Multiplexing is the process where multiple channels are combined for transmission over a common transmission path.
There are two types of multiplexing techniques:-
FDM (Frequency Division Mux)
TDM (Time Division MUX)
WDM: Wavelength Division Multiplex and Multiplexer
In fiber-optic communications, wavelength-division multiplexing (WDM) is a technology which multiplexes multiple optical carrier signals on a single optical fiber by using different wavelengths (colors) of laser light to carry different signals. This allows for a multiplication in capacity, in addition to enabling bidirectional communications over one strand of fiber. This is a form of frequency division multiplexing (FDM) but is commonly called wavelength division multiplexing.
"Wavelength Division Multiplexing (WDM) is a technique that multiple signals are carried together as separate wavelengths (color) of light in a multiplexed signal".
WDM is used in optical fiber networks. WDM and FDM (Frequency Division multiplex) are both based on the same principles but WDM applies to wavelengths of light in optical fiber while FDM is used in electrical analog transmission. A WDM optical system using a diffraction grating is completely passive, unlike electrical FDM, and thus is highly reliable. Further, a carrier wave of each WDM optical channel is higher than that of an FDM channel by a million times in frequency (THz versus MHz).
Wavelength Division Multiplexer is a device that combines optical signals from multiple different single-wavelength end devices onto a single fiber. Wavelength Division Multiplexer carries two to four wavelengths per fiber. The original WDM systems were dual-channel 1310/1550 nm systems. Typically, the same device can also perform the reverse process with the same WDM techniques: de-compose the data stream with multiple wavelength into multiple single wavelength data streams, a process call De-multiplexing. Therefore, it is very often a Wavelength Division Multiplexer and De-multiplexer are in the same box.
Click the picture to enlarge.
Time division multiplexing samples a number of different information channels sequentially for transmission on a single carrier. Frequency division multiplexing generally involves summing a multitude of independent sub-carrier frequencies, each with its retinue of modulation sidebands. In polarization division multiplexing, two different information signals are modulated onto carriers of the same frequency, which are propagated with their electrostatic fields oriented orthogonally. Wavelength division multiplexing incorporates independently modulated carriers of different frequencies sharing a common transmission medium. It is this latter technique to which the present study directs itself.
Figure 1 depicts a traditional fiber optic wavelength division multiplexing scheme. Three infrared sources (lasers or light emitting diodes-LEDs) of different wavelengths are modulated independently with their respective channel intelligence. Although any number of optical carrier frequencies might be used, common practice is to generate carriers at wavelengths typically near 850, 1300, and 1550 nm.
As you can see in Figure 2,3 these wavelengths represent low loss windows for typical single-mode glass fiber. Optical star couplers combine the three modulated carriers onto a single glass or plastic light pipe. At the receive end of the fiber, couplers split the output beam into three identical components, each containing elements of all three signals. Bandpass filters at the coupler outputs let each of three photo detectors "see" and respond to a single carrier frequency. De-multiplexing could be achieved as readily using wavelength selective couplers.
Encoding is the process of transforming information from one format into another. The opposite operation is called decoding. This is often used in many digital devices
There are a number of more specific meanings that apply in certain contexts:
Encoding (in cognition) is a basic perceptual process of interpreting incoming stimuli; technically speaking, it is a complex, multi-stage process of converting relatively objective sensory input (e.g., light, sound) into subjectively meaningful experience.
Data Encoding
This outlines basics of encoding analog or digital data using analog or digital signals.
Encoding signals
Basics - Fourrier analysis of periodic signal into harmonic components
Three parameters to sinusoid function:
amplitude
frequency
phase
frequency and phase are "angle" parameters
varying these parameters over time allows encoding of a signal
Encoding signals -
analog/analog encoding (modulation)
rationales :
1. necessity (may require high frequencies to transmit over medium)
2. ability to use FDM
AM - Amplitude modulation
value of amplitude encodes modulating signal
increased amplitude of modulating signal increases power of output power, but not bandwidth
DSBTC - double sideband transmitted carrier
Waveform:
s(t) = [1+n_a x(t)]cos(2 pi f_c t) where
s(t) = output signal
x(t) = input signal (normalized)
f_c = carrier frequency
cos(2 pi f_c t) = carrier signal (normalized)
n_a = modulation index, ratio of amplitude of input signal to carrier (so that the modulating signal, n_a x(t) has amplitude <>
1 = dc component to avoid loss of information (otherwise the peak negative amplitudes would cause the output signal to cross the axis)
multiply carrier signal by modulating signal plus a dc component to obtain output
output has redundancy in that the output is the sum of the carrier signal plus symmetric components spaced at f_m from the carrier frequency (where f_m is the modulating signal's frequency)
spectrum of AM signal is f_c +- spectrum of modulating signal: the upper sideband (above f_c) is the mirror image of the lower sideband (spectrum below f_c). Both are 1/2 power replicas of the original spectrum, with the lower sideband frequency-reversed.
Power transmitted = P_t = P_c(1+n_a^2/2), where P_c is the carrier power. Hence, n_a should be as large as possible, so that most of the signal power carries information, but remaining <>
Bandwidth needed B_t = 2B_m, where B_m is original bandwidth
SSB - Single Sideband
Since the spectrum of the AM signal has redundant information in the two sideband, and no information in the carrier signal, less power and less bandwidth can be used to transmit the same information by sending only one sideband. Half the bandwidth of DSBTC is used, and less power. Synchronization in the carrier signal is lost.
DSBSC - Double Sideband Supressed Carrier
Less power is used since carrier is not sent, but the same bandwidth is used and carrier sync lost.
VSB - Vestigial Sideband
One sideband is transmitted, and a reduced-power carrier.
FM - Frequency modulation (a form of angle modulation)
time derivative of phase angle encodes modulating signal
Waveform:
s(t) = A_c cos[2 pi f_c + phi(t)],
where phi'(t) = n_f m(t) is the derivative of the phase, and
A_c, f_c = carrier amplitude, frequency
m(t) = input modulating signal
Increased amplitude of input signal does not increase output power, but does increase bandwidth required.
Bandwidth required is, in theory, infinite, since the spectrum will contain components at f_c + K f_m for K=0,1,...
In practice, Carson's rule for FM says that B_t = 2[(n_f A_m)/(1 pi B_m) + 1]B_m
PM - Phase modulation (a form of angle modulation)
value of phase angle encodes modulating signal
Waveform:
s(t) = A_c cos[2 pi f_c + n_p m(t)],
where n_p = the phase modulation index (to normalize
n_p m(t) to the range 0..2 pi)
A_c, f_c = carrier amplitude, frequency
m(t) = input modulating signal
Bandwidth required is, in theory, infinite, since the spectrum will contain components at f_c + K f_m for K=0,1,...
In practice, Carson's rule for PM says that B_t = 2(n_p A_m + 1)B_m
PAM - pulse amplitude modulation
Thm: analog samples taken at more than twice the highest significant signal frequency will contain all the information of the original signal. The original signal may be reconstructed from the samples by use of a low-pass filter.
Bell's Dimension PBX products use this - analog samples are taken at twice the frequency of the signal, and only very short pulses reflecting these samples is transmitted.
PCM - pulses of PAM are quantized into discrete levels, then these levels are encoded in log (# levels) bits and sent. Two choices : sampling rate and number of levels
Noise -
quantizing noise - the original signal cannot be recovered since the pulses have been quantized.
SNR = 6n + 1.8 dB, where n = #bits/sample
Non-linear encoding - reduces quantizing distortion of weak signals by having finer gradations of discrete levels at the lower signal power levels, and coarser at higher power.
Companding function - same effect as n-l encoding may be had by compressing the analog signal before digitizing and expanding it after decoding. The companding function will giver greater gain to the weaker signals and less gain to strong signals.
Differential Encoding - only encode difference in signal rather than its absolute value
Delta Modulation (DM) - binary staircase function used to approximate the signal - on each time interval, either the staircase function goes up a quantum or down a quantum, (encoded as a 1 or a 0).
note: there is no limit on the amplitude of the signal encoded
note: similar to chain coding for contours in discrete 2-D.
Two choices : sampling rate and quantum size
Noise - two types: quantizing and slope overload
Large delta increases quantizing noise
Small delta increases slope overload noise
Predictive encoding - like DM only use an extrapolated value for point from which the difference is calculated. This is a generalized version of DM, with DM the special case where the predicted value is the same as the previous value, ie, a zero-order (constant) extrapolation function is used. DM has problems with rapid changes in the first derivative; in general, PDM using an nth order predictive encoding will have problems with rapid changes in the (n+1)th derivative.
digital/analog encoding (shift keying)
ASK - V amplitude levels can encode lg V bits per signal unit
FSK - V frequency levels can encode lg V bits per signal unit
PSK/QPSK - V phases can encode lg V bits per signal unit. QPSK uses four phases offset 90 degrees.
QAM - Mixed PSK and ASK, use pairs (p,a) to describe signal units, where p is the phase and a is the amplitude. This can keep power requirements lower and discrimination better than either technique alone.
PPM - pulse position modulation May be used with either analog or digital transmission to encode digital signals Signal is divided into frames, each frame has N slots (plus some synchronization overhead every so often) In each frame, exactly one slot has a pulse in it, A<>0; the other slots have A = 0 (no pulse). This allows lg N bits/frame to be encoded. It is useful when power requirements must be kept low and the transmission medium may be pulsed easily (e.g., lasers in deep-space communication)
digital/digital encoding
Needs - synch, no dc component
Evaluation
spectrum - max frequency (increased bandwidth requirement)
dc component (drift, direct coupling required)
synchronization
signal-based error-detection
susceptibility to interference (expressed as bit error rate)
cost/complexity
Methods
Level (L) - use same form to represent same bit value (1's always look the same, 0's always look the same)
Differential - use a change in signal element form to indicate a 1 (Mark or M), or a 0 (Space or S)
Examples -
NRZ-L: 1 = high, 0 = low
NRZ-M: 1 = transition at start of bit, 0 = no transition
NRZ-S: 1 = no transition, 0 = transition at start of bit
RZ: 1 = pulse to high, dropping back to low, 0 = low
bipolar: like NRZ except marks (1's) alternate polarity +1-1+1 (aka Bipolar AMI - alternate mark inversion) (sometime RZ technique used) (note: this will be shown below using _ for 0, + for a positive pulse, and - for a negative pulse)
pseudoternary - like bipolar only spaces (0's)
biphase-L (Manchester): always transition in middle of bit, 1 = high/low, 0 = low/high
differential Manchester: always transition in middle of bit, 1 = no transition, 0 = transition at start of bit
biphase-M: always transition at start of bit, 1 = transition in middle of bit, 0 = no transition
biphase-S: always transition at start of bit, 1 = no transition, 0 = transition in middle of bit
delay (Miller): 1 = transition in middle of bit, 0 = no transition if followed by a 1, transition at end of bit if followed by a 0
§To add synchronization capabilities to bipolar-AMI, which looses synchronization when a long string of 0's is sent, a run of 0's may be encoded by one of two fixed strings of 0's and 1's of the same length. These are distinguished from normal data by having one (or more) code violations (having two pulses of the same polarity in a row) in short succession, so that the receiver may detect these substituted patterns and decode them as a run of 0's.
§Examples -
§Bipolar with 8 Zero Substitution (B8ZS) - whenever 8 consecutive 0's are encountered in the data, then the 8 0's are replaced with the pattern 000VB0VB, where {V,B} = {+,-} with V the polarity of the last mark (causing a code violation) and B a pulse of the opposite polarity of the preceeding one. Popular in the USA. Requires only memory of polarity of last pulse and number of consecutive 0's seen, buffering the last 5 bits.
§EX:
§data= 011001000000000000100
§
§^^^^^^^^- run of 8 0's
§
§encoded = 0+-00+000+-0-+0000-00
§
§^---^^-^- code violations
§
§||||||||- special pattern
§
§High-density Bipolar - 3 Zeros (HDB3) - Popular in Europe and Japan. Starts with Bipolar-AMI, and substitutes 4 consecutive 0's with one of four patterns, according to both the polarity of the preceding pulse and the number of pulses since the last substitution (so that no dc component is introduced). Requires memory of polarity of last pulse, parity of number of pulses seen since last substitution, and number of consecutive 0's seen. Must buffer 4 bits.
§Pulses since last substitution
§
§Preceding PolarityOddEven
§
§-000-+00+
§
§+000+-00-
§
§IN GENERAL000VBOOV
§
§EX:
§data = 011001000000100000100
§
§^^^^^^^^- runs of 4 zeros each
§
§HDB3 = 0+-00+-00-00+000+0-00
§
§^--^^---^- code violations
§
§||||||||- special patterns
§
§Note: since repeating a pulse of the same polarity to cause a code violation will introduce dc bias, this must be remedied - B8ZS does this by balancing the number of positive and negative pulses in its substitution pattern, but the pattern is long, so 7 0's in the data can go through unsubstituted. HDB3 compensates for imbalance its shorter pattern could introduce by a more complex method for choosing the pattern used, so that the net imbalance is never bad. It does this by insuring that the unbalanced substitution patterns are always of alternating polarity.
Manchester Phase Encoding (MPE)
802.3 Ethernet uses Manchester Phase Encoding (MPE). A data bit '1' from the level-encoded signal (i.e. that from the digital circuitry in the host machine sending data) is represented by a full cycle of the inverted signal from the master clock which matches with the '0' to '1' rise of the phase-encoded signal (linked to the phase of the carrier signal which goes out on the wire). i.e. -V in the first half of the signal and +V in the second half.
The data bit '0' from the level-encoded signal is represented by a full normal cycle of the master clock which gives the '1' to '0' fall of the phase-encoded signal. i.e. +V in the first half of the signal and -V in the second half.
The above diagram shows graphically how MPE operates. The example at the bottom of the diagram indicates how the digital bit stream 10110 is encoded.
A transition in the middle of each bit makes it possible to synchronize the sender and receiver. At any instant the ether can be in one of three states: transmitting a 0 bit (-0.85v), transmitting a 1 bit (0.85v) or idle (0 volts). Having a normal clock signal as well as an inverted clock signal leads to regular transitions which means that synchronisation of clocks is easily achieved even if there are a series of '0's or '1's. This results in highly reliable data transmission. The master clock speed for Manchester encoding always matches the data speed and this determines the carrier signal frequency, so for 10Mbps Ethernet the carrier is 10MHz.
Differential Manchester Encoding (DME)
A '1' bit is indicated by making the first half of the signal, equal to the last half of the previous bit's signal i.e. no transition at the start of the bit-time. A '0' bit is indicated by making the first half of the signal opposite to the last half of the previous bit's signal i.e. a zero bit is indicated by a transition at the beginning of the bit-time. In the middle of the bit-time there is always a transition, whether from high to low, or low to high. Each bit transmitted means a voltage change always occurs in the middle of the bit-time to ensure clock synchronisation. Token Ring uses DME and this is why a preamble is not required in Token Ring, compared to Ethernet which uses Manchester encoding.
Non Return to Zero (NRZ)
NRZ encoding uses 0 volts for a data bit of '0' and a +V volts for a data bit of '1'. The problem with this is that it is difficult to distinguish a series of '1's or '0's due to clock synchronisation issues. Also, the average DC voltage is 1/2V so there is high power output. In addition, the bandwidth is large i.e. from 0Hz to half the data rate because for every full signal wave, two bits of data can be transmitted (remember that with MPE the data rate equals the bit rate which is even more inefficient!) i.e. two bits of information are transmitted for every cycle (or hertz).
After 50m of cable attenuation the signal amplitude may have been reduced to 100mV giving an induced noise tolerance of 100mV.
Return to Zero (RZ)
With RZ a '0' bit is represented by 0 volts whereas a '1' data bit is represented by +V volts for half the cycle and 0 volts for the second half of the cycle. This means that the average DC voltage is reduced to 1/4V plus there is the added benefit of there always being a voltage change even if there are a series of '1's. Unfortunately, the efficiency of bandwidth usage decreases if there are a series of '1's since now a '1' uses a whole cycle.
Non Return to Zero Invertive (NRZ-I)
With NRZ-I a '1' bit is represented by 0 volts or +V volts depending on the previous level. If the previous voltage was 0 volts then the '1' bit will be represented by +V volts, however if the previous voltage was +V volts then the '1' bit will be represented by 0 volts. A '0' bit is represented by whatever voltage level was used previously. This means that only a '1' bit can 'invert' the voltage, a '0' bit has no effect on the voltage, it remains the same as the previous bit whatever that voltage was.
This can be demonstrated in the following examples for the binary patterns 10110 and 11111:
Note how that a '1' inverts the voltage whilst a '0' leaves it where it is. This means that the encoding is different for the same binary pattern depending on the voltage starting point.
The bandwidth usage is minimised with NRZ-I, plus there are frequent voltage changes required for clock synchronisation.
With fibre there are no issues with power output so a higher clock frequency is fine whereas with copper NRZ-I would not be acceptable.
4B/5B
4B/5B encoding is sometimes called 'Block coding'. To get around this problem, an intermediate encoding takes place before the MLT-3 encoding. Each 4-bit 'nibble' of received data has an extra 5th bit added. If input data is dealt with in 4-bit nibbles there are 24 = 16 different bit patterns. With 5-bit 'packets' there are 25 = 32 different bit patterns. As a result, the 5-bit patterns can always have two '1's in them even if the data is all '0's a translation occurs to another of the bit patterns. This enables clock synchronisations required for reliable data transfer.
Notice that the clock frequency is 125MHz. The reason for this is due to the 4B/5B encoding. A 100MHz signal would not have been enough to give us 100Mbps, we need a 125MHz clock.
5B/6B
Same idea as 4B/5B but you can have DC balance (3 zero bits and 3 one bits in each group of 6) to prevent polarisation. 5B/6B Encoding is the process of encoding the scrambled 5-bit data patterns into predetermined 6-bit symbols. This creates a balanced data pattern, containing equal numbers of 0's and 1's, to provide guaranteed clock transitions synchronization for receiver circuitry, as well as an even power value on the line.
5B6B encoding also provides an added error-checking capability. Invalid symbols and invalid data patterns, such as more than three 0's or three 1's in a row, are easily detected
For 100VG-AnyLAN for instance, the clock rate on each wire is 30MHz, therefore 30Mbits per second are transmitted on each pair giving a total data rate of 120Mbits/sec. Since each 6-bits of data on the line represents 5 bits of real data due to the 5B/6B encoding, the rate of real data being transmitted is 25Mbits/sec on each pair, giving a total rate of real data of 100Mbits/sec. For 2-pair STP and fiber, the data rate is 120Mbits/sec on the transmitting pair, for a real data transmission rate of 100Mbits/sec.
8B/6T
8B/6T means send 8 data bits as six ternary (one of three voltage levels) signals. 3/4 (6/8) wave transitions transitions per bit i.e. the carrier just needs to be running at 3/4 of the speed of the data rate.
The incoming data stream is split into 8-bit patterns. Each 8-bit data pattern with two voltage levels 0 volts and V volts is examined. This 8-bit pattern is then converted into a 6-bit pattern but using three voltage levels -V, 0 and V volts, so each 8-bit pattern has a unique 6T code. For example the bit pattern 0000 0000 (0x00) uses the code +-00+- and 0000 1110 (0x)E) uses the code -+0-0+. There are 36 = 729 possible patterns (symbols). The rules for the symbols are that there must be at least two voltage transitions (to maintain clock synchronisation) and the average DC voltage must be zero (this is called 'DC balance' that is the overall DC voltage is summed up to 0v, the +V and -V transitions are evenly balanced either side of 0V) which stops any polarisation on the cable.
The maximum frequency that the 6T codes could generate on one carrier is 37.5MHz. FCC rules do not allow anything above 30MHz on cables and Category 3 cable does not allow anything above 16MHz (which is what 100BaseT4 was designed for). The 100BaseT4 standard uses 8B/6T encoding on three pairs in a round robin fashion such that the maximum carrier frequency on any single pair is 37.5/3 = 12.5MHz.
8B/10B
Each octet of data is examined and assigned a 10 bit code group. The data octet is split up into the 3 most significant bits and the 5 least significant bits. This is then represented as two decimal numbers with the least significant bits first e.g. for the octet 101 00110 we get the decimal 6.5. 10 bits are used to create this code group and the naming convention follows the format /D6.5/. There are also 12 special code groups which follow the naming convention /Kx.y/.
The 10 bit code groups must either contain five ones and five zeros, or four ones and six zeros, or six ones and four zeros. This ensures that not too many consecutive ones and zeros occurs between code groups thereby maintaining clock synchronisation. Two 'commas' are used to aid in bit synchronisation, these 'commas' are the 7 bit patterns 0011111 (+comma)and 1100000 (-comma).
In order to maintain a DC balance, a calculation called the Running Disparity calculation is used to try to keep the number of '0's transmitted the same as the number of '1's transmitted.
This uses 10 bits for each 8 bits of data and therefore drops the data rate speed relative to the line speed, for instance in order to gain a data rate of 1Gbps the line peed has to be 10/8 x 1 = 1.25Gbps .
MLT-3
This scheme was specified by ANSI X3T9.5 committee. It is used by FDDI and TP-PMD to obtain 100MB/s out of a 31.25MHz signal.
UTP is low pass in nature, meaning that it hinders high frequency signal (like a low-pass filter). So it is not feasible to merely increase the clock frequency by 10 to 100MHz and use Manchester encoding to give us 100Mbps. In addition, the FCC (Federal Communications Commission) have severely curtailed the power that is allowed to be emitted above 30MHz. We have to use another encoding technique in order to transmit high data rates across UTP.
If you take an averaging spectrum analyser and look at the output signal of the 10Mbps Ethernet phase-encoded signal, you will see a power peak at 10MHz where there is a stream of '1's or '0's, you will see a smaller harmonic at 30MHz and if there is a stream of '1's and '0's, you will see a peak at 5MHz. Now 100BaseT uses a master clock running at 125MHz instead of 10MHz. The equivalent peaks would then be at 125MHz, 375MHz and 62.5MHz. Transmission electronics designed to work within the FCC rules will block the frequencies higher than 30MHz.
To get around this issue we need to concentrate the signal power below 30MHz if possible. To do this the encoding method Multi-Level Transition 3 (MLT-3) is used. This involves using the pattern 1, 0, -1, 0. If the next data signal is a '1' then the output 'transitions' to the next bit in the pattern e.g. if the last output bit was a '-1', and the input bit is a '1', then the next output bit is a '0'. If the next data signal is a '0' then there is no transition which means that the next output bit is the same as last time, in our case a '0'.
The cycle length of the output signal is therefore going to be 1/4 that of the MPE method so that instead of the main signal peak being at 125MHz as measured by the averaging spectrum analyser, it will be at 31.25MHz which is near enough to be OK as far as FCC are concerned. 5 bits are transmitted for every 4 bits of data so that the data bit rate is actually 125Mb/s for 100Mb/s data throughput.
There is an issue with this in that you can end up with a series of '0's or '1's which force the local circuitry to count the bits using its own free running clock rather than have the check of the clock synchronisation from the transmit source.
PAM-5
This employs multi-level amplitude signalling. To encode 8 bits, 28 = 256 codes or symbols, are required since there are 256 possible pattern combinations. A five level signal (e.g. -2v, -1v, 0v, 1v and 2v) called Pulse Amplitude Modulation 5 is used (This works in a similar manner to MLT-3). Bearing in mind that there are 4 separate pairs being used for transmission and reception of data, this gives us a possibility of 54 = 625 codes to choose from when using all four pairs. Actually only four levels are used for data, the fifth level (0v) is used for the 4-dimensional 8-state Trellis Forward Error Correction used to recover the transmitted signal from the high noise.
If you plot time (nanoseconds) against voltage you will see an 'eye pattern' effect showing the different signal levels. Comparing a plot for MLT-3 against PAM-5 will demonstrate how that the separate levels for PAM-5 are less discreet. This is why extra convolution coding is used called Trellis coding, which uses Viterbi decoding for error detection and correction.
2 bits are represented per symbol and the symbol rate is 125Mbps in each direction on a pair because the clock rate is set at 125MHz. This gives 250Mbps data per pair and therefore 1000Mbps for the whole cable.
This type of encoding is used by Gigabit Ethernet. The data signals have distinct and measurable amplitude and phases relative to a 'marker signal'. Using this two way matrix allows more data bits per cycle, in the case of Gigabit Ethernet 1000Mbps is squeezed into 125MHz signals. The electronics are more complex and the technology is more susceptible to noise.
Thnx To Mr. R. E. Newman-Wolfe, University of Florida
The old telephone system (PSTN) uses circuit switching to transmit voice data whereas VoIP uses packet-switching to do so. The difference in the way these two types of switching work is the thing that made VoIP so different and successful.
To understand switching, you need to realize that the network in place between two communicating persons is a complex field of devices and machines, especially if the network is the Internet. Consider a person in Mauritius having a phone conversation with another person on the other side of the globe, say in the US. There are a large number of routers, switches and other kinds of devices that take the data transmitted during the communication from one end to the other.
Switching and routing Switching and routing are technically two different things, but for the sake of simplicity, let us take switches and routers (which are devices that make switching and routing respectively) as devices doing one job: make a link in the connection and forward data from the source to the destination.
Paths or circuits The important thing to look for in transmitting information over such a complex network is the path or circuit. The devices making up the path are called nodes. For instance, switches, routers and some other network devices, are nodes.
In circuit-switching, this path is decided upon before the data transmission starts. The system decides on which route to follow, based on a resource-optimizing algorithm, and transmission goes according to the path. For the whole length of the communication session between the two communicating bodies, the route is dedicated and exclusive, and released only when the session terminates.
Packet Switching
Packet switching is a digital network communications method that groups all transmitted data, irrespective of content, type, or structure into suitably-sized blocks, called packets. The network over which packets are transmitted is a shared network which routes each packet independently from all others and allocates transmission resources as needed. The principal goals of packet switching are to optimize utilization of available link capacity, minimize response times and increase the robustness of communication. When traversing network adapters, switches and other network nodes, packets are buffered and queued, resulting in variable delay and throughput, depending on the traffic load in the network.
Network resources are managed by statistical multiplexing or dynamic bandwidth allocation in which a physical communication channel is effectively divided into an arbitrary number of logical variable-bit-rate channels or data streams. Each logical stream consists of a sequence of packets, which normally are forwarded by a network node asynchronously using first-in, first-out buffering. Alternatively, the packets may be forwarded according to some scheduling discipline for fair queuing or for differentiated or guaranteed quality of service, such as pipeline forwarding or time-driven priority (TDP). Any buffering introduces varying latency and throughput in transmission. In case of a shared physical medium, the packets may be delivered according to some packet-mode multiple access schemes.
Packet switching contrasts with another principal networking paradigm, circuit switching, a method which sets up a specific circuit with a limited number dedicated connection of constant bit rate and constant delay between nodes for exclusive use during the communication session.
Packet mode (or packet-oriented, packet-based) communication may be utilized with or without intermediate forwarding nodes (packet switches)..
Connectionless and connection-oriented packet switching The service actually provided to the user by networks using packet switching nodes can be either connectionless (based on datagram messages), or virtual circuit switching (also known as connection oriented). Some connectionless protocols are Ethernet, IP, and UDP; connection oriented packet-switching protocols include X.25, Frame relay, Asynchronous Transfer Mode (ATM), Multiprotocol Label Switching (MPLS), and TCP.
In connection oriented networks, each packet is labeled with a connection ID rather than an address. Address information is only transferred to each node during a connection set-up phase, when an entry is added to each switching table in the network nodes.
In connectionless networks, each packet is labeled with a destination address, and may also be labeled with the sequence number of the packet. This precludes the need for a dedicated path to help the packet find its way to its destination. Each packet is dispatched and may go via different routes. At the destination, the original message/data is reassembled in the correct order, based on the packet sequence number. Thus a virtual connection, also known as a virtual circuit or byte stream is provided to the end-user by a transport layer protocol, although intermediate network nodes only provides a connectionless network layer service.
Packet switching in networks Packet switching is used to optimize the use of the channel capacity available in digital telecommunication networks such as computer networks, to minimize the transmission latency (i.e. the time it takes for data to pass across the network), and to increase robustness of communication.
The most well-known use of packet switching is the Internet and local area networks. The Internet uses the Internet protocol suite over a variety of Link Layer protocols. For example, Ethernet and frame relay are very common. Newer mobile phone technologies (e.g., GPRS, I-mode) also use packet switching.
X.25 is a notable use of packet switching in that, despite being based on packet switching methods, it provided virtual circuits to the user. These virtual circuits carry variable-length packets. In 1978, X.25 was used to provide the first international and commercial packet switching network, the International Packet Switched Service (IPSS). Asynchronous Transfer Mode (ATM) also is a virtual circuit technology, which uses fixed-length cell relay connection oriented packet switching.
Datagram packet switching is also called connectionless networking because no connections are established. Technologies such as Multiprotocol Label Switching (MPLS) and the Resource Reservation Protocol (RSVP) create virtual circuits on top of datagram networks. Virtual circuits are especially useful in building robust failover mechanisms and allocating bandwidth for delay-sensitive applications.
MPLS and its predecessors, as well as ATM, have been called "fast packet" technologies. MPLS, indeed, has been called "ATM without cells". Modern routers, however, do not require these technologies to be able to forward variable-length packets at multigigabit speeds across the network.
Virtual circuit packet switching
Protocols which route data from a node or hop to another hop between two end hosts in a network are called network-layer or L3 protocols. In the Internet, the only currently available network-layer protocol is IP, which comes in two flavors: IPv4 makes use of 32-bit long addresses and IPv6 uses 128-bit long addresses. We will mostly focus on IPv4 as IPv4 is the currently deployed version of IP in the Internet. Because of the advantages of virtual circuit packet switching and the growing popularity of IP, many Internet service providers send IP packets over virtual circuits. Virtual circuit packet switching technologies that have been used in the Internet backbones are ATM and, more recently, MPLS.
IP-over-ATM
ATM is a VC-switching technology which was standardized starting in the late 1980s. ATM uses fixed-length payloads with a length of 48 bytes and a 5-byte header, yielding 53-byte long ATM cells. Among the 40 header bits of a cell, 28 are reserved to identify the virtual circuit to which the cell belongs. The corresponding fields are called VCI/VPI (Virtual Circuit Identifier/Virtual Path Identifier). The VCI/VPI fields are updated at each switch.
A first issue that arises when trying to send IP packets over ATM virtual circuits is the need for a definition of an encapsulation of IP packets in ATM cells, i.e., how to put IP data inside ATM cells. Encapsulation is performed by an adaptation layer (AAL). Moreover, most IP packets are too large to fit in a 53-byte ATM cell. An IP header is at least 20 bytes long, and hosts cannot be coerced to send IP packets of at most 53 bytes. Therefore, IP packets must be cut into smaller pieces in a process called segmentation before they can be encapsulated and put in ATM cells. The last router on the path of IP packets must reassemble the fragments to reconstitute the original IP packets. Segmentation and Reassembly (SAR) is a complex and time consuming process.
Sending IP traffic over an ATM infrastructure proves to be complex. Far from solving the circuit interruption problem that arises on link failure with VC-switching, IP over ATM introduces the aforementioned issues and requires additional hardware as combined IP routers/ATM switches do not exist.
Brief comparison
Circuit switching is old and expensive, and it is what PSTN uses. Packet switching is more modern.
When you are making a PSTN call, you are actually renting the lines, with all it implies. See why international calls are expensive? So if you speak for, say 10 minutes, you pay for ten minutes of dedicated line. You normally speak only when your correspondent is silent, and vice versa. Taking also into consideration the amount of time no one speaks, you finally use much less than half of what you are paying for. With VoIP, you actually can use a network or circuit even if there are other people using it at the same time. There is no circuit dedication. The cost is shared.
Circuit-switching is more reliable than packet-switching. When you have a circuit dedicated for a session, you are sure to get all information across. When you use a circuit which is open for other services, then there is a big possibility of congestion (which is for a network what a traffic jam is for the road), and hence the delays or even packet loss. This explains the relatively lower quality of VoIP voice compared to PSTN. But you actually have other protocols giving a helping hand in making packet-switching techniques to make connections more reliable. An example is the TCP protocol. Since voice is to some extent tolerant to some packet loss (unless text - since a comma lost can mean a big difference), packet-switching is finally ideal for VoIP.
