UNIT-3

Types of Signals
Conveying an information by some means such as gestures, sounds, actions, etc., can be
termed as signaling. Hence, a signal can be a source of energy which transmits some
information. This signal helps to establish a communication between the sender and the
receiver.
An electrical impulse or an electromagnetic wave which travels a distance to convey a
message, can be termed as a signal in communication systems.
Depending on their characteristics, signals are mainly classified into two types: Analog and
Digital. Analog and Digital signals are further classified, as shown in the following figure.

Periodic Signal
Any analog or digital signal, that repeats its pattern over a period of time, is called as
a Periodic Signal. This signal has its pattern continued repeatedly and is easy to be
assumed or to be calculated.
Aperiodic Signal
Any analog or digital signal, that doesn’t repeat its pattern over a period of time is called
as Aperiodic Signal. This signal has its pattern continued but the pattern is not repeated. It
is also not so easy to be assumed or to be calculated.

Analog communication
Analog Communication is a data transmitting technique in a format that utilizes continuous
signals to transmit data including voice, image, video, electrons etc. An analog signal is a
variable signal continuous in both time and amplitude which is generally carried by use of
modulation. Digital communications is the physical transfer over a point-to-point or point-
to-multi point transmission medium. Examples of such media are copper wires, optical
fibers, wireless communication media, and storage media.
Analog communication is a communication method of conveying voice, data, image, signal
or video information using a continuous signal which varies in amplitude, phase, or some other
property in proportion to that of a variable. It could be the transfer of an analog source signal using
an analog modulation method such as FM or AM, or no modulation at all.

Analog transmission is still very popular, in particular for shorter distances, due to
significantly lower costs and complex multiplexing and timing equipment is unnecessary,
and in small “short-haul” systems that simply do not need multiplexed digital transmission.
However, in situations where a signal often has high signal-to-noise ratio and cannot
achieve source linearity, or in long distance, high output systems, analog is unattractive due
to attenuation problems. Furthermore, as digital techniques continue to be refined, analog
systems are increasingly becoming legacy equipment.
Recently, some nations, such as the Netherlands, have completely ceased analog
transmissions on certain media, such as television, for the purposes of the government
saving money.
Analog systems are very tolerant to noise, make good use of bandwidth, and are easy to
manipulate mathematically. However, analog signals require hardware receivers and
transmitters that are designed to perfectly fit the particular transmission. If you are
working on a new system, and you decide to change your analog signal, you need to
completely change your transmitters and receivers.
Analog signals are signals with continuous values. Analog signals are used in many systems,
although the use of analog signals has declined with the advent of cheap digital signals.

DIGITAL COMMUNICATION
Digital communications is the physical transfer of data(a digital bit stream) over a point-to-
point or point-to-multipoint transmission medium. Examples of such media are copper
wires, optical fibers, wireless communication media, and storage media. The data is often
represented as an electro-magnetic signal, such as an electrical voltage signal or an infra-
red signal.
Data transmitted may be digital messages originating from a data source, for example a
computer or a keyboard. It may also be ananalog signal such as a phone call or a video
signal, digitized into a bit-stream for example using pulse-code modulation(PCM) or more
advanced source coding(data compression) schemes. This source coding and decoding is
carried out by codec equipment.
Digital communication enables the data to be transmitted in an efficient manner through
the use of digitally encoded information sent through data signals. These data signals are
easily compressed and, as such, can be transmitted with accuracy and speed.
Unlike in an analog communications where the continuity of a varying signal can not be
broken, in a digital communication a digital transmission can be broken down into packets
as discrete messages. Transmitting data in discrete messages not only facilitates the error
detection and correction but also enables a greater signal processing capability. Digital
communication has, in large part, replaced analog communication as the ideal form of
transmitting information through computer and mobile technologies.
The information source generates particular symbols at a particular rate. The source
encoder translates these symbols in sequences of 0’s and 1’s. The channel encoder is
oriented towards translating sequences of 0’s and 1’s to other sequences of 0’s and 1’s, to
realize high transmission reliability and efficiency. The modulator accepts streams of 0’s
and 1’s, and converts them to electrical waveforms suitable for transmission.
The communication channel provides the electrical connection between the source and
destination. It has a finite bandwidth, and the waveform transmitted suffers from
amplitude distortion and phase distortion. In addition to distortion, power is decreased
due to attenuation of the channel. Finally, the waveform is corrupted by unwanted
electrical signals, referred to as noise. The primary objective of a communication system is
to suppress the bad effects of noise as much as possible.
The inverse process takes place at the destination side. The demodulator converts the
electrical waveforms to sequences of 0’s and 1’s, the channel decoder translates the
sequence of 0’s and 1’s to the original sequence of 0’s and 1’s. It also performs error
correction and clock recovery. The source decoder finally translates the sequence of 0’s and
1’s into symbols.

COMPARISON OF ANALOG AND DIGITAL COMMUNICATION

ANALOG
Analog signals are signals with continuous values. Analog signals are used in many systems,
although the use of analog signals has declined with the advent of cheap digital signals.
Analog systems are very tolerant to noise, make good use of bandwidth, and are easy to
manipulate mathematically.
However, analog signals require hardware receivers and transmitters that are designed to
perfectly fit the particular transmission. If you are working on a new system, and you
decide to change your analog signal, you need to completely change your transmitters and
receivers.
DIGITAL
Digital signals are signals that are represented by binary numbers, “1” or “0”. The 1 and 0
values can correspond to different discrete voltage values, and any signal that doesnt quite
fit into the scheme just gets rounded off. Digital signals are intolerant to noise, and digital
signals can be completely corrupted in the presence of excess noise. In digital signals, noise
could cause a 1 to be interpreted as a 0 and vice versa, which makes the received data
different than the original data. Imagine if the army transmitted a position coordinate to a
missile digitally, and a single bit was received in error? This single bit error could cause a
missile to miss its target by miles. Luckily, there are systems in place to prevent this sort of
scenario, such as checksums and CRCs, which tell the receiver when a bit has been
corrupted and ask the transmitter to resend the data. The primary benefit of digital signals
is that they can be handled by simple, standardized receivers and transmitters, and the
signal can be then dealt with in software (which is comparatively cheap to change).
Analog communication systems, amplitude modulation (AM) radio being a typifying
example, can inexpensively communicate a bandlimited analog signal from one location to
another (point-to-point communication) or from one point to many (broadcast). Although
it is not shown here, the coherent receiver provides the largest possible signal-to-noise
ratio for the demodulated message. An analysis of this receiver thus indicates that some
residual error will always be present in an analog system’s output.
Although analog systems are less expensive in many cases than digital ones for the same
application, digital systems offer much more efficiency, better performance, and much
greater flexibility.
1. Efficiency: The Source Coding Theorem allows quantification of just how complex a given
message source is and allows us to exploit that complexity by source coding (compression).
In analog communication, the only parameters of interest are message bandwidth and
amplitude. We cannot exploit signal structure to achieve a more efficient communication
system.
2. Performance: Because of the Noisy Channel Coding Theorem, we have a specific criterion
by which to formulate error-correcting codes that can bring us as close to error-free
transmission as we might want. Even though we may send information by way of a noisy
channel, digital schemes are capable of error-free transmission while analog ones cannot
overcome channel disturbances; see this problem for a comparison.
3. Flexibility: Digital communication systems can transmit real-valued discrete-time
signals, which could be analog ones obtained by analog-to-digital conversion, and
symbolic-valued ones (computer data, for example). Any signal that can be transmitted by
analog means can be sent by digital means, with the only issue being the number of bits
used in A/D conversion (how accurately do we need to represent signal amplitude). Images
can be sent by analog means (commercial television), but better communication
performance occurs when we use digital systems (HDTV). In addition to digital
communication’s ability to transmit a wider variety of signals than analog systems, point-
to-point digital systems can be organized into global (and beyond as well) systems that
provide efficient and flexible information transmission. Computer networks, explored in
the next section, are what we call such systems today. Even analog-based networks, such as
the telephone system, employ modern computer networking ideas rather than the purely
analog systems of the past.
Consequently, with the increased speed of digital computers, the development of
increasingly efficient algorithms, and the ability to interconnect computers to form a
communications infrastructure, digital communication is now the best choice for many
situations.

Advantages and disadvantages of analog vs digital communication:

1. The first advantage of digital communication against analog is it’s noise immunity. In any
transmission path some unwanted voltage or noise is always present which cannot be
eliminated fully. when signal is transmitted this noise gets added to the original signal
causing the distortion of the signal. However in a digital communication at the receiving
end this additive noise can be eliminated to great extent easily resulting in better recovery
of actual signal. In the case of analog communication it’s difficult to remove the noise once
added to the signal.
2. Security is the another priority of messaging the services in the modern days. The Digital
communication provides better security to messages than the analog communication. It can
be achieved through various coding techniques available in digital communication.
3. In a digital communication the signal is digitized to a stream of 0 s and 1 s. So at the
receiver side a simple decision has to me made whether the received signal is a 0 or a
1.Accordingly the receiver circuit becomes simpler as compared to the analog receiver
circuit.
4. When signal is travelling through it’s transmission path gets faded gradually. So on it’s
path it needs to be reconstructed to it’s actual form and re-transmitted many times. For
that reason AMPLIFIERS are used for analog communication and REPEATERS are used in
digital communication. amplifiers are needed every 2 to 3 Kms apart where as repeaters
are need every 5 to 6 Kms apart. So definitely digital communication is cheaper. Amplifiers
also often add non-linearities that distort the actual signal.
5. Bandwidth is another scarce resource. Various Digital communication techniques are
available that use the available bandwidth much efficiently than analog communication
techniques.
6. When audio and video signals are transmitted digitally an ADC(Analog to Digital)
converter is needed at transmitting side and a DAC(Digital to Analog) converter is again
needed at receiver side. While transmitted in analog communication these devices are not
needed.
7. Digital signals are often an approximation of the analog data(like voice or video) that is
obtained through a process called quantization. The digital representation is never the
exact signal but it’s most closely approximated digital form. So it’s accuracy depends on the
degree of approximation taken in quantization process.
3.1 Representing data as Analog Signals
3.1.1 Analog data-to-Analog Signal Conversion
Analog signals are modified to represent analog data. This conversion is also known as
Analog Modulation. Analog modulation is required when bandpass is used. Analog to
analog conversion can be done in three ways:

 Amplitude Modulation
In this modulation, the amplitude of the carrier signal is modified to reflect the analog data.

Amplitude modulation is implemented by means of a multiplier. The amplitude of

modulating signal (analog data) is multiplied by the amplitude of carrier frequency, which
then reflects analog data.
The frequency and phase of carrier signal remain unchanged.
 Frequency Modulation
In this modulation technique, the frequency of the carrier signal is modified to reflect the
change in the voltage levels of the modulating signal (analog data).

The amplitude and phase of the carrier signal are not altered.

 Phase Modulation
In the modulation technique, the phase of carrier signal is modulated in order to reflect the
change in voltage (amplitude) of analog data signal.
Phase modulation is practically similar to Frequency Modulation, but in Phase modulation
frequency of the carrier signal is not increased. Frequency of carrier is signal is changed
(made dense and sparse) to reflect voltage change in the amplitude of modulating signal.

3.1.2 Converting Digital data to Analog Signals

To send the digital data over an analog media, it needs to be converted into analog signal.
There can be two cases according to data formatting.
Bandpass: The filters are used to filter and pass frequencies of interest. A bandpass is a
band of frequencies which can pass the filter.
Low-pass: Low-pass is a filter that passes low frequencies signals.
When digital data is converted into a bandpass analog signal, it is called digital-to-analog
conversion. When low-pass analog signal is converted into bandpass analog signal, it is
called analog-to-analog conversion.
When data from one computer is sent to another via some analog carrier, it is first
converted into analog signals. Analog signals are modified to reflect digital data.
An analog signal is characterized by its amplitude, frequency, and phase. There are three
kinds of digital-to-analog conversions:
 Amplitude Shift Keying
In this conversion technique, the amplitude of analog carrier signal is modified to reflect
binary data.

When binary data represents digit 1, the amplitude is held; otherwise it is set to 0. Both
frequency and phase remain same as in the original carrier signal.
 Frequency Shift Keying
In this conversion technique, the frequency of the analog carrier signal is modified to
reflect binary data.
This technique uses two frequencies, f1 and f2. One of them, for example f1, is chosen to
represent binary digit 1 and the other one is used to represent binary digit 0. Both
amplitude and phase of the carrier wave are kept intact.
 Phase Shift Keying
In this conversion scheme, the phase of the original carrier signal is altered to reflect the
binary data.
When a new binary symbol is encountered, the phase of the signal is altered. Amplitude
and frequency of the original carrier signal is kept intact.
 Quadrature Phase Shift Keying
QPSK alters the phase to reflect two binary digits at once. This is done in two different
phases. The main stream of binary data is divided equally into two sub-streams. The serial
data is converted in to parallel in both sub-streams and then each stream is converted to
digital signal using NRZ technique. Later, both the digital signals are merged together.

3.2 Representing Data as Digital Signals

3.2.1 Converting Analog data to Digital Signals
A signal is pulse code modulated to convert its analog information into a binary sequence,
i.e., 1s and 0s. The output of a PCM will resemble a binary sequence. The following figure
shows an example of PCM output with respect to instantaneous values of a given sine wave.

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process
is called as digital. Each one of these digits, though in binary code, represent the
approximate amplitude of the signal sample at that instant.
In Pulse Code Modulation, the message signal is represented by a sequence of coded pulses.
This message signal is achieved by representing the signal in discrete form in both time and
amplitude.
Basic Elements of PCM
The transmitter section of a Pulse Code Modulator circuit consists of Sampling,
Quantizing and Encoding, which are performed in the analog-to-digital converter section.
The low pass filter prior to sampling prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of impaired signals,
decoding, and reconstruction of the quantized pulse train. Following is the block diagram
of PCM which represents the basic elements of both the transmitter and the receiver
sections.

Low Pass Filter

This filter eliminates the high frequency components present in the input analog signal
which is greater than the highest frequency of the message signal, to avoid aliasing of the
message signal.
1. Sampling – The first step in PCM is sampling. Sampling is a process of measuring
the amplitude of a continuous-time signal at discrete instants, converting the
continuous signal into a discrete signal. There are three sampling methods:
(i) Ideal Sampling: In ideal Sampling also known as Instantaneous sampling pulses from
the analog signal are sampled. This is an ideal sampling method and cannot be easily
implemented.
(ii) Natural Sampling: Natural Sampling is a practical method of sampling in which pulse
have finite width equal to T. The result is a sequence of samples that retain the shape of the
analog signal.

(iii) Flat top sampling: In comparison to natural sampling flat top sampling can be easily
obtained. In this sampling technique, the top of the samples remains constant by using a
circuit. This is the most common sampling method used.

Nyquist Theorem:
One important consideration is the sampling rate or frequency. According to the Nyquist
theorem, the sampling rate must be at least 2 times the highest frequency contained in the
signal. It is also known as the minimum sampling rate and given by:
Fs =2*fh
 2. Quantization –
The result of sampling is a series of pulses with amplitude values between the
maximum and minimum amplitudes of the signal. The set of amplitudes can be
infinite with non-integral values between two limits.
The following are the steps in Quantization:
1. We assume that the signal has amplitudes between Vmax and Vmin
2. We divide it into L zones each of height d where,
d= (Vmax- Vmin)/ L

3. The value at the top of each sample in the graph shows the actual amplitude.
4. The normalized pulse amplitude modulation(PAM) value is calculated using
the formula amplitude/d.
5. After this we calculate the quantized value which the process selects from the
middle of each zone.
 6. The Quantized error is given by the difference between quantised value and
normalised PAM value.
The Quantization code for each sample based on quantization levels at the left of the graph.
Encoder
The digitization of analog signal is done by the encoder. It designates each quantized level
by a binary code. The sampling done here is the sample-and-hold process. These three
sections LPF, Sampler, and QuantizerLPF, Sampler, and Quantizer will act as an analog to
digital converter. Encoding minimizes the bandwidth used.
Regenerative Repeater
This section increases the signal strength. The output of the channel also has one
regenerative repeater circuit, to compensate the signal loss and reconstruct the signal, and
also to increase its strength.
Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original signal.
This circuit acts as the demodulator.
Reconstruction Filter
After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a
low-pass filter is employed, called as the reconstruction filter to get back the original signal.
Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and
samples it, and then transmits it in an analog form. This whole process is repeated in a
reverse pattern to obtain the original signal.

b. DELTA MODULATION :
Since PCM is a very complex technique, other techniques have been developed to reduce
the complexity of PCM. The simplest is delta Modulation. Delta Modulation finds the change
from the previous value.
Modulator – The modulator is used at the sender site to create a stream of bits from an
analog signal. The process records a small positive change called delta. If the delta is
positive, the process records a 1 else the process records a 0. The modulator builds a
second signal that resembles a staircase. The input signal is then compared with this
gradually made staircase signal.
We have the following rules for output:
1. If the input analog signal is higher than the last value of the staircase signal, increase
delta by 1, and the bit in the digital data is 1.
2. If the input analog signal is lower than the last value of the staircase signal, decrease
delta by 1, and the bit in the digital data is 0.

C. ADAPTIVE DELTA MODULATION:

The performance of a delta modulator can be improved significantly by making the step
size of the modulator assume a time-varying form. A larger step-size is needed where the
message has a steep slope of modulating signal and a smaller step-size is needed where the
message has a small slope. The size is adapted according to the level of the input signal.
This method is known as adaptive delta modulation (ADM).

3.2.2. Converting Digital Data to Digital Signals

Transmitting digital data (e.g. output from a computer ) across a digital network(e.g. an
Ethernet LAN) requires representing the digital data as a digital signal. Three common
coding techniques used for this task are
1. Manchester coding
2. Differential Manchester coding
3. Non return to zero, invert to ones (NRZI)

Manchester coding
 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.

Non return to zero, invert to ones (NRZI)

Non Return to Zero (NRZ) is a binary code used in telecommunications transmission, where a
data bit of 1 is positive voltage, and a data bit of 0 is negative voltage. NRZ code does not have a
neutral state, versus Return to Zero (RZ) code, which has a rest state.

In the absence of independent clock signals, certain mechanisms are required when NRZ data is
asynchronously coded. NRZI maps binary signals to physical signals during transmission. If a
data bit is 1, NRZI transitions at the clock boundary. If a data bit is 0, there is no transition.
NRZI may have long series of 0s or 1s, resulting in clock recovery difficulties.

An additional encoding mechanism must be used to ensure clock recovery. Run-length limited
(RLL) encoding, such as that used with magnetic disk storage devices, is preferred over
Universal Serial Bus (USB) bit stuffing, which often results in variable data transfer rates (DTR).

3.3 Data Rate and Baud Rate

Bit rate and Baud rate, these two terms are often used in data communication. Bit rate is
simply the number of bits (i.e., 0’s and 1’s) transmitted in per unit time. While Baud rate is
the number of signal units transmitted per unit time that is needed to represent those
bits.
The crucial difference between bit rate and baud rate that one change of state can transfer
one bit, or slightly more or less than one bit that relies on the modulation technique used.
Hence, the given equation defines the relation between the two:
Bit rate = baud rate x the number of bit per baud
If we talk about computer efficiency, the bit rate is the more important where we want to
know how long it takes to process each piece of information. But when we are more
concerned about how that data is moved from one place to another we emphasize on the
baud rate. The fewer signals required, the more efficient the system and the less bandwidth
needed to transmit more bits.
An analogy can illustrate the concept of bauds and bits. In transportation, a baud is
comparable to a bus, a bit analogous to a passenger. A bus can carry multiple passengers. If
1000 buses go from one point to another carrying only one passenger (the driver), then
1000 passengers are transported. However, if each bus carries twenty passengers
(suppose), then 20000 passengers are transported. In this case, busses determine traffic
not the number of passengers consequently broader highways are needed. Likewise, the
number of bauds determines the required bandwidth, not the number of bits.

BASIS FOR
BIT RATE BAUD RATE
COMPARISON

Basic Bit rate is the count of bits per Baud rate is the count of signal units
second. per second.

Meaning It determines the number of It determines how many times the

bits traveled per second. state of a signal is changing.

Term usually used While the emphasis is on While data transmission over the
computer efficiency. channel is more concerned.

Bandwidth Can not determine the It can determine how much bandwidth
determination bandwidth. is required to send the signal.

Equation Bit rate = baud rate x the count Baud rate = bit rate / the number of
BASIS FOR
BIT RATE BAUD RATE
COMPARISON

of bits per signal unit bits per signal unit

Definition of Bit Rate

Bit rate can be defined as the number of bit intervals per second. And bit interval is
referred to as the time needed to transfer one single bit. In simpler words, the bit rate is the
number of bits sent in one second, usually expressed in bits per second (bps). For example,
kilobits per second (Kbps), Megabits per second (Mbps), Gigabits per second (Gbps), etc.

Definition of Baud Rate

Baud rate is expressed in the number of times a signal can change on transmission line
per second. Usually, the transmission line uses only two signal states, and make the baud
rate equal to the number of bits per second that can be transferred.

3.4 Digital carrier systems

The T-carrier is a member of the series of carrier systems developed by AT&T Bell
Laboratories for digital transmission of multiplexed telephone calls.
The first version, the Transmission System 1 (T1), was introduced in 1962 in the Bell
System, and could transmit up to 24 telephone calls simultaneously over a single
transmission line of copper wire. Subsequent specifications carried multiples of the basic
T1 (1.544 Mbit/s) data rates, such as T2 (6.312 Mbit/s) with 96 channels, T3 (44.736
Mbit/s) with 672 channels, and others.
The T-carrier is a hardware specification for carrying multiple time-division
multiplexed (TDM) telecommunications channels over a single four-wire transmission
circuit. It was developed by AT&T at Bell Laboratories.
The T-carriers are commonly used for trunking between switching centers in a telephone
network, including to private branch exchange (PBX) interconnect points.
It uses the same twisted pair copper wire that analog trunks used, employing one pair for
transmitting, and another pair for receiving. Signal repeaters may be used for extended
distance requirements.
Before the digital T-carrier system, carrier wave systems such as 12-channel carrier
systems worked by frequency division multiplexing; each call was an analog signal. A T1
trunk could transmit 24 telephone calls at a time, because it used a digital carrier
signal called Digital Signal 1 (DS-1). DS-1 is a communications
protocol for multiplexing the bitstreams of up to 24 telephone calls, along with two
special bits: a framing bit (for frame synchronization) and a maintenance-signaling bit. T1's
maximum data transmission rate is 1.544 megabits per second.
Europe and most of the rest of the world, except Japan, have standardized the E-
carrier system, a similar transmission system with higher capacity that is not directly
compatible with the T-carrier.
The T-carrier system is entirely digital, using pulse code modulation (PCM) and time-
division multiplexing (TDM). The system uses four wires and provides duplex capability
(two wires for receiving and two for sending at the same time). The T1 digital stream
consists of 24 64-Kbps channels that are multiplexed. (The standardized 64 Kbps channel
is based on the bandwidth required for a voice conversation.) The four wires were
originally a pair of twisted pair copper wires, but can now also include coaxial cable, optical
fiber, digital microwave, and other media. A number of variations on the number and use of
channels are possible.
A T1 line in which each channel serves a different application is known as integrated T1 or
channelized T1. Another commonly installed service is a fractional T1, which is the rental
of some portion of the 24 channels in a T1 line, with the other channels going unused.
In the T1 system, voice or other analog signals are sampled 8,000 times a second and each
sample is digitized into an 8-bit word. With 24 channels being digitized at the same time, a
192-bit frame (24 channels each with an 8-bit word) is thus being transmitted 8,000 times
a second. Each frame is separated from the next by a single bit, making a 193-bit block. The
192 bit frame multiplied by 8,000 and the additional 8,000 framing bits make up the T1's
1.544 Mbps data rate. The signaling bits are the least significant bits in each frame.

Fractional T1
Fractional T1 is a fraction or part of a T1 line. In data transmission, T1 is one of the many
kinds of transmission lines that transport high-quality data at a lightning-fast rate. A T1
line has 24 channels, each of which can transfer data at a speed of 64 kilobytes per second.
In a fractional T1 line, only a part of the 24 channels are being rented out. This is done so
that clients who do not need all of the T1 capability can benefit from the speed of the
expensive line at a fraction of the cost.

Optical Carrier transmission rates

SONET and OC circuits
Optical Carrier transmission rates are a standardized set of specifications of
transmission bandwidth for digital signals that can be carried on Synchronous Optical
Networking (SONET) fiber optic networks.[1] Transmission rates are defined by rate of
the bitstream of the digital signal and are designated by hyphenation of the
acronym OC and an integer value of the multiple of the basic unit of rate, e.g., OC-48. The
base unit is 51.84 Mbit/s.[2] Thus, the speed of optical-carrier-classified lines labeled as OC-
n is n × 51.84 Mbit/s.
Optical Carrier classifications are based on the abbreviation OC followed by a number
specifying a multiple of 51.84 Mbit/s: n × 51.84 Mbit/s => OC-n. For example, an OC-3
transmission medium has 3 times the transmission capacity of OC-1.
OC-1
OC-1 is a SONET line with transmission speeds of up to 51.84 Mbit/s (payload:
50.112 Mbit/s; overhead: 1.728 Mbit/s) using optical fiber.
OC-3
OC-3 is a network line with transmission data rate of up to 155.52 Mbit/s (payload:
148.608 Mbit/s; overhead: 6.912 Mbit/s, including path overhead) using fiber optics.
Depending on the system OC-3 is also known as STS-3 (electrical level) and STM-1 (SDH).
OC-3c / STM-1
OC-3c ("c" stands for "concatenated") concatenates three STS-1 (OC-1) frames into a single
OC-3 look alike stream. The three STS-1 (OC-1) streams interleaved with each other such
that the first column is from the first stream, the second column is from the second stream,
and the third is from the third stream. Concatenated STS (OC) frames carry only one
column of path overhead because they cannot be divided into finer granularity signals.
Hence, OC-3c can transmit more payload to accommodate a CEPT-4 139.264 Mbit/s signal.
The payload rate is 149.76 Mbit/s and overhead is 5.76 Mbit/s.
OC-12 / STM-4
OC-12 is a network line with transmission speeds of up to 622.08 Mbit/s (payload:
601.344 Mbit/s; overhead: 20.736 Mbit/s).
OC-12 lines are commonly used by ISPs as wide area network (WAN) connections. While a
large ISP would not use an OC-12 as a backbone (main link), it would for smaller, regional
or local connections. This connection speed is also often used by mid-sized (below Tier 2)
internet customers, such as web hosting companies or smaller ISPs buying service from
larger ones.

OC-24
OC-24 is a network line with transmission speeds of up to 1244.16 Mbit/s (payload:
1202.208 Mbit/s (1.202208 Gbit/s); overhead: 41.472 Mbit/s). Implementations of OC-24
in commercial deployments are rare.
OC-48 / STM-16 / 2.5G SONET
OC-48 is a network line with transmission speeds of up to 2488.32 Mbit/s (payload:
2405.376 Mbit/s (2.405376 Gbit/s); overhead: 82.944 Mbit/s).
With relatively inexpensive interface prices, and being faster than OC-3, OC-12 connections,
and even surpassing gigabit Ethernet, OC-48 connections are used as the backbones of
many regional ISPs. Interconnections between large ISPs for purposes
of peering or transit are quite common. As of 2005, the only connections in widespread use
that surpass OC-48 speeds are OC-192 and 10 Gigabit Ethernet.
OC-48 is also used as transmission speed for tributaries from OC-192 nodes in order to
optimize card slot utilization where lower speed deployments are used. Dropping at OC-12,
OC-3 or STS-1 speeds are more commonly found on OC-48 terminals, where use of these
cards on an OC-192 would not allow for full use of the available bandwidth.
OC-192 / STM-64 / 10G SONET
OC-192 is a network line with transmission speeds of up to 9953.28 Mbit/s (payload:
9510.912 Mbit/s (9.510912 Gbit/s); overhead: 442.368 Mbit/s).
A standardized variant of 10 Gigabit Ethernet (10GbE), called WAN PHY, is designed to
inter-operate with OC-192 transport equipment while the common version of 10GbE is
called LAN PHY (which is not compatible with OC-192 transport equipment in its native
form). The naming is somewhat misleading, because both variants can be used on a wide
area network.
OC-768 / STM-256
OC-768 is a network line with transmission speeds of up to 39,813.12 Mbit/s (payload:
38,486.016 Mbit/s (38.486016 Gbit/s); overhead: 1,327.104 Mbit/s (1.327104 Gbit/s)).
On October 23, 2008, AT&T announced the completion of upgrades to OC-768 on 80,000
fiber-optic wavelength miles of their IP/MPLS backbone network.[3] OC-768 SONET
interfaces have been available with short-reach optical interfaces from Cisco since 2006.
Infinera made a field trial demonstration data transmission on a live production network
involving the service transmission of a 40 Gbit/s OC-768/STM-256 service over a 1,969 km
terrestrial network spanning Europe and the U.S. In November 2008, an OC-768 connection
was successfully brought up on the TAT-14/SeaGirt transatlantic cable,[4] the longest hop
being 7,500 km.
OC-1920 / STM-640
OC-1920 is a network line with transmission speeds of up to 99,532.8 Mbit/s
(99.5328 Gbit/s).
OC-3840 / STM-1280
OC-3840 is a network line with transmission speeds of up to 200 Gbit/s