakage from a microwave oven, and only slightly more than a cell phone, the relatively diffuse microwave beam necessitates a large rectenna area for a significant amount of energy to be transmitted. Research has involved exposing multiple generations of animals to microwave radiation of this or higher intensity, and no health issues have been found.[2]

[EDIT] PROPOSED USES

Main article: Solar power satellite MPT is the most commonly proposed method for transferring energy to the surface of the Earth from solar power satellites or other in-orbit power sources. MPT is occasionally proposed for the power supply in [beam-powered propulsion] for orbital lift space ships. Even though lasers are more commonly proposed, their low efficiency in light generation and reception has led some designers to opt for microwave based systems.

Heinrich-Hertz-Turm in Germany Microwave radio relay is a technology for transmitting digital and analog signals, such as long-distance telephone calls and the relay of television programs to transmitters, between two locations on a line of sight radio path. In microwave radio relay, radio waves are transmitted between the two locations with directional antennas, forming a fixed radio connection between the two points. Long daisy-chained series of such links form transcontinental telephone and/or television communication systems.

[EDIT] CURRENT STATUS

[EDIT] HOW MICROWAVE RADIO RELAY LINKS ARE FORMED

Danish military radio relay node

[EDIT] PLANNING CONSIDERATIONS

Because of the high frequencies used, a quasi-optical line of sight between the stations is generally required. Additionally, in order to form the line of sight connection between the two stations, the first Fresnel zone must be free from obstacles so the radio waves can propagate across a nearly uninterrupted path. Obstacles in the signal field cause unwanted attenuation, and are as a result Relay towers on Frazier Mountain, Southern California Because a line of sight radio link is made, the radio frequencies used occupy only a narrow path between stations (with the exception of a certain radius of each station). Antennas used must have a high directive effect; these antennas are installed in elevated locations such as large radio towers in order to be able to transmit across long distances. Typical types of antenna used in radio relay link installations are parabolic reflectors, shell antennas and horn radiators, which have a diameter of up to 4 meters. Highly directive antennas permit an economical use of the available frequency spectrum, despite long transmission distances. Multiple antennas provide space diversity Obstacles, the curvature of the Earth, the geography of the area and reception issues arising from the use of nearby land (such as in manufacturing and forestry) are important issues to consider when planning radio links. In the planning process, it is essential that "path profiles" are produced, which provide information about the terrain and Fresnel zones affecting the transmission only acceptable in exceptional cases. High mountain peak or ridge positions are often ideal: Europe's highest radio relay station, the Richtfunkstation Jungfraujoch, is situated atop the Jungfraujoch ridge at an altitude of 3,705 meters (12,156 ft) above sea level.

path. The presence of a water surface, such as a lake or river, in the mid-path region also must be taken into consideration as it can result in a near-perfect reflection (even modulated by wave or tide motions), creating multipath distortion as the two received signals ("wanted" and "unwanted") swing in and out of phase. Multipath fades are usually deep only in a small spot and a narrow frequency band, so space and frequency diversity schemes were usually applied in the third quarter of the 20th century. The effects of atmospheric stratification cause the radio path to bend downward in a typical situation so a major distance is possible as the earth equivalent curvature increases from 6370 km to about 8500 km (a 4/3 equivalent radius effect). Rare events of temperature, humidity and pressure profile versus height, may produce large deviations and distortion of the propagation and affect transmission quality. High intensity rain and snow must also be considered as an impairment factor, especially at frequencies above 10 GHz. All previous factors, collectively known as path loss, make it necessary to compute suitable power margins, in order to maintain the link operative for a high percentage of time, like the standard 99.99% or 99.999% used in 'carrier class' services of most telecommunication operators. Portable microwave rig for television news

[EDIT] OVER-HORIZON MICROWAVE RADIO RELAY

In over-horizon, or tropospheric scatter, microwave radio relay, unlike a standard microwave radio relay link, the sending and receiving antennas do not use a line of sight transmission path. Instead, the stray signal transmission, known as "tropo - scatter" or simply "scatter," from the sent signal is picked up by the receiving station. Signal clarity obtained by this method depends on the weather and other factors, and as a result a high level of technical difficulty is involved in the creation of a reliable over horizon radio relay link. Over horizon radio relay links are therefore only used where standard radio relay links are unsuitable (for example, in providing a microwave link to an island).

[EDIT] USAGE OF MICROWAVE RADIO RELAY SYSTEMS

During the 1950s the AT&T Communications system of microwave radio grew to carry the majority of US Long Distance

telephone traffic, as well as intercontinental television network signals. The prototype was called TDX and was tested with a connection between New York City and Murray Hill, the location of Bell Laboratories in 1946. The TDX system was set up between New York and Boston in 1947. The TDX was improved to the TD2, which still used klystrons, and then later to the TD3 that used solid state electronics. The main motivation in 1946 to use microwave radio instead of cable was that a large capacity could be installed quickly and at less cost. It was expected at that time that the annual operating costs for microwave radio would be greater than for cable. There were two main reasons that a large capacity had to be introduced suddenly: Pent up demand for long distance telephone service, because of the hiatus during the war years, and the new medium of television, which needed more bandwidth than radio. Similar systems were soon built in many countries, until the 1980s when the technology lost its share of fixed operation to newer technologies such as fiber-optic cable and optical radio relay links, both of which offer larger data capacities at lower cost per bit. Communication satellites, which are also microwave radio relays, better retained their market share, especially for television. At the turn of the century, microwave radio relay systems are being used increasingly in portable radio applications. The technology is particularly suited to this application because of lower operating costs, a more efficient infrastructure, and provision of direct hardware access to the portable radio operator.

A microwave link is a communications system that uses a beam of radio waves in the microwave frequency range to transmit video, audio, or data between two locations, which can be from just a few feet or meters to several miles or kilometers apart. Microwave links are commonly used by television broadcasters to transmit programmes across a country, for instance, or from an outside broadcast back to a studio. Mobile units can be camera mounted, allowing cameras the freedom to move around without trailing cables. These are often seen on the touchlines of sports fields on Steadicam systems. [EDIT ] PROPERTIES OF MICROWAVE LINKS

Involve line of sight (LOS) communication technology Affected greatly by environmental constraints, including rain fade Have limited penetration capabilities Sensitive to high pollen count Signals can be degraded during Solar proton events [EDIT ] USES OF MICROWAVE LINKS
[8]

In communications between satellites and base stations As backbone carriers for cellular systems In short range indoor communications

[EDIT] TUNABLE MICROWAVE DEVICE

A tunable microwave device is a device that works at radio frequency range with the dynamic tunable capabilities, especially an electric field. The material systems for such a device usually have multilayer structure. Usually, magnetic or ferroelectric film

[EDIT] MICROWAVE LINK

on ferrite or superconducting film is adopted. The former two are used as the property tunable component to control the working frequency of the whole system. Devices of this type include tunable varators, tunable microwave filters, tunable phase shifters, and tunable resonators. The main application of them is re-configurable microwave networks, for example, reconfigurable wireless communication, wireless network, and reconfigurable phase array antenna.[9][10]

has a much higher data bandwidth than the data being communicated. An analogy to the problem of multiple access is a room (channel) in which people wish to talk to each other simultaneously. To avoid confusion, people could take turns speaking (time division), speak at different pitches (frequency division), or speak in different languages (code division). CDMA is analogous to the last example where people speaking the same language can understand each other, but other languages are perceived as noise and rejected. Similarly, in radio CDMA, each group of users is given a shared code. Many codes occupy the same channel, but only users associated with a particular code can communicate.

CODE DIVISION MULTIPLE ACCESS

From Wikipedia, the free encyclopedia Code division multiple access (CDMA) is a channel access method used by various radio communication technologies. It should not be confused with the mobile phone standards called cdmaOne and CDMA2000 (which are often referred to as simply CDMA), which use CDMA as an underlying channel access method. One of the basic concepts in data communication is the idea of allowing several transmitters to send information simultaneously over a single communication channel. This allows several users to share a band of frequencies (see bandwidth). This concept is called Multiple Access. CDMA employs spread-spectrum technology and a special coding scheme (where each transmitter is assigned a code) to allow multiple users to be multiplexed over the same physical channel. By contrast, time division multiple access (TDMA) divides access by time, while frequency-division multiple access (FDMA) divides it by frequency. CDMA is a form of spread-spectrum signalling, since the modulated coded signal

CONTENTS

1 Uses 2 Steps in CDMA Modulation 3 Code division multiplexing (Synchronous CDMA) o 3.1 Example 4 Asynchronous CDMA o 4.1 Advantages of asynchronous CDMA over other techniques o 4.2 Spread-spectrum characteristics of CDMA 5 See also 6 References 7 External links

[EDIT ] USES
One of the early applications for code division multiplexing is in GPS. This predates and is distinct from cdmaOne.

The Qualcomm standard IS-95, marketed as cdmaOne. The Qualcomm standard IS-2000, known as CDMA2000. This standard is used by several mobile phone companies, including the Globalstar satellite phone network.

CDMA has been used in the OmniTRACS satellite system for transportation logistics.

[EDIT ] STEPS IN CDMA MODULATION

CDMA is a spread spectrum multiple access[1] technique. A spread spectrum technique spreads the bandwidth of the data uniformly for the same transmitted power. Spreading code is a pseudo-random code that has a narrow Ambiguity function, unlike other narrow pulse codes. In CDMA a locally generated code runs at a much higher rate than the data to be transmitted. Data for transmission is combined via bitwise XOR (exclusive OR) with the faster code. The figure shows how spread spectrum signal is generated. The data signal with pulse duration of XORed with the code signal with pulse duration of bandwidth is proportional to Each user in a CDMA system uses a different code to modulate their signal. Choosing the codes used to modulate the signal is very important in the performance of CDMA systems. The best performance will occur when there is good separation between the signal of a desired user and the signals of other users. The separation of the signals is made by correlating the received signal with the locally generated code of the desired user. If the signal matches the desired user's code then the correlation function will be high and the system can extract that signal. If the desired user's code has nothing in common with the signal the correlation should be as close to zero as possible (thus eliminating the signal); this is referred to as cross correlation. If the code is correlated with the signal at any time offset other than zero, the correlation should be as close to zero as possible. This is referred to as auto-correlation and is used to reject multipath interference.[3] In general, CDMA belongs to two basic categories: synchronous (orthogonal codes) and asynchronous (pseudorandom codes).

Tb is

Tc. (Note:

1 / T where T = bit time) Therefore, the bandwidth of the data signal is 1 / Tb and the bandwidth of the spread spectrum signal is 1 / Tc. Since Tc is much smaller than Tb, the bandwidth of the spread spectrum signal is much larger than the bandwidth of the original signal. The ratio Tb / Tc
is called spreading factor or processing gain and determines to a certain extent the upper limit of the total number of users supported simultaneously by a base station.
[2]

[EDIT ] CODE DIVISION MULTIPLEXING (SYNCHRONOUS CDMA)

Synchronous CDMA exploits mathematical properties of orthogonality between vectors representing the data strings. For example, binary string 1011 is represented by the vector (1, 0, 1, 1). Vectors can be multiplied by taking their dot product, by summing the products of their respective components. If the dot product is zero, the two vectors are said to be orthogonal to each other (note: if u = (a, b) and v = (c, d), the dot product uv = ac + bd). Some properties of the dot product aid understanding of how W-CDMA works. If vectors a and b are orthogonal, then and:

An example of four mutually orthogonal digital signals. Each user in synchronous CDMA uses a code orthogonal to the others' codes to modulate their signal. An example of four mutually orthogonal digital signals is shown in the figure. Orthogonal codes have a cross-correlation equal to zero; in other words, they do not interfere with each other. In the case of IS-95 64 bit Walsh codes are used to encode the signal to separate different users. Since each of the 64 Walsh codes are orthogonal to one another, the signals are channelized into 64 orthogonal signals. The following example demonstrates how each user's signal can be encoded and decoded. [EDIT ] EXAMPLE Start with a set of vectors that are mutually orthogonal. (Although mutual orthogonality is the only condition, these vectors are usually constructed for ease of decoding, for example columns or rows from Walsh matrices.) An example of orthogonal functions is shown in the picture on the left. These vectors will be assigned to individual users and are called the code, chip code, or chipping code. In the interest of brevity, the rest of this example uses codes, v, with only 2 bits. Each user is associated with a different code, say v. A 1 bit is represented by transmitting a positive code, v, and a 0 bit is represented by a negative code, v. For example, if v = (1, 1) and the data that the user wishes to transmit is (1, 0, 1, 1), then

the transmitted symbols would be (1, 1, 1, 1) v = (v0, v1, v0, v1, v0, v1, v0, v1) = (1, 1, 1, 1, 1, 1, 1, 1), where is the Kronecker product. For the purposes of this article, we call this constructed vector the transmitted vector. Each sender has a different, unique vector v chosen from that set, but the construction method of the transmitted vector is identical. Now, due to physical properties of interference, if two signals at a point are in phase, they add to give twice the amplitude of each signal, but if they are out of phase, they subtract and give a signal that is the difference of the amplitudes. Digitally, this behaviour can be modelled by the addition of the transmission vectors, component by component. If sender0 has code (1, 1) and data (1, 0, 1, 1), and sender1 has code (1, 1) and data (0, 0, 1, 1), and both senders transmit simultaneously, then this table describes the coding steps: Ste p 0 1 Encode sender0 Encode sender1 3 4 1 2 Ste p 0

(1, 1, 1, 1, 1, 1, 1, 1) + (1, 1, 1, 1, 1, 1, 1, 1) = (0, 2, 2, 0, 2, 0, 2, 0) This raw signal is called an interference pattern. The receiver then extracts an intelligible signal for any known sender by combining the sender's code with the interference pattern, the receiver combines it with the codes of the senders. The following table explains how this works and shows that the signals do not interfere with one another: Decode sender0 Decode sender1

code0 = (1, 1), signal = (0, code1 = (1, 1), signal = (0, 2, 2, 2, 0, 2, 0, 2, 0) 2, 0, 2, 0, 2, 0) decode0 = pattern.vector0 decode1 = pattern.vector1

decode0 = ((0, 2), (2, 0), (2, decode1 = ((0, 2), (2, 0), (2, 0), (2, 0)).(1, 1) 0), (2, 0)).(1, 1) decode0 = ((0 + 2), (2 + 0), (2 + 0), (2 + 0)) data0=(2, 2, 2, 2), meaning (1, 0, 1, 1) decode1 = ((0 2), (2 + 0), (2 + 0), (2 + 0)) data1=(2, 2, 2, 2), meaning (0, 0, 1, 1)

code0 = (1, 1), data0 = (1, code1 = (1, 1), data1 = (0, 0, 0, 1, 1) 1, 1) encode0 = 2(1, 0, 1, 1) (1, encode1 = 2(0, 0, 1, 1) (1, 1, 1, 1) 1, 1, 1) = (1, 1, 1, 1) signal0 = encode0 code0 = (1, 1, 1, 1) (1, 1) = (1, 1, 1, 1, 1, 1, 1, 1) = (1, 1, 1, 1) signal1 = encode1 code1 = (1, 1, 1, 1) (1, 1) = (1, 1, 1, 1, 1, 1, 1, 1)

Further, after decoding, all values greater than 0 are interpreted as 1 while all values less than zero are interpreted as 0. For example, after decoding, data0 is (2, 2, 2, 2), but the receiver interprets this as (1, 0, 1, 1). Values of exactly 0 means that the sender did not transmit any data, as in the following example: Assume signal0 = (1, 1, 1, 1, 1, 1, 1, 1) is transmitted alone. The following table shows the decode at the receiver:

Because signal0 and signal1 are transmitted at the same time into the air, they add to produce the raw signal:

Ste p 0 1 2 3 4

Decode sender0

Decode sender1

On the other hand, the mobile-to-base links cannot be precisely coordinated, particularly due to the mobility of the handsets, and require a somewhat different approach. Since it is not mathematically possible to create signature sequences that are both orthogonal for arbitrarily random starting points and which make full use of the code space, unique "pseudo-random" or "pseudo-noise" (PN) sequences are used in asynchronous CDMA systems. A PN code is a binary sequence that appears random but can be reproduced in a deterministic manner by intended receivers. These PN codes are used to encode and decode a user's signal in Asynchronous CDMA in the same manner as the orthogonal codes in synchronous CDMA (shown in the example above). These PN sequences are statistically uncorrelated, and the sum of a large number of PN sequences results in multiple access interference (MAI) that is approximated by a Gaussian noise process (following the central limit theorem in statistics). Gold codes are an example of a PN suitable for this purpose, as there is low correlation between the codes. If all of the users are received with the same power level, then the variance (e.g., the noise power) of the MAI increases in direct proportion to the number of users. In other words, unlike synchronous CDMA, the signals of other users will appear as noise to the signal of interest and interfere slightly with the desired signal in proportion to number of users. All forms of CDMA use spread spectrum process gain to allow receivers to partially discriminate against unwanted signals. Signals encoded with the specified PN sequence (code) are received, while signals with different codes (or the same code but a different timing offset) appear as wideband noise reduced by the process gain.

code0 = (1, 1), signal = (1, code1 = (1, 1), signal = (1, 1, 1, 1, 1, 1, 1, 1, 1) 1, 1, 1, 1, 1, 1) decode0 = pattern.vector0 decode1 = pattern.vector1

decode0 = ((1, 1), (1, 1), (1, decode1 = ((1, 1), (1, 1), (1, 1), (1, 1)).(1, 1) 1), (1, 1)).(1, 1) decode0 = ((1 + 1), (1 1),(1 decode1 = ((1 1), (1 + 1),(1 + 1), (1 + 1)) 1), (1 1)) data0 = (2, 2, 2, 2), meaning data1 = (0, 0, 0, 0), meaning (1, 0, 1, 1) no data

When the receiver attempts to decode the signal using sender1's code, the data is all zeros, therefore the cross correlation is equal to zero and it is clear that sender1 did not transmit any data.

[EDIT ] ASYNCHRONOUS CDMA

See also: Direct-sequence spread spectrum and near-far problem The previous example of orthogonal Walsh sequences describes how 2 users can be multiplexed together in a synchronous system, a technique that is commonly referred to as code division multiplexing (CDM). The set of 4 Walsh sequences shown in the figure will afford up to 4 users, and in general, an NxN Walsh matrix can be used to multiplex N users. Multiplexing requires all of the users to be coordinated so that each transmits their assigned sequence v (or the complement, v) so that they arrive at the receiver at exactly the same time. Thus, this technique finds use in base-to-mobile links, where all of the transmissions originate from the same transmitter and can be perfectly coordinated.

Since each user generates MAI, controlling the signal strength is an important issue with CDMA transmitters. A CDM (synchronous CDMA), TDMA, or FDMA receiver can in theory completely reject arbitrarily strong signals using different codes, time slots or frequency channels due to the orthogonality of these systems. This is not true for Asynchronous CDMA; rejection of unwanted signals is only partial. If any or all of the unwanted signals are much stronger than the desired signal, they will overwhelm it. This leads to a general requirement in any asynchronous CDMA system to approximately match the various signal power levels as seen at the receiver. In CDMA cellular, the base station uses a fast closed-loop power control scheme to tightly control each mobile's transmit power.

adjacent channels will interfere, but decrease the utilization of the spectrum. Flexible Allocation of Resources Asynchronous CDMA offers a key advantage in the flexible allocation of resources i.e. allocation of a PN codes to active users. In the case of CDM, TDMA, and FDMA the number of simultaneous orthogonal codes, time slots and frequency slots respectively is fixed hence the capacity in terms of number of simultaneous users is limited. There are a fixed number of orthogonal codes, timeslots or frequency bands that can be allocated for CDM, TDMA, and FDMA systems, which remain underutilized due to the bursty nature of telephony and packetized data transmissions. There is no strict limit to the number of users that can be supported in an asynchronous CDMA system, only a practical limit governed by the desired bit error probability, since the SIR (Signal to Interference Ratio) varies inversely with the number of users. In a bursty traffic environment like mobile telephony, the advantage afforded by asynchronous CDMA is that the performance (bit error rate) is allowed to fluctuate randomly, with an average value determined by the number of users times the percentage of utilization. Suppose there are 2N users that only talk half of the time, then 2N users can be accommodated with the same average bit error probability as N users that talk all of the time. The key difference here is that the bit error probability for N users talking all of the time is constant, whereas it is a random quantity (with the same mean) for 2N users talking half of the time. In other words, asynchronous CDMA is ideally suited to a mobile network where large numbers of transmitters each generate a relatively small amount of traffic at irregular intervals. CDM

[EDIT] ADVANTAGES OF ASYNCHRONOUS CDMA OVER OTHER TECHNIQUES

Efficient Practical utilization of Fixed Frequency Spectrum In theory, CDMA, TDMA and FDMA have exactly the same spectral efficiency but practically, each has its own challenges power control in the case of CDMA, timing in the case of TDMA, and frequency generation/filtering in the case of FDMA. TDMA systems must carefully synchronize the transmission times of all the users to ensure that they are received in the correct timeslot and do not cause interference. Since this cannot be perfectly controlled in a mobile environment, each timeslot must have a guard-time, which reduces the probability that users will interfere, but decreases the spectral efficiency. Similarly, FDMA systems must use a guard-band between adjacent channels, due to the unpredictable doppler shift of the signal spectrum because of user mobility. The guard-bands will reduce the probability that

(synchronous CDMA), TDMA, and FDMA systems cannot recover the underutilized resources inherent to bursty traffic due to the fixed number of orthogonal codes, time slots or frequency channels that can be assigned to individual transmitters. For instance, if there are N time slots in a TDMA system and 2N users that talk half of the time, then half of the time there will be more than N users needing to use more than N timeslots. Furthermore, it would require significant overhead to continually allocate and deallocate the orthogonal code, time-slot or frequency channel resources. By comparison, asynchronous CDMA transmitters simply send when they have something to say, and go off the air when they don't, keeping the same PN signature sequence as long as they are connected to the system.

either spread its energy over the entire bandwidth of the signal or jam only part of the entire signal.[4] CDMA can also effectively reject narrowband interference. Since narrowband interference affects only a small portion of the spread spectrum signal, it can easily be removed through notch filtering without much loss of information. Convolution encoding and interleaving can be used to assist in recovering this lost data. CDMA signals are also resistant to multipath fading. Since the spread spectrum signal occupies a large bandwidth only a small portion of this will undergo fading due to multipath at any given time. Like the narrowband interference this will result in only a small loss of data and can be overcome. Another reason CDMA is resistant to multipath interference is because the delayed versions of the transmitted pseudo-random codes will have poor correlation with the original pseudo-random code, and will thus appear as another user, which is ignored at the receiver. In other words, as long as the multipath channel induces at least one chip of delay, the multipath signals will arrive at the receiver such that they are shifted in time by at least one chip from the intended signal. The correlation properties of the pseudo-random codes are such that this slight delay causes the multipath to appear uncorrelated with the intended signal, and it is thus ignored. Some CDMA devices use a rake receiver, which exploits multipath delay components to improve the performance of the system. A rake receiver combines the information from several correlators, each one tuned to a different path delay, producing a stronger version of the signal than a simple receiver with a single correlator tuned to the path delay of the strongest signal.[5]

[EDIT] SPREAD-SPECTRUM CHARACTERISTICS OF CDMA

Most modulation schemes try to minimize the bandwidth of this signal since bandwidth is a limited resource. However, spread spectrum techniques use a transmission bandwidth that is several orders of magnitude greater than the minimum required signal bandwidth. One of the initial reasons for doing this was military applications including guidance and communication systems. These systems were designed using spread spectrum because of its security and resistance to jamming. Asynchronous CDMA has some level of privacy built in because the signal is spread using a pseudo-random code; this code makes the spread spectrum signals appear random or have noise-like properties. A receiver cannot demodulate this transmission without knowledge of the pseudo-random sequence used to encode the data. CDMA is also resistant to jamming. A jamming signal only has a finite amount of power available to jam the signal. The jammer can

Frequency reuse is the ability to reuse the same radio channel frequency at other cell sites within a cellular system. In the FDMA and TDMA systems frequency planning is an important consideration. The frequencies used in different cells must be planned carefully to ensure signals from different cells do not interfere with each other. In a CDMA system, the same frequency can be used in every cell, because channelization is done using the pseudo-random codes. Reusing the same frequency in every cell eliminates the need for frequency planning in a CDMA system; however, planning of the different pseudo-random sequences must be done to ensure that the received signal from one cell does not correlate with the signal from a nearby cell.[6] Since adjacent cells use the same frequencies, CDMA systems have the ability to perform soft handoffs. Soft handoffs allow the mobile telephone to communicate simultaneously with two or more cells. The best signal quality is selected until the handoff is complete. This is different from hard handoffs utilized in other cellular systems. In a hard handoff situation, as the mobile telephone approaches a handoff, signal strength may vary abruptly. In contrast, CDMA systems use the soft handoff, which is undetectable and provides a more reliable and higher quality signal.[6]

Telecommunications Standards Institute (ETSI) in response to the earlier CDPD and i-mode packet switched cellular technologies. It is now maintained by the 3rd Generation Partnership Project (3GPP).[1][2] It is a best-effort service, as opposed to circuit switching, where a certain quality of service (QoS) is guaranteed during the connection. In 2G systems, GPRS provides data rates of 56114 kbit/second.[3] 2G cellular technology combined with GPRS is sometimes described as 2.5G, that is, a technology between the second (2G) and third (3G) generations of mobile telephony.[4] It provides moderate-speed data transfer, by using unused time division multiple access (TDMA) channels in, for example, the GSM system. GPRS is integrated into GSM Release 97 and newer releases. GPRS usage charging is based on volume of data, either as part of a bundle or on a pay as you use basis. An example of a bundle is up to 5 GB per month for a fixed fee. Usage above the bundle cap is either charged for per megabyte or disallowed. The pay as you use charging is typically per megabyte of traffic. This contrasts with circuit switching data, which is typically billed per minute of connection time, regardless of whether or not the user transfers data during that period.

GENERAL PACKET RADIO SERVICE

From Wikipedia, the free encyclopedia General packet radio service (GPRS) is a packet oriented mobile data service on the 2G and 3G cellular communication systems global system for mobile communications (GSM). The service is available to users in over 200 countries worldwide. GPRS was originally standardized by European

CONTENTS
1 Technical overview o 1.1 Services offered o 1.2 Protocols supported o 1.3 Hardware o 1.4 Addressing 2 Coding schemes and speeds o 2.1 Multiple access schemes

2.2 Channel encoding 2.3 Multislot Class 2.3.1 Multislot Classes for GPRS/EGPRS 2.3.2 Attributes of a multislot class 3 Usability 4 See also 5 References
o o

GPRS supports the following protocols:[citation needed]

internet protocol (IP). In practice, built-in mobile browsers use IPv4 since IPv6 is not yet popular. point-to-point protocol (PPP). In this mode PPP is often not supported by the mobile phone operator but if the mobile is used as a modem to the connected computer, PPP is used to tunnel IP to the phone. This allows an IP address to be assigned dynamically to the mobile equipment.

6 External links

[EDIT ] TECHNICAL OVERVIEW

See also: GPRS Core Network

X.25 connections. This is typically used for applications like wireless payment terminals, although it has been removed from the standard. X.25 can still be supported over PPP, or even over IP, but doing this requires either a network based router to perform encapsulation or intelligence built in to the end-device/terminal; e.g., user equipment (UE).

[EDIT] SERVICES OFFERED

GPRS extends the GSM circuit switched data capabilities and makes the following services possible:

"Always on" internet access Multimedia messaging service (MMS) Push to talk over cellular (PoC/PTT) Instant messaging and presencewireless village Internet applications for smart devices through wireless application protocol (WAP) Point-to-point (P2P) service: inter-networking with the Internet (IP)

When TCP/IP is used, each phone can have one or more IP addresses allocated. GPRS will store and forward the IP packets to the phone even during handover. The TCP handles any packet loss (e.g. due to a radio noise induced pause).

[EDIT] HARDWARE
Devices supporting GPRS are divided into three classes: Class A Can be connected to GPRS service and GSM service (voice, SMS), using both at the same time. Such devices are known to be available today.

If SMS over GPRS is used, an SMS transmission speed of about 30 SMS messages per minute may be achieved. This is much faster than using the ordinary SMS over GSM, whose SMS transmission speed is about 6 to 10 SMS messages per minute.

[EDIT] PROTOCOLS SUPPORTED

Class B

Can be connected to GPRS service and GSM service (voice, SMS), but using only one or the other at a given time. During GSM service (voice call or SMS), GPRS service is suspended, and then resumed automatically after the GSM service (voice call or SMS) has concluded. Most GPRS mobile devices are Class B. Class C Are connected to either GPRS service or GSM service (voice, SMS). Must be switched manually between one or the other service. A true Class A device may be required to transmit on two different frequencies at the same time, and thus will need two radios. To get around this expensive requirement, a GPRS mobile may implement the dual transfer mode (DTM) feature. A DTMcapable mobile may use simultaneous voice and packet data, with the network coordinating to ensure that it is not required to transmit on two different frequencies at the same time. Such mobiles are considered pseudo-Class A, sometimes referred to as "simple class A". Some networks are expected to support DTM in 2007.

models have connector for external antenna. Modems can be added as cards (for laptops) or external USB devices which are similar in shape and size to a computer mouse, or nowadays more like a pendrive.

[EDIT] ADDRESSING
A GPRS connection is established by reference to its access point name (APN). The APN defines the services such as wireless application protocol (WAP) access, short message service (SMS), multimedia messaging service (MMS), and for Internet communication services such as email and World Wide Web access. In order to set up a GPRS connection for a wireless modem, a user must specify an APN, optionally a user name and password, and very rarely an IP address, all provided by the network operator.

[EDIT ] CODING SCHEMES AND SPEEDS

The upload and download speeds that can be achieved in GPRS depend on a number of factors such as:

the number of BTS TDMA time slots assigned by the operator the channel encoding used. the maximum capability of the mobile device expressed as a GPRS multislot class

Huawei E220 3G/GPRS Modem USB 3G/GPRS modems use a terminal-like interface over USB 1.1, 2.0 and later, data formats V.42bis, and RFC 1144 and some

[EDIT] MULTIPLE ACCESS SCHEMES

The multiple access methods used in GSM with GPRS are based on frequency division duplex (FDD) and TDMA. During a session, a user is assigned to one pair of up-link and down-link frequency channels. This is combined with time domain statistical multiplexing; i.e., packet mode communication, which makes it possible for several users to share the same frequency channel. The packets have constant length, corresponding to a GSM time slot. The down-link uses first-come first-served packet scheduling, while the up-link uses a scheme very similar to reservation ALOHA (R-ALOHA). This means that slotted ALOHA (SALOHA) is used for reservation inquiries during a contention phase, and then the actual data is transferred using dynamic TDMA with first-come first-served scheduling.

coding scheme (CS-1) is used when the mobile station (MS) is further away from a BTS. Using the CS-4 it is possible to achieve a user speed of 20.0 kbit/s per time slot. However, using this scheme the cell coverage is 25% of normal. CS-1 can achieve a user speed of only 8.0 kbit/s per time slot, but has 98% of normal coverage. Newer network equipment can adapt the transfer speed automatically depending on the mobile location. In addition to GPRS, there are two other GSM technologies which deliver data services: circuit-switched data (CSD) and high-speed circuit-switched data (HSCSD). In contrast to the shared nature of GPRS, these instead establish a dedicated circuit (usually billed per minute). Some applications such as video calling may prefer HSCSD, especially when there is a continuous flow of data between the endpoints. The following table summarises some possible configurations of GPRS and circuit switched data services. Technolog y CSD HSCSD HSCSD GPRS GPRS EGPRS Download (kbit/s) 9.6 28.8 43.2 80.0 60.0 236.8 Upload (kbit/s) 9.6 14.4 14.4 20.0 (Class 8 & 10 and CS-4) 40.0 (Class 10 and CS-4) 59.2 (Class 8, TDMA Timeslots allocated 1+1 2+1 3+1 4+1 3+2 4+1

[EDIT] CHANNEL ENCODING

Channel encoding is based on a convolutional code at different code rates and GMSK modulation defined for GSM. The following table summarises the options: Codin Spee g d schem (kbit/s e ) CS-1 CS-2 CS-3 CS-4 8.0 12.0 14.4 20.0

The least robust, but fastest, coding scheme (CS-4) is available near a base transceiver station (BTS), while the most robust

(EDGE) EGPRS (EDGE) 177.6

10 and MCS-9) 118.4 (Class 10 and MCS-9) 3+2

i.e. when the best EDGE modulation and coding scheme can be used, 5 timeslots can carry a bandwidth of 5*59.2 kbit/s = 296 kbit/s. In uplink direction, 3 timeslots can carry a bandwidth of 3*59.2 kbit/s = 177.6 kbit/s.[5]

[EDIT] MULTISLOT CLASS

The multislot class determines the speed of data transfer available in the Uplink and Downlink directions. It is a value between 1 to 45 which the network uses to allocate radio channels in the uplink and downlink direction. Multislot class with values greater than 31 are referred to as high multislot classes. A multislot allocation is represented as, for example, 5+2. The first number is the number of downlink timeslots and the second is the number of uplink timeslots allocated for use by the mobile station. A commonly used value is class 10 for many GPRS/EGPRS mobiles which uses a maximum of 4 timeslots in downlink direction and 2 timeslots in uplink direction. However simultaneously a maximum number of 5 simultaneous timeslots can be used in both uplink and downlink. The network will automatically configure the for either 3+2 or 4+1 operation depending on the nature of data transfer. Some high end mobiles, usually also supporting UMTS also support GPRS/EDGE multislot class 32. According to 3GPP TS 45.002 (Release 6), Table B.2, mobile stations of this class support 5 timeslots in downlink and 3 timeslots in uplink with a maximum number of 6 simultaneously used timeslots. If data traffic is concentrated in downlink direction the network will configure the connection for 5+1 operation. When more data is transferred in the uplink the network can at any time change the constellation to 4+2 or 3+3. Under the best reception conditions,

[EDIT ] MULTISLOT CLASSES FOR GPRS/EGPRS Multisl ot Class 1 2 3 4 5 6 7 8 9 10 11 12 30 31 32 33 34 Downli Uplin Activ nk TS k TS e TS 1 2 2 3 2 3 3 4 3 4 4 4 5 5 5 5 5 1 1 2 1 2 2 3 1 2 2 3 4 1 2 3 4 5 2 3 3 4 4 4 4 5 5 5 5 5 6 6 6 6 6

[EDIT ] ATTRIBUTES OF A MULTISLOT CLASS Each multislot class identifies the following:

operators. With these enhancements the active round-trip time can be reduced, resulting in significant increase in applicationlevel throughput speeds.

the maximum number of Timeslots that can be allocated on uplink the maximum number of Timeslots that can be allocated on downlink the total number of timeslots which can be allocated by the network to the mobile the time needed for the mobile phone to perform adjacent cell signal level measurement and get ready to transmit the time needed for the MS to get ready to transmit the time needed for the MS to perform adjacent cell signal level measurement and get ready to receive the time needed for the MS to get ready to receive.

FM BROADCASTING IN INDIA
From Wikipedia, the free encyclopedia In the mid-nineties, when India first experimented with private FM broadcasts, the small tourist destination of Goa was the fifth place in this country of one billion where private players got FM slots. The other four centres were the big metro cities: Delhi, Mumbai, Kolkata and Chennai. These were followed by stations in Bangalore, Hyderabad, Jaipur and Lucknow. Indian policy currently states that these broadcasters are assessed a One-Time Entry Fee (OTEF), for the entire license period of 10 years. Under the Indian accounting system, this amount is amortised over the 10 year period at 10% per annum. Annual license fee for private players is either 4% of revenue share or 10% of Reserve Price, whichever is higher.

The different multislot class specification is detailed in the Annex B of the 3GPP Technical Specification 45.002 (Multiplexing and multiple access on the radio path)

[EDIT ] USABILITY
Earlier, India's attempts to privatise its FM channels ran into The maximum speed of a GPRS connection offered in 2003 was similar to a modem connection in an analog wire telephone network, about 32-40 kbit/s, depending on the phone used. Latency is very high; round-trip time (RTT) is typically about 600700 ms and often reaches 1 s. GPRS is typically prioritized lower than speech, and thus the quality of connection varies greatly. Devices with latency/RTT improvements (via, for example, the extended UL TBF mode feature) are generally available. Also, network upgrades of features are available with certain

rough weather when private players bid heavily and most could not meet their commitments to pay the government the amounts they owed.

CONTENTS
1 2 3 4 Content FM stations in New Delhi FM stations in MUMBAI FM stations in Bangalore

5 FM stations in chennai 6 Market view 7 List of FM radio Stations in India 8 Current allocation process

[EDIT ] FM STATIONS IN MUMBAI

Radio City 91.1 Big FM 92.7 Red FM 93.5 Radio One 94.3 Win FM 94.6 (The Station is closed) Radio Mirchi 98.3 AIR FM Gold 100.7 Fever 104 FM 104.0 Meow 104.8 AIR FM Rainbow 107.1 Mumbai One Gyan Vani Radio MUST Radio Jamia 90.4 FM

[EDIT ] CONTENT
News in not permitted on private FM, although the Federal Minister for Information-Broadcasting (I. and B. Ministry, Govt. of India) says this may be reconsidered in two to three years. Nationally, many of the current FM players, including the Times of India, Hindustan Times, Mid-Day, and BBC are essentially newspaper chains or media, and they are already making a strong pitch for news on FM.

[EDIT ] FM STATIONS IN NEW DELHI

AIR FM Rainbow / FM-1 (107.1 MHz) AIR FM Gold /FM-2 (Early Morning till Midnight) (106.4 MHz) AIR Rajdhani/Gyanvani Channel (Non-Regular broadcast) (105.6 MHz) Meow FM (104.8 MHz) Fever 104 (104 MHz) Radio Mirchi FM (98.3 MHz) Hit FM (95 MHz) Radio One FM (94.3 MHz) Red FM (93.5 MHz) Big FM (92.7 MHz) Radio City (91.1 MHz) Delhi University Educational Radio (Available only in University area) (DU Radio FM) (90.4 MHz)

[EDIT ] FM STATIONS IN BANGALORE

Main article: List of FM radio stations in Bangalore Radio City 91.1 FM - Kannada Radio Indigo 91.9 FM - English Big 92.7 FM - Kannada Red FM 93.5 FM - Kannada

[EDIT ] FM STATIONS IN CHENNAI

AIR FM - RAINBOW AIR FM - GOLD Hello FM (106.4), suryan FM, Aaha FM,

Big FM, radio city FM, radio mirchi FM, Radio-1 FM.

Antennas at a ham operator's station. Amateur radio or ham radio is a hobby that is practised by over 16,000 licenced users in India.[1] Licences are granted by the Wireless and Planning and Coordination Wing (WPC), a branch of the Ministry of Communications and Information Technology. In addition, the WPC allocates frequency spectrum in India. The Indian Wireless Telegraphs (Amateur Service) Rules, 1978 lists five licence categories:[2] To obtain a licence in the first four categories, candidates must pass the Amateur Station Operator's Certificate examination conducted by the WPC. This exam is held monthly in Delhi, Mumbai, Kolkata and Chennai, every two months in Ahmedabad, Nagpur and Hyderabad, and every four months in some smaller cities.[3] The examination consists of two 50-mark written sections: Radio theory and practice, Regulations; and a practical test consisting of a demonstration of Morse code proficiency in sending and receiving.[4] After passing the examination, the candidate must clear a police interview. After clearance, the WPC grants the licence along with the user-chosen call sign. This procedure can take up to one year.[5] This licence is valid for up to five years.[6] Each licence category has certain privileges allotted to it, including the allotment of frequencies, output power, and the emission modes. This article list the various frequencies allotted to various classes, and the corresponding emission modes and input DC power.

[EDIT ] MARKET VIEW

India's new private FM channels could also change the advertising scenario. Traditionally, radio accounts for 7% to 8% of advertiser expenditures around the world. In India, it is less than 2% at present.[citation needed]

[EDIT ] LIST OF FM RADIO STATIONS IN INDIA

See also: List of FM radio stations in India

[EDIT ] CURRENT ALLOCATION PROCESS

In FM Phase II the latest round of the long-delayed opening up of private FM in India some 338 frequencies were offered of which about 237 were sold.[citation needed] The government may go for rebidding of unsold frequencies quite soon. In Phase III of FM licensing, smaller towns and cities will be opened up for FM radio. Reliance and South Asia FM (Sun group) bid for most of the 91 cities, although they were allowed only 15% of the total allocated frequencies. Between them, they have had to surrender over 40 licenses.

LIST OF AMATEUR RADIO FREQUENCY BANDS IN INDIA

From Wikipedia, the free encyclopedia

CONTENTS

14.350

1 Allotted spectrum 2 Emission designations 3 Licence categories o 3.1 Short Wave Listener o 3.2 Grade II Restricted o 3.3 Grade II o 3.4 Grade I o 3.5 Advanced Grade 4 See also 5 Notes 6 References

7 7 7 7 8

18.068 18.168 21.000 21.450 24.890 24.990 28.000 29.700 144146 434438 1260 1300 3300 3400 5725 5840

17 m 15 m 12 m 10 m 2m 70 cm 23 cm 9 cm 5 cm

HF HF HF HF VHF UHF UHF SHF SHF

[EDIT ] ALLOTTED SPECTRUM

The following table lists the frequencies that amateur radio operators in India can operate on.

9 9 10 10

Band refers to the International Telecommunication Union (ITU) radio band designation Frequency is measured in megahertz Wavelength is measured in metres and centimetres Type refers to the radio frequency classification

[EDIT ] EMISSION DESIGNATIONS

Main article: Types of radio emissions The International Telecommunication Union uses an internationally agreed system for classifying radio frequency signals. Each Type of radio emission is classified according to its bandwidth, method of modulation, nature of the modulating signal, and Type of information transmitted on the carrier signal. It is based on characteristics of the signal, not on the transmitter used.

Ban Frequenc Waveleng Typ d y (MHz) th e 6 7 7 7 7 1.820 1.860 3.500 3.700 3.890 3.900 7.000 7.100 14.000 160 m 80 m 80 m 40 m 20 m MF HF HF HF HF

An emission designation is of the form BBBB 123 45, where BBBB is the bandwidth of the signal, 1 is a letter indicating the Type of modulation used, 2 is a digit representing the Type of modulating signal, 3 is a letter corresponding to the Type of information transmitted, 4 is a letter indicating the practical details of the transmitted information, and 5 is a letter that represents the method of multiplexing. The 4 and 5 fields are optional. For example, an emission designation would appear read as 500H A3E, where 500H translates to 500 Hz, and A3E is the emission mode as permitted. The WPC has authorized the following emission modes:[7] Emission

F2B

Electronic telegraphy, intended to be decoded by machine (radio teletype and digital modes) Frequency modulation, Single channel containing digital information, using a subcarrier, Electronic telegraphy, intended to be decoded by machine (radio teletype and digital modes) Frequency modulation, Single channel containing analogue information, Telephony (audio) Frequency modulation, Single channel containing analogue information, Facsimile (still images) Single-sideband with full carrier, Single channel containing analogue information, Telephony (audio) Single-sideband with suppressed carrier (e.g. Shortwave utility and amateur stations), Single channel containing analogue information, Telephony (audio) Single-sideband with reduced or variable carrier, Single channel containing analogue information, Telephony (audio)

F3E

F3C

Details Single channel containing digital information, no subcarrier, Aural telegraphy, intended to be decoded by ear, such as Morse code Single channel containing digital information, using a subcarrier, Aural telegraphy, intended to be decoded by ear, such as Morse code Double-sideband amplitude modulation (AM radio), Single channel containing analogue information, Single channel containing analogue information, None of the other listed types of emission Single channel containing analogue information, Video (television signals) Frequency modulation, Single channel containing digital information, no subcarrier, H3E

A1A

J3E

A2A

R3E

A3E

[EDIT ] LICENCE CATEGORIES [EDIT] SHORT WAVE LISTENER

The Short Wave Listener's Amateur Wireless Telegraph Station Licence allows listening on all amateur radio frequency bands, but prohibits transmission. The minimum age is 12.[8]

A3X

A3F[nb 1]

F1B

[EDIT] GRADE II RESTRICTED

The Restricted Amateur Wireless Telegraph Station Licence licence requires a minimum score of 40% in each section of the written examination, and 50% overall.[9] The minimum age is 12 years.[8] The licence allows a user to make terrestrial radiotelephony (voice) transmission in two VHF frequency bands. The maximum power allowed is 10 W. Ban Frequen Waveleng Typ d cy (MHz) th e 8 9 144146 2 m 434 438[nb 3] 70 cm
[2]

6 7 7 7 7

1.820 160 m 1.860[nb 4] 3.500 80 m 3.700[nb 4] 3.890 3.900 7.000 7.100 14.000 14.350 80 m 40 m 20 m

MF HF HF HF HF

A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E A1A, A2A, A3E, H3E, J3E, R3E

50 50 50 50 50

Emission

Powe r (W) 10[nb 2] 10[nb 2]

VHF A3E, H3E, J3E, R3E, F3E UHF A3E, H3E, J3E, R3E, F3E

18.068 18.168[nb 17 m
5]

HF

50

21.000 21.450 24.890 24.990 28.000 29.700

15 m 12 m 10 m

HF HF HF VHF UHF

50 50 50 10[nb 2] 10[nb 2]

[EDIT] GRADE II
The Amateur Wireless Telegraph Station Licence, GradeII licence requires the same scores as the Grade II Restricted, and in addition a demonstration of proficiency in sending and receiving Morse code at five words a minute.
[9]

7 7 8 9

The minimum age is 12

144146 2 m 434 438[nb 3] 70 cm

years.[8] The licence allows the user to make radiotelegraphy (Morse code) and radiotelephony transmission in 11 frequency bands. The maximum power allowed is 50 W. A Grade II licence holder can only be authorized the use of radio telephony emission on frequency bands below 30 MHz on submission of proof that 100 contacts have been made with other amateurs operators using CW (Morse code).[2] Ban Frequen Waveleng Typ d cy (MHz) th e Emission Powe r (W)

[EDIT] GRADE I
The Amateur Wireless Telegraph Station Licence, GradeI requires a minimum of 50% in each section of the written examination, and 55% overall, and a demonstration of proficiency in sending and receiving Morse code at 12 words a minute.[9] The minimum age is 14 years.[8] The licence allows a

user to make radiotelegraphy and radiotelephony transmission in 14 frequency bands. The maximum power allowed is 150 W. In addition, satellite communication, facsimile, and television modes are permitted.[2] Ban Frequen Waveleng Typ d cy (MHz) th e 6 1.820 160 m 1.860[nb 4] 3.500 80 m 3.700[nb 4] 3.890 3.900 7.000 7.100 14.000 14.350 MF Emission A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2A, F3E, F3C, A3C, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, Powe r (W) 150 9 434 438[nb 3] 70 cm 8 144146 2 m

A3X, A3F A1A, A2A, A3E, H3E, R3E, VHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, UHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, UHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, SHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, SHF J3E, F1B, F2B, F3E, F3C, A3X, A3F 25[nb 2]

25[nb 2]

1260 1300[nb 3] 23 cm
[nb 6]

25[nb 2]

HF

150

10

3300 9 cm 3400[nb 3] 5725 5 cm 5840[nb 3]

25[nb 2]

80 m

HF

150

10

25[nb 2]

40 m

HF

150

[EDIT] ADVANCED GRADE

The Advanced Amateur Wireless Telegraph Station Licence is the highest licence category. To obtain the licence, an applicant must be 18 years of age.[8] pass an advanced electronics examination, along with the Rules and Regulations section and Morse code sending and receiving at 12 words per minute.[9] The maximum power permitted is 400 W in selected sub-bands.[2]

20 m

HF

150

18.068 18.168[nb 17 m
5]

HF

150

21.000 21.450 24.890 24.990 28.000 29.700

15 m

HF

150

Ban Frequen Waveleng Typ d cy (MHz) th e 6 7 1.820 160 m 1.860[nb 4] 3.500 80 m 3.700[nb 4] MF HF

Emission A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C,

Powe r (W) 150 150

7 7

12 m 10 m

HF HF

150 150

A3X, A3F 7 3.890 3.900 7.000 7.100 14.000 14.350 80 m HF A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F 10 150

5725 5 cm 5840[nb 3]

A1A, A2A, A3E, H3E, R3E, SHF J3E, F1B, F2B, F3E, F3C, A3X, A3F

25[nb 2]

400 W sub-bands 150 Ban Frequen Waveleng Typ d cy (MHz) th e 7 3.520 80 m 3.540[nb 4] 3.890 3.900 7.050 7.100 14.050 14.150 14.220 14.320 21.100 21.400 HF Emission A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, J3E, F1B, F2B, F3E, F3C, A3X, A3F Powe r (W) 400

40 m

HF

20 m

HF

150

18.068 18.168[nb 17 m
5]

HF

150

80 m

HF

400

21.000 21.450 24.890 24.990 28.000 29.700

15 m

HF

150

40 m

HF

400

12 m

HF

150

20 m

HF

400

10 m

HF

150

20 m

HF

400

144146 2 m

A1A, A2A, A3E, H3E, R3E, VHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, UHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, UHF J3E, F1B, F2B, F3E, F3C, A3X, A3F A1A, A2A, A3E, H3E, R3E, SHF J3E, F1B, F2B, F3E, F3C, A3X, A3F

50

15 m

HF

400

434 438[nb 3]

70 cm

25[nb 2]

1260 1300[nb 3] 23 cm
[nb 6]

25[nb 2]

10

3300 9 cm 3400[nb 3]

25[nb 2]