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UNIT
– 3 |
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Telephone Network |
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Unit-03/Lecture-01 |
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Telephone
Network (RGPV dec 2011) A telephone network is a telecommunications
network used for telephone
calls between two or more
parties. There are a number of different types
of telephone network:
Public telephone
operators (PTOs) own and build
networks of the first two types and provide services to the public under
license from the national government. Virtual
Network Operators (VNOs) lease capacity
wholesale from the PTOs and sell on telephony service to the public directly. Network
Topology Network
topology is the arrangement of the various elements
(links, nodes, etc.) of a computer
network. Essentially, it is the topological structure of a network, and may be
depicted physically or logically. Physical
topology refers to the placement of the network's various components,
including device location and cable installation, while logical topology shows how data flows within a network,
regardless of its physical design. Distances
between nodes, physical interconnections, transmission rates, and/or signal types may differ between two networks, yet their
topologies may be identical. A good example is a local
area network (LAN): Any given node in
the LAN has one or more physical links to other
devices in the network; graphically mapping these links results in a
geometric shape that can be used to describe the physical topology of the
network. Conversely, mapping the data flow between the components determines
the logical topology of the network. 1. Mesh Topology In a mesh network, devices are connected with many redundant
interconnections between network nodes. In a true mesh topology every node
has a connection to every other node in the network. There are two types of mesh topologies: Full mesh topology occurs when every node has a circuit
connecting it to every other node in a network. Full mesh
is very expensive to implement but yields the greatest amount of redundancy,
so in the event that one of those nodes fails, network traffic can be
directed to any of the other nodes. Full mesh is usually reserved for
backbone networks.
Partial mesh topology is less expensive to implement and yields less redundancy than full
mesh topology. With partial mesh, some nodes are organized in a full mesh scheme but others are only connected to one
or two in the network. Partial mesh topology is commonly found in peripheral
networks connected to a full meshed backbone. 2. Star Topology In a star network devices are connected to a
central computer, called a hub. Nodes communicate
across the network by passing data through the hub. Advantage: In a star network, one malfunctioning node doesn't affect the rest of
the network. Disadvantage: If the central computer fails, the entire network becomes unusable. 3. Bus Topology Bus Topology: In networking a bus is the central cable -- the main wire -- that connects all
devices on a local-area network (LAN). It is also called the backbone. This is often used to describe the main
network connections composing the Internet. Bus networks are relatively
inexpensive and easy to install for small networks. Ethernet systems use a bus topology. Advantage: It's easy to connect a computer or device and typically it requires
less cable than a star topology. Disadvantage: The entire network shuts down if there is a break in the main wire and
it can be difficult to identify the problem if the network shuts down.
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Unit-03/Lecture-02 |
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4. Ring Topology Ring Topology:
A local-area network (LAN) whose topology is a ring. That is, all of the nodes are connected in
a closed loop. Messages travel around the ring, with each node reading those
messages addressed to it. One main advantage to a ring network is that it can
span larger distances than other types of networks, such as bus networks,
because each node regenerates messages as they pass through it. 5. Tree Topology This is a "hybrid" topology that combines characteristics of
linear bus and star topologies. In a tree network, groups of star-configured
networks are connected to a linear bus backbone cable. Advantage: A Tree topology is a good choice for
large computer networks as the tree topology
"divides" the whole network into parts that are more easily
manageable. Disadvantage: The entire network depends on a central hub and a failure of the
central hub can cripple the whole network. Signaling (RGPV dec 2013) Signalling
System No. 7 (SS7) is a set of telephony signaling protocols which are being used to set up
most of the world's public switched telephone network (PSTN) telephone calls. The main purpose
is to set up and tear down telephone calls. Other uses include number translation, local
number portability, prepaid billing mechanisms, short
message service (SMS), and a variety of
other mass market services. Functionality
The term signaling, when used in telephony, refers to the
exchange of control information associated with the setup and release of a
telephone call on a telecommunications circuit. An example of this control
information is the digits dialed by the caller, the caller's billing number,
and other call-related information. When the signaling is
performed on the same circuit that will ultimately carry the conversation of
the call, it is termed channel associated signaling (CAS). This is the case for earlier
analogue trunks, MF and R2 digital trunks, and DSS1/DASS PBX trunks. In contrast, SS7
signaling is termed Common Channel Signaling (CCS) in that the path and
facility used by the signaling is separate and distinct from the
telecommunications channels that will ultimately carry the telephone
conversation. With CCS, it becomes possible to exchange signaling without
first seizing a voice channel, leading to significant savings and performance
increases in both signaling and channel usage. Because of the mechanisms
used by signaling methods prior to SS7 (battery reversal, multi-frequency digit outpulsing, A-
and B-bit signaling), these older methods
could not communicate much signaling information. Usually only the dialed
digits were signaled, and only during call setup. For charged calls, dialed
digits and charge number digits were outpulsed. SS7, being a high-speed and
high-performance packet-based communications protocol, can communicate
significant amounts of information when setting up a call, during the call,
and at the end of the call. This permits rich call-related services to be
developed. Some of the first such services were call management related, call forwarding (busy and no answer), voice mail, call waiting, conference
calling, calling
name and number display, call
screening, malicious caller identification, busy
callback. The earliest deployed
upper layer protocols in the SS7 signaling suite
were dedicated to the setup, maintenance, and release of telephone calls. The
Telephone
User Part (TUP) was adopted in Europe and the Integrated Services Digital Network (ISDN) User Part (ISUP) adapted for public switched telephone network (PSTN) calls was adopted in North America.
ISUP was later used in Europe when the European
networks upgraded to the ISDN. (North America never accomplished full
upgrade to the ISDN and the predominant telephone
service is still the older POTS). Due to its richness and the need for an
out-of-band channel for its operation, SS7 signaling is mostly used for
signaling between telephone
switches and not for signaling
between local
exchanges and customer-premises
equipment (CPE). Because SS7 signaling
does not require seizure of a channel for a conversation prior to the
exchange of control information, non-facility associated signalling (NFAS) became possible. NFAS is signaling
that is not directly associated with the path that a conversation will
traverse and may concern other information located at a centralized database
such as service subscription, feature activation, and service logic. This
makes possible a set of network-based services that do not rely upon the call
being routed to a particular subscription switch at which service logic would
be executed, but permits service logic to be distributed throughout the telephone network and executed more expediently at originating switches far in advance of call routing. It
also permits the subscriber increased mobility due to the decoupling of
service logic from the subscription switch. Another characteristic of ISUP made possible by SS7 with NFAS is the
exchange of signaling information during the middle of a call. Also possible with SS7 is
Non-Call-Associated Signaling, which is signaling that is not directly
related to the establishment of a telephone call. An example of this is the
exchange of the registration information used between a mobile telephone and
a home
location register (HLR) database: a
database that tracks the location of the mobile. Other examples include Intelligent
Network and local
number portability databases.
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Unit-03/Lecture-03 |
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Physical network
SS7 separates signalling
from the voice circuits. An SS7 network must be made up of SS7-capable
equipment from end to end in order to provide its full functionality. The
network can be made up of several link types (A, B, C, D, E, and F) and three
signaling nodes - Service
switching point (SSPs), signal
transfer point (STPs), and service
control point (SCPs). Each node is
identified on the network by a number, a signalling point code. Extended
services are provided by a database interface at the SCP level using the SS7
network. The links between nodes
are full-duplex 56, 64, 1,536, or 1,984 kbit/s graded communications
channels. In Europe they are usually one (64 kbit/s) or all (1,984 kbit/s) timeslots (DS0s) within an E1 facility; in North America one (56 or 64
kbit/s) or all (1,536 kbit/s) timeslots (DS0As or DS0s) within a T1 facility. One or more signaling links can
be connected to the same two endpoints that together form a signaling link
set. Signaling links are added to link sets to increase the signaling
capacity of the link set. In Europe, SS7 links
normally are directly connected between switching exchanges using F-links.
This direct connection is called associated
signaling. In North America, SS7 links are normally indirectly
connected between switching exchanges using an intervening network of STPs.
This indirect connection is called quasi-associated
signaling. Quasi-associated signaling reduces the number of SS7 links necessary to interconnect
all switching exchanges and SCPs in an SS7 signaling network. Dial-Up
Modems Dial-up Internet access is a form of Internet
access that uses the facilities
of the public
switched telephone network (PSTN) to establish a dialed connection to
an Internet service provider (ISP) via telephone lines. The user's computer
or router uses an attached modem to encode and decode Internet Protocol packets and control information into and
from analogue audio frequency signals, respectively. Dial-up connections to the Internet require no infrastructure other than the telephone
network and the modems and
servers needed to make and answer the calls. Where telephone access is widely
available, dial-up remains useful and it is often the only choice available
for rural or remote areas, where broadband installations are not prevalent due to low
population density and high infrastructure cost. Dial-up access may also be
an alternative for users on limited budgets, as it is offered free by some
ISPs, though broadband is increasingly available at lower prices in many countries due to market competition. Dial-up requires time to
establish a telephone connection (up to several seconds, depending on the location)
and perform configuration for protocol synchronization before data transfers
can take place. In locales with telephone connection
charges, each connection incurs an incremental cost. If calls are
time-metered, the duration of the connection incurs costs. Dial-up access is a transient connection, because either
the user, ISP or phone company terminates the
connection. Internet service providers will often
set a limit on connection durations to allow sharing of resources, and will
disconnect the user—requiring reconnection and the costs and delays
associated with it. Technically inclined users often find a way to disable
the auto-disconnect program such that they can remain connected for days. Modem
(RGPV
dec 2013) A modem (modulator-demodulator) is a device that modulates an analog carrier signal to encode digital
information and demodulates the signal to decode the transmitted
information. The goal is to produce a signal that can be transmitted easily and decoded
to reproduce the original digital data. Modems can be used with any means of transmitting analog
signals, from light
emitting diodes to radio. The most familiar type is a voice band modem that turns the digital
data of a computer into modulated electrical
signals in the voice frequency
range of a telephone channel. These signals can be transmitted
over telephone lines and demodulated by another modem at the receiver side to recover the digital data. Modems are generally classified by the
amount of data they can send in a given unit of time, usually expressed in bits
per second (bit/s or bps), or bytes
per second (B/s). Modems can also
be classified by their symbol
rate, measured in baud. The baud unit denotes symbols per second,
or the number of times per second the modem sends a new signal. For example,
the ITU V.21 standard used audio frequency shift keying with two possible frequencies,
corresponding to two distinct symbols (or one bit
per symbol), to carry 300 bits per second using 300 baud. By contrast, the
original ITU V.22 standard, which could transmit and receive four distinct
symbols (two bits per symbol), transmitted 1,200 bits by sending 600 symbols
per second (600 baud) using phase shift keying. Modem
standards Dial-up modems utilize traditional copper phone lines to
transmit analog signals. Out of all types of modems —
dial-up, ISDN, DSL, cable — dial-up connections offer the slowest
transmission speeds. The CCITT, an international committee that specifies
the way modems and fax machines transmit information to ensure compatibility
among modems, has classified dial-up modems according to the following
modulation standards:
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Unit-03/Lecture-04 |
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List of dialup speeds
These values are maximum
values, and actual values may be slower under certain conditions (for
example, noisy phone lines). For a complete list see the companion article list of device bandwidths. A baud is one symbol per second; each symbol may
encode one or more data bits. Digital subscriber line (RGPV
dec 2013) Digital subscriber line is a family of technologies that provide Internet access by transmitting digital data over the wires of a local telephone
network. In telecommunications
marketing, the term DSL is widely understood to mean asymmetric digital subscriber line (ADSL), the most commonly installed DSL
technology. DSL service is delivered simultaneously with wired
telephone service on the same telephone
line. This is possible
because DSL uses higher frequency bands for data. On the customer premises, a DSL
filter on each non-DSL outlet blocks any high frequency interference, to
enable simultaneous use of the voice and DSL services. The bit
rate of consumer DSL services
typically ranges from 256 kbit/s to over 100 Mbit/s in the direction to the
customer (downstream), depending on DSL technology, line
conditions, and service-level implementation. Bit rates of 1 Gbit/s have been
reached in trials. In ADSL, the data throughput in the upstream direction, (the direction to the service
provider) is lower, hence the designation of asymmetric service. In symmetric digital subscriber line (SDSL) services, the downstream and
upstream data rates are equal. When you connect to the
Internet, you might connect through a regular modem, through a local-area network connection in your office, through a cable modem or through a digital subscriber line (DSL) connection. DSL is a very high-speed
connection that uses the same wires as a regular telephone line. Advantages
of DSL:
Disadvantages:
Telephone
Lines
If you have read How Telephones Work, then you know that a standard telephone
installation in the United States consists of a
pair of copper wires that the phone company installs in your home. The copper
wires have lots of room for carrying more than your phone conversations --
they are capable of handling a much greater bandwidth, or range of frequencies, than that demanded for voice. DSL
exploits this "extra capacity" to carry information on the wire
without disturbing the line's ability to carry conversations. The entire plan
is based on matching particular frequencies to specific tasks. To understand DSL, you first
need to know a couple of things about a normal telephone line the kind that
telephone professionals call POTS, for Plain Old
Telephone Service. One of the ways that POTS makes the most of the
telephone company's wires and equipment is by limiting the frequencies that
the switches, telephones and other equipment will carry. Human voices,
speaking in normal conversational tones, can be carried in a frequency range
of 0 to 3,400 Hertz (cycles per second -- see How Telephones Work for a great demonstration of this). This
range of frequencies is tiny. For example, compare this to the range of most
stereo speakers, which cover from roughly 20 Hertz to 20,000 Hertz. And the wires themselves have the
potential to handle frequencies up to several million Hertz in most cases. The use of such a small
portion of the wire's total bandwidth is historical remember that the
telephone system has been in place, using a pair of copper wires to each
home, for about a century. By limiting the frequencies carried over the
lines, the telephone system can pack lots of wires into a very small space
without worrying about interference between lines. Modern equipment that sends digital rather than
analog data can safely use much more of the telephone line's capacity. DSL
does just that. A DSL internet connection
is one of many effective communication tools for
keeping employees in touch with the office. |
|
S.NO |
RGPV
QUESTIONS |
Year |
Marks |
|
Q.1 |
Discuss
DSL with its advantages and disadvantages. |
DEC2013 |
4 |
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Unit-03/Lecture-05 |
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ADSL
(RGPV
dec 2012) Asymmetric
digital subscriber line (ADSL) is a type of digital
subscriber line (DSL) technology, a data
communications technology that enables faster data transmission over copper telephone lines than a conventional voiceband modem can provide. It does this by utilizing
frequencies that are not used by a voice telephone call. A splitter, or DSL
filter, allows a single
telephone connection to be used for both ADSL service
and voice calls at the same time. ADSL can generally only be distributed over
short distances from the telephone
exchange (the last
mile), typically less than 4
kilometres , but has been known to exceed 8 kilometres if the originally laid wire
gauge allows for further
distribution. At the telephone exchange
the line generally terminates at a digital subscriber line access
multiplexer (DSLAM) where another
frequency splitter separates the voice band signal for the conventional phone
network. Data carried by the ADSL are typically routed over the telephone
company's data network and
eventually reach a conventional Internet
Protocol network. FDD uses two separate
frequency bands, referred to as the upstream and downstream bands. The upstream band is used for communication from the
end user to the telephone central office. The downstream band is used for communicating from the
central office to the end user. Currently, most ADSL
communication is full-duplex. Full-duplex ADSL communication is usually
achieved on a wire pair by either frequency-division duplex (FDD),
echo-cancelling duplex (ECD), or time-division
duplex (TDD). FDD uses two
separate frequency bands, referred to as the upstream and downstream
bands. The upstream band is used for communication from the
end user to the telephone central office. The downstream band is used for communicating from the
central office to the end user. Frequency
plan for ADSL Red area is the frequency range used by normal voice telephony
(PSTN), the green (upstream) and
blue (downstream) areas are used for ADSL. With commonly deployed
ADSL over POTS , the band from 26.075 kHz to 137.825 kHz is used for upstream
communication, while 138 kHz – 1104 kHz is used for downstream communication.
Under the usual DMT scheme, each of these is further divided
into smaller frequency channels of 4.3125 kHz. These frequency channels are
sometimes termed bins. During
initial training to optimize transmission quality and speed, the ADSL
modem tests each of the bins
to determine the signal-to-noise
ratio at each bin's frequency.
Distance from the telephone
exchange, cable characteristics,
interference from AM radio stations, and local interference and electrical
noise at the modem's location can adversely affect the signal-to-noise
ratio at particular
frequencies. Bins for frequencies exhibiting a reduced signal-to-noise ratio
will be used at a lower throughput rate or not at all; this reduces the
maximum link capacity but allows the modem to maintain an adequate
connection. The DSL modem will make a plan on how to exploit each of the
bins, sometimes termed "bits per bin" allocation. Those bins that
have a good signal-to-noise ratio (SNR) will be chosen to transmit signals
chosen from a greater number of possible encoded values (this range of
possibilities equating to more bits of data sent) in each main clock cycle.
The number of possibilities must not be so large that the receiver might
incorrectly decode which one was intended in the presence of noise. Noisy
bins may only be required to carry as few as two bits, a choice from only one
of four possible patterns, or only one bit per bin in the case of ADSL2+, and
very noisy bins are not used at all. If the pattern of noise versus
frequencies heard in the bins changes, the DSL modem can alter the
bits-per-bin allocations, in a process called "bitswap", where bins
that have become more noisy are only required to carry fewer bits and other
channels will be chosen to be given a higher burden. The data transfer
capacity the DSL modem therefore reports is determined by the total of the
bits-per-bin allocations of all the bins combined. Higher signal-to-noise
ratios and more bins being in use gives a higher total link capacity, while
lower signal-to-noise ratios or fewer bins being used gives a low link
capacity. The total maximum
capacity derived from summing the bits-per-bin is reported by DSL modems and
is sometimes termed sync rate.
This will always be rather misleading, as the true maximum link capacity for
user data transfer rate will be significantly lower; because extra data are
transmitted that are termed protocol overhead, reduced figures for PPPoA
connections of around 84-87 percent, at most, being common. In addition, some
ISPs will have traffic policies that limit maximum transfer rates further in
the networks beyond the exchange, and traffic congestion on the Internet,
heavy loading on servers and slowness or inefficiency in customers' computers
may all contribute to reductions below the maximum attainable. When a
wireless access point is used, low or unstable wireless signal quality can
also cause reduction or fluctuation of actual speed. In fixed-rate mode, the sync rate is predefined by the
operator and the DSL modem chooses a bits-per-bin allocation that yields an
approximately equal error rate in each bin.In variable-rate mode, the
bits-per-bin are chosen to maximize the sync rate, subject to a tolerable
error risk. These choices can either be conservative, where the modem chooses
to allocate fewer bits per bin than it possibly could, a choice which makes
for a slower connection, or less conservative in which more bits per bin are
chosen in which case there is a greater risk case of error should future
signal-to-noise ratios deteriorate to the point where the bits-per-bin
allocations chosen are too high to cope with the greater noise present. This
conservatism, involving a choice of using fewer bits per bin as a safeguard against
future noise increases, is reported as the signal-to-noise ratio margin or SNR margin. The telephone exchange can indicate a suggested
SNR margin to the customer's DSL modem when it initially connects, and the
modem may make its bits-per-bin allocation plan accordingly. A high SNR
margin will mean a reduced maximum throughput, but greater reliability and
stability of the connection. A low SNR margin will mean high speeds, provided
the noise level does not increase too much; otherwise, the connection will
have to be dropped and renegotiated (resynced). ADSL2+ can better accommodate
such circumstances, offering a feature termed seamless rate adaptation (SRA), which can accommodate changes
in total link capacity with less disruption to communications. Frequency spectrum of modem on ADSL line Vendors may support usage
of higher frequencies as a proprietary extension to the standard. However,
this requires matching vendor-supplied equipment on both ends of the line,
and will likely result in crosstalk problems that affect other lines in the
same bundle. There is a direct
relationship between the number of channels available and the throughput
capacity of the ADSL connection. The exact data capacity per channel depends
on the modulation method used. ADSL initially existed in two versions
(similar to VDSL), namely CAP and DMT. CAP was the de facto standard for ADSL deployments up until 1996, deployed
in 90 percent of ADSL installations at the time. However, DMT was chosen for
the first ITU-T ADSL standards, G.992.1 and G.992.2
(also called G.dmt and G.lite respectively). Therefore all
modern installations of ADSL are based on the DMT modulation scheme. |
|
S.NO |
RGPV
QUESTIONS |
Year |
Marks |
|
Q.1 |
What
is the purpose of DSLAM. Why is ADSL is unsuitable for business. |
Dec
2012 |
7 |
|
Q.2 |
Which
DSL technology is best suited for businessman. |
DEC
2013 |
4 |
|
UNIT 3/LECTURE 6 |
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SDSL Symmetric
digital subscriber line (SDSL) can have two meanings:
Proprietary SDSL technology
SDSL is a rate-adaptive
digital subscriber line (DSL) variant with T1/E1-like data rates (T1: 1.544 Mbit/s, E1: 2.048 Mbit/s). It runs over one pair
of copper wires, with a maximum range of 10,000 feet (3,000 m). It cannot co-exist with a
conventional voice service on the same pair as it takes over the entire
bandwidth. VDSL
(RGPV
dec 2012) Very-high-bit-rate
digital subscriber line (VDSL or VHDSL) is
a digital
subscriber line (DSL) technology
providing data transmission faster than ADSL over a single flat untwisted or twisted
pair of copper wires (up to
52 Mbit/s downstream and 16 Mbit/s upstream), and on coaxial
cable (up to 85 Mbit/s down-
and upstream) using the frequency band from 25 kHz to 12 MHz. These rates mean that VDSL is
capable of supporting applications such as high-definition
television, as well as telephone
services (voice over IP) and general Internet access, over a single connection. VDSL is
deployed over existing wiring used for analog
telephone service and lower-speed
DSL connections. This standard was approved by ITU in November 2001. Second-generation systems
(VDSL2; ITU-T G.993.2 approved in February 2006)
use frequencies of up to 30 MHz to provide data rates exceeding 100 Mbit/s
simultaneously in both the upstream and downstream directions. The maximum
available bit rate is achieved at a range of about 300 meters; performance
degrades as the loop attenuation increases. VDSL standards
A VDSL connection uses up
to seven frequency bands, so one can allocate the data rate between upstream
and downstream differently depending on the service offering and spectrum
regulations. First generation VDSL standard specified both quadrature amplitude modulation (QAM) and discrete multi-tone modulation (DMT). In 2006, ITU-T
standardized VDSL in recommendation G.993.2 which specified only DMT
modulation for VDSL2. Multiplexing In telecommunications and computer
networks, multiplexing is a method by which multiple analog message
signals or digital data streams are combined into one signal over a shared
medium. The aim is to share an
expensive resource. For example, in telecommunications, several telephone
calls may be carried using one
wire. Multiplexing originated in telegraphy in the 1870s, and is now widely applied in
communications. In telephony, George
Owen Squier is credited with the
development of telephone carrier multiplexing in
1910. The multiplexed signal is
transmitted over a communication
channel, which may be a physical
transmission medium. The multiplexing divides the capacity of the high-level
communication channel into several low-level logical channels, one for each
message signal or data
stream to be transferred. A reverse process, known as demultiplexing, can
extract the original channels on the receiver side. A device that performs
the multiplexing is called a multiplexer (MUX), and a device that performs the reverse process is called a demultiplexer (DEMUX or DMX). Inverse
multiplexing (IMUX) has the opposite
aim as multiplexing, namely to break one data stream into several streams,
transfer them simultaneously over several communication channels, and
recreate the original data stream. Frequency-division multiplexing (RGPV dec 2013) Frequency-division multiplexing (FDM): The
spectrum of each input signal is shifted to a distinct frequency range. Frequency-division multiplexing (FDM) is
inherently an analog technology. FDM achieves the combining of several
signals into one medium by sending signals in several distinct frequency
ranges over a single medium. One of FDM's most common
applications is the old traditional radio and television broadcasting from
terrestrial, mobile or satellite stations, using the natural atmosphere of
Earth, or the cable television. Only one cable reaches a customer's
residential area, but the service provider can send multiple television channels
or signals simultaneously over that cable to all subscribers without
interference. Receivers must tune to the appropriate frequency (channel) to
access the desired signal.
|
|
UNIT 3/LECTURE 7 |
Time-division multiplexing
Time-division
multiplexing (TDM) is a digital (or
in rare cases, analog) technology which uses time, instead of space or
frequency, to separate the different data streams. TDM involves sequencing
groups of a few bits or bytes from each individual input stream, one after
the other, and in such a way that they can be associated with the appropriate
receiver. If done sufficiently quickly, the receiving devices will not detect
that some of the circuit time was used to serve another logical communication
path. Consider an application
requiring four terminals at an airport to reach a central computer. Each
terminal communicated at 2400 bit/s, so rather than acquire four individual
circuits to carry such a low-speed transmission, the airline has installed a
pair of multiplexers. A pair of 9600 bit/s modems and one dedicated analog
communications circuit from the airport ticket desk back to the airline data
center are also installed. Carrier sense multiple access and multidrop communication methods are similar to
time-division multiplexing in that multiple data streams are separated by
time on the same medium, but because the signals have separate origins
instead of being combined into a single signal, are best viewed as channel
access methods, rather than a form of
multiplexing. Wave
length division multiplexing In fiber-optic
communications, wavelength-division multiplexing (WDM) is a technology which multiplexes a number of optical
carrier signals onto a single optical
fiber by using different wavelengths (i.e., colors) of laser light. This technique enables bidirectional communications over one strand of fiber,
as well as multiplication of capacity. The term wavelength-division multiplexing is
commonly applied to an optical carrier (which is typically described by its
wavelength), whereas frequency-division multiplexing typically applies to a radio carrier
(which is more often described by frequency). Since wavelength and frequency
are tied together through a simple directly inverse relationship, in which
the product of frequency and wavelength equals c (the
propagation speed of light), the two terms actually describe the same
concept. Wavelength-division
multiplexing (WDM) is a method of combining multiple signals on laser beams at various infared (IR) wavelengths for transmission along fiber optic media. Each laser is modulated by an
independent set of signals. Wavelength-sensitive filters, the IR analog of
visible-light color filters, are used at the receiving end. WDM is similar to
frequency-division multiplexing (FDM). But instead of taking place at radio
frequencies (RF), WDM is done in the IR portion of the electromagnetic spectrum. Each IR channel carries several RF
signals combined by means of FDM or time-division multiplexing (TDM). Each multiplexed IR channel is
separated, or demultiplexed, into the original signals at the destination.
Using FDM or TDM in each IR channel in combination with WDM or several IR
channels, data in different formats and at different speeds can be
transmitted simultaneously on a single fiber. In early WDM systems,
there were two IR channels per fiber. At the destination, the IR channels
were demultiplexed by a dichroic (two-wavelength) filter with a cutoff
wavelength approximately midway between the wavelengths of the two channels.
It soon became clear that more than two multiplexed IR channels could be
demultiplexed using cascaded dichroic filters, giving rise to coarse
wavelength-division multiplexing (CWDM) and dense wavelength-division
multiplexing (DWDM). In CWDM, there are usually eight
different IR channels, but there can be up to 18. In DWDM, there can be
dozens. Because each IR channel carries its own set of multiplexed RF signals, it is theoretically possible
to transmit combined data on a single fiber at a total effective speed of
several hundred gigabitsper second (Gbps). The use of WDM can
multiply the effective bandwidth of a fiber optic communications system by
a large factor, but its cost must be weighed against the alternative of using
multiple fibers bundled into a cable. A fiber optic repeater device called the erbium amplifier can make WDM a cost-effective long-term
solution. Pulse
code modulation Pulse code modulation
(PCM) is a digital scheme for transmitting analogdata. The signals in PCM are binary; that
is, there are only two possible states, represented by logic 1 (high) and
logic0 (low). This is true no matter how complex the analog waveform happens
to be. Using PCM, it is possible to digitize all forms of analog data,
including full-motion video, voices, music, telemetry, and virtual reality
(VR). To obtain PCM from an analog waveform at the source (transmitter end) of a
communications circuit, the analog signal amplitude is sampled (measured) at
regular time intervals.The sampling rate, or number of samples per second, is
several times the maximum frequency of the analog
waveform in cycles per second or hertz. The instantaneous amplitude of the analog
signal at each sampling is rounded off to the nearest of several specific,
predetermined levels. This process is called quantization. The number of
levels is always a power of 2.for example, 8, 16, 32, or 64. These numbers
can be represented by three, four, five, or six binary digits
(bits)respectively. The output of a pulse code modulator is thus a series of
binary numbers, each represented by some power of 2bits. At the destination
(receiver end) of the communications circuit, a pulse code demodulator
converts the binary numbers back into pulses having the same quantum levels
as those in the modulator. These pulses are further processed to restore the
original analog waveform. PDH (RGPV dec 2012/2013) The plesiochronous digital hierarchy (PDH) is a technology used in telecommunications
networks to transport large
quantities of data over digital transport equipment such as fibre
optic and microwave radio systems. The term plesiochronous is derived from Greek
plēsios, meaning near,
and chronos, time, and refers
to the fact that PDH networks run in a state where
different parts of the network are nearly, but not quite perfectly, synchronised. PDH is typically being
replaced by synchronous digital hierarchy (SDH) or synchronous optical networking
(SONET) equipment in most telecommunications networks. PDH allows transmission
of data streams that are nominally running at the same rate, but allowing
some variation on the speed around a nominal rate. By analogy, any two
watches are nominally running at the same rate, clocking up 60 seconds every
minute. However, there is no link between watches to guarantee they run at
exactly the same rate, and it is highly likely that one is running slightly
faster than the other. Implementation
The data rate is
controlled by a clock in the equipment generating the data. The rate is
allowed to vary by ±50 ppm of 2.048 Kbit/s (according to ITU-T
recommendation. This means that different data streams can be
(probably are) running at slightly different rates to one another. In order to move multiple
data streams from one place to another, they are multiplexed in groups of
four. This is done by taking 1 bit from stream #1, followed by 1 bit from
stream #2, then #3, then #4. The transmitting multiplexer also adds additional
bits in order to allow the far end receiving multiplexer to decode
which bits belong to which data stream, and so correctly reconstitute the
original data streams. These additional bits are called
"justification" or "stuffing" bits. Because each of the four data streams is not necessarily running at the same rate,
some compensation has to be introduced. The transmitting multiplexer combines
the four data streams assuming that they are running at their maximum allowed
rate. This means that occasionally, (unless the 2 Mbit/s really is running at
the maximum rate) the multiplexer will look for the next bit but it will not
have arrived. In this case, the multiplexer signals to the receiving
multiplexer that a bit is "missing". This allows the receiving
multiplexer to correctly reconstruct the original data for each of the four 2
Mbit/s data streams, and at the correct, different, plesiochronous rates. The resulting data stream
from the above process runs at 8.448 Mbit/s (about
8 Mbit/s). Similar techniques are used to combine four × 8 Mbit/s together,
plus bit stuffing, giving 34 Mbit/s. Four × 34 Mbit/s, gives
140. Four × 140 gives 565.
|
|
S.NO |
RGPV
QUESTION |
YEAR |
MARKS |
|
Q.1 Q2 |
What
is the difference between pleisynchronous digital hierarchy and synchronous
digital hierarchy Discuss
the standard of pleisynchronous digital hierarchy. |
Dec2013 DEC
2012 |
7 7 |
|
UNIT 3/LECTURE 8 |
|
Difference Between Pdh And Sdh (RGPV dec 2011/2013) SDH which stands for Synchronous Digital Hierarchy was developed
recently and was made to do away with PDH's weaknesses while PDH which stands
for Plesiochronous Digital Hierarchy, is a type that is used for of data
transferring data mainly in large file groups. PDH was made around 1990 is
cheap and compatible compared to SDH. Stm
1 frame STM-1 The STM-1 (Synchronous Transport Module level-1) is the SDH ITU-T fiber
optic network transmission standard. It has a bit rate
of 155.52 Mbit/s. Higher levels go up by a factor of 4 at a time: the other
currently supported levels are STM-4, STM-16, STM-64 and STM-256. Beyond this we have wavelength-division multiplexing (WDM) commonly used in submarine cabling. Frame structure The STM-1 frame is the
basic transmission format for SDH (Synchronous Digital
Hierarchy). A STM-1 frame has a byte-oriented structure with 9 rows
and 270 columns of bytes, for a total of 2,430 bytes (9 rows * 270 columns =
2430 bytes). Each byte corresponds to a 64kbit/s channel. TOH: Transport
Overhead (RSOH + AU4P + MSOH)
VC4: Virtual Container-4 payload (POH + VC-4 Data)
Frame characteristics
The STM-1 base frame is structured with the following
characteristics:
RSOH (regenerator section overhead)
The Regenerator Section
OverHead uses the first three rows & nine columns in the STM-1 frame
|
|
S.NO |
RGPV
QUESTION |
YEAR |
MARKS |
|
Q.1 |
Discuss
the standards of sdh hierarchy. |
Dec2013 Dec2011 |
7 4 |
|
UNIT 3/LECTURE 9 |
MSOH (multiplex section overhead) (RGPV dec
2011)
X = Bytes reserved for national use. The Multiplex Section
OverHead uses the 5th through 9th rows, and first 9 columns in the
STM-1 frame.
Q : Five channels each with a 100khz bandwidth are to be
multiplied together.What is the minimum bandwidth of the link if there is a
need for a guard band of 10 khz between the channels to prevent interference. Sol : For five channels we need at least four guard
bands.This means that the required bandwidth is at least 5 |
|
S.NO |
RGPV
QUESTION |
YEAR |
MARKS |
|
Q.1 |
Define
multiplex section overhead. |
Dec
2011 |
4 |
